Solved: Branch office CUBE configuration

Matt Smith · ‎05-01-2013

Hello,

I have a voice routing topology question and would like some hints in which documentation I can find what I'm looking for:

Currently:

Running CUCM BE6K at our HQ in North America.

Have a local SIP trunk to our ITSP which is proxy'd by a 2951

Device partition for NA devices

A small branch office in Germany is using a different phone system and needs to be migrated over to CUCM.

Near Future:

2901 CUBE and phones only devices at branch office in Germany.

Have a local SIP trunk to our ITSP which is proxy'd by a 2901

Device partition for German office devices.

I would like to avoid the media stream traversing the ocean back to NA for phone calls originating/terminating in EMEA, and instead have them go out the locally provided trunk.

Any Inter-office call media streams would of course traverse the VPN between offices. I would also like to setup SRST on the 2901 for the German office, whereby the local phones can register if there is an issue with the VPN connection back to CUCM.

What I need:

I'm having some issues finding documentation which outlines how to configure this setup from both the CUCM side and from the CUBE side, and I'm wondering if this setup is supported.

I've found some information that leads me to think a dspfarm on the 2901 is part of the solution to transcode between the trunk and German office phones.

I believe configuring the 2901 as an MTP on CUCM is part of the solution.

Additionally I see some potential problems:

Unity is located in NA and would handle all voicemail for both locations

Uccx is located in NA and would handle IVR (rare occurance)

Any help is appreciated, I currently have the hardware as dicussed and am working through the configurations.

Steve

Ayodeji Okanlawon · ‎05-24-2013

From the logs I can see that you are doing DO to DO. That implies that you are not offering any SDP to your ITSP for outbound calls...

Configure this and test again on the german cube

conf t

voice service voip

sip

early-offer forced

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View solution in original post

dsobrinho · ‎05-01-2013

Hi Steve,

It is a lot of information. Let's party like Jack. Give more information about the config of CUBE.

Best regards,

Daniel

Daniel Sobrinho

Matt Smith · ‎05-02-2013

What info are you looking for? The branch office 2901 config is pretty minimal. (and as it stands the branch equipment is sitting on my desk).

Ayodeji Okanlawon · ‎05-02-2013

Steve,

It is pretty straight forward...

1. You will need to configure two sip trunks on CUCM

a. NA SIP trunk pointd to NA CUBE

b. Germany SIP turnk..Points to Gernamy CUBE

2. You will need to create seperate RL, RG and RP..

The NA RP will point to the NA RG

The Germany RP will pointo tthe Germany RG

3. Each phone will have access to their respective RP

4. You then need to configure your CUBE in each location.

You can look at this thread for some recommended best practice on setting up CUBE etc

https://supportforums.cisco.com/message/3884309#3884309

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Matt Smith · ‎05-23-2013

Ok, had some time to look at this again.

Currently I have the German CUBE up and running, but I'm not sure how to stop the media stream from transitioning between German CUBE and NA CUCM.

Right now I believe the media stream is:

Inbound call to German # --> German CUBE --> SIP trunk to CUCM --> SCCP to phone over VPN --> SCCP to CUCM --> SIP trunk to CUBE --> to ITSP

I took a look at the other thread you mentioned, but I'm not sure which part applies. I'd like the CUBE to signal CUCM that a call is destined for one of the phones in the German phones partition, and have the local German phone make the media connection to the local German CUBE.

I think the MTP resources of the German CUBE is what I'm missing.

Here is the current config of the German CUBE:

!

! Last configuration change at 13:38:48 UTC Wed May 22 2013 by root

version 15.2

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname 2901-01.miovision.corp

!

boot-start-marker

boot-end-marker

!

logging buffered 51200 warnings

!

no aaa new-model

!

ip cef

!

ip dhcp excluded-address 10.10.10.1

!

ip domain name yourdomain.com

ip name-server 8.8.8.8

ip name-server 8.8.4.4

no ipv6 cef

!

multilink bundle-name authenticated

!

crypto pki trustpoint TP-self-signed-3973529459

enrollment selfsigned

subject-name cn=IOS-Self-Signed-Certificate-3973529459

revocation-check none

rsakeypair TP-self-signed-3973529459

!

crypto pki certificate chain TP-self-signed-3973529459

certificate self-signed 01

[snip]

quit

voice-card 0

dspfarm

dsp services dspfarm

!

voice service voip

ip address trusted list

no ip address trusted authenticate

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

pass-thru content sdp

registration passthrough

!

voice class codec 1

codec preference 1 g711alaw

!

voice class sip-profiles 102

request INVITE sip-header From modify "10.0.249.6" "sip.dcalling.de"

!

voice class sip-profiles 103

request INVITE sip-header From modify "500@10.0.249.6" "[snip]@sip.dcalling.de"

!

media service

!

license udi pid CISCO2901/K9 sn FGL170810CM

hw-module pvdm 0/0

!

username root privilege 15 secret 4 [snip]

!

redundancy

!

ip ssh time-out 60

ip ssh source-interface GigabitEthernet0/0.7

ip ssh version 2

ip ssh pubkey-chain

username root

csdb tcp synwait-time 30

csdb tcp idle-time 3600

csdb tcp finwait-time 5

csdb tcp reassembly max-memory 1024

csdb tcp reassembly max-queue-length 16

csdb udp idle-time 30

csdb icmp idle-time 10

csdb session max-session 65535

!

interface Embedded-Service-Engine0/0

no ip address

shutdown

!

interface GigabitEthernet0/0

description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$

no ip address

duplex auto

speed auto

!

interface GigabitEthernet0/0.7

encapsulation dot1Q 7

ip address 10.0.249.6 255.255.255.0

!

interface GigabitEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

ip forward-protocol nd

!

ip http server

ip http access-class 23

ip http authentication local

ip http secure-server

ip http timeout-policy idle 60 life 86400 requests 10000

!

ip nat sip-sbc

ip route 0.0.0.0 0.0.0.0 10.0.249.1

!

access-list 23 permit 10.0.6.30

access-list 23 permit 10.0.9.39

access-list 23 permit 10.10.10.0 0.0.0.7

access-list 23 permit 10.0.0.0 0.255.255.255

access-list 99 permit 85.88.5.252

access-list 99 permit 85.88.5.240

!

control-plane

!

mgcp profile default

!

sccp local GigabitEthernet0/0.7

sccp ccm 10.0.6.30 identifier 1 version 7.0

!

dial-peer voice 10 voip

description ** Incoming call from dcalling trunk DID:22145580030 **

destination-pattern 22145580030

session protocol sipv2

session target ipv4:10.0.6.30

session transport udp

incoming called-number 22145580030

voice-class codec 1

dtmf-relay rtp-nte

dtmf-interworking standard

no vad

!

dial-peer voice 50 voip

!

dial-peer voice 30 voip

description ** Outgoing call to dcalling trunk **

destination-pattern .T

session protocol sipv2

session target dns:sip.dcalling.de

session transport udp

incoming called-number .T

voice-class codec 1

voice-class sip profiles 103

dtmf-interworking standard

no vad

!

dial-peer voice 11 voip

description ** Incoming call, push to cucm for account [snip] **

destination-pattern [snip]

session protocol sipv2

session target ipv4:10.0.6.30

session transport udp

voice-class codec 1

dtmf-relay rtp-nte

dtmf-interworking standard

no vad

!

sip-ua

credentials number [snip] username [snip] password 7 [snip] realm sip.dcalling.de

authentication username [snip] password 7 [snip] realm sip.dcalling.de

registrar 1 dns:sip.dcalling.de expires 60

!

gatekeeper

shutdown

!

telephony-service

sdspfarm units 1

sdspfarm transcode sessions 20

sdspfarm tag 1 2901-01

max-conferences 8 gain -6

transfer-system full-consult

!

banner exec

% Password expiration warning.

-----------------------------------------------------------------------

Cisco Configuration Professional (Cisco CP) is installed on this device

and it provides the default username "cisco" for one-time use. If you have

already used the username "cisco" to login to the router and your IOS image

supports the "one-time" user option, then this username has already expired.

You will not be able to login to the router with this username after you exit

this session.

It is strongly suggested that you create a new username with a privilege level

of 15 using the following command.

username privilege 15 secret 0

Replace and with the username and password you want to

use.

-----------------------------------------------------------------------

!

line con 0

login local

line aux 0

line 2

no activation-character

no exec

transport preferred none

transport input all

transport output pad telnet rlogin lapb-ta mop udptn v120 ssh

stopbits 1

line vty 0 4

session-timeout 60

access-class 23 in

privilege level 15

login local

transport input telnet ssh

line vty 5 15

access-class 23 in

privilege level 15

login local

transport input telnet ssh

!

scheduler allocate 20000 1000

!

end

Ayodeji Okanlawon · ‎05-23-2013

Steve,

Media never terminates on CUCM, unless in scenarios where an MTP is used. Media will always terminate on CUBE and the IP Phone. So if a call comes in to your germany CUBE and this call is for a phone in germany then your media will sit locally between the cube and the ip phone. What you will have going to CUCM in NA is signalling traffic. You are going to have both sccp traffic and sip traffic going to NA..since thats where the CUCM is.

If you are seeing media stream going to CUCM then it is most likely that MTP is been invoked for your calls. This is likely due to DTMF mismatch between the dmtf method negotiated between the endpoints and the sip trunk

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Matt Smith · ‎05-23-2013

Interesting.

I had the "Media Termination Point Required" box checked on the trunk config in CUCM.

After unchecking it, I tried to dial out from the German side and have no inbound audio (including ring tones, absolutely zero speaker audio), but do have outbound audio.

If I dial out from NA to Germany I have audio in both directions.

On the NA CUBE, when calling NA -> Germany #show sip calls:

Total SIP call legs:3, User Agent Client:1, User Agent Server:2

SIP UAC CALL INFO

Call 1

SIP Call ID : A97BA436-C32B11E2-873C8982-4AB77B30@10.0.250.4

State of the call : STATE_ACTIVE (7)

Substate of the call : SUBSTATE_NONE (0)

Calling Number : anonymous

Called Number : 5195132407

Bit Flags : 0xC04018 0x10000100 0x80

CC Call ID : 85334

Source IP Address (Sig ): 10.0.250.4

Destn SIP Req Addr:Port : [10.0.6.30]:5060

Destn SIP Resp Addr:Port: [10.0.6.30]:5060

Destination Name : 10.0.6.30

Number of Media Streams : 2

Number of Active Streams: 1

RTP Fork Object : 0x0

Media Mode : flow-through

Media Stream 1

State of the stream : STREAM_ACTIVE

Stream Call ID : 85334

Stream Type : voice+dtmf (1)

Stream Media Addr Type : 1

Negotiated Codec : g711ulaw (160 bytes)

Codec Payload Type : 0

Negotiated Dtmf-relay : rtp-nte

Dtmf-relay Payload Type : 101

QoS ID : -1

Local QoS Strength : BestEffort

Negotiated QoS Strength : BestEffort

Negotiated QoS Direction : None

Local QoS Status : None

Media Source IP Addr:Port: [10.0.250.4]:27046

Media Dest IP Addr:Port : [10.0.6.30]:30010

Media Stream 2

State of the stream : STREAM_DEAD

Stream Call ID : -1

Stream Type : video (7)

Stream Media Addr Type : 1

Negotiated Codec : No Codec (0 bytes)

Codec Payload Type : 255 (None)

Negotiated Dtmf-relay : inband-voice

Dtmf-relay Payload Type : 0

QoS ID : -1

Local QoS Strength : BestEffort

Negotiated QoS Strength : BestEffort

Negotiated QoS Direction : None

Local QoS Status : None

Media Source IP Addr:Port: [10.0.250.4]:0

Media Dest IP Addr:Port : [10.0.6.30]:0

Options-Ping ENABLED:NO ACTIVE:NO

Number of SIP User Agent Client(UAC) calls: 1

SIP UAS CALL INFO

Call 1

SIP Call ID : 0453a39242d8d1071de6042362e0f79f@173.46.30.12

State of the call : STATE_ACTIVE (7)

Substate of the call : SUBSTATE_NONE (0)

Calling Number : 1264

Called Number : 5195132407

Bit Flags : 0xC0401C 0x10000100 0x4

CC Call ID : 85332

Source IP Address (Sig ): 10.0.1.67

Destn SIP Req Addr:Port : [173.46.30.202]:5060

Destn SIP Resp Addr:Port: [173.46.30.202]:5060

Destination Name : 173.46.30.202

Number of Media Streams : 2

Number of Active Streams: 1

RTP Fork Object : 0x0

Media Mode : flow-through

Media Stream 1

State of the stream : STREAM_ACTIVE

Stream Call ID : 85332

Stream Type : voice+dtmf (0)

Stream Media Addr Type : 1

Negotiated Codec : g711ulaw (160 bytes)

Codec Payload Type : 0

Negotiated Dtmf-relay : rtp-nte

Dtmf-relay Payload Type : 101

QoS ID : -1

Local QoS Strength : BestEffort

Negotiated QoS Strength : BestEffort

Negotiated QoS Direction : None

Local QoS Status : None

Media Source IP Addr:Port: [10.0.1.67]:27132

Media Dest IP Addr:Port : [173.46.30.202]:20766

Media Stream 2

State of the stream : STREAM_DEAD

Stream Call ID : -1

Stream Type : video (7)

Stream Media Addr Type : 1

Negotiated Codec : h264 (0 bytes)

Codec Payload Type : 99

Negotiated Dtmf-relay : inband-voice

Dtmf-relay Payload Type : 0

QoS ID : -1

Local QoS Strength : BestEffort

Negotiated QoS Strength : BestEffort

Negotiated QoS Direction : None

Local QoS Status : None

Media Source IP Addr:Port: [10.0.1.67]:0

Media Dest IP Addr:Port : [173.46.30.202]:0

On the German CUBE, calling NA -> Germany #sip show calls:

Total SIP call legs:2, User Agent Client:1, User Agent Server:1

SIP UAC CALL INFO

Call 1

SIP Call ID : 66B40749-C32B11E2-AEFBE3B7-54AA1348@10.0.249.6

State of the call : STATE_ACTIVE (7)

Substate of the call : SUBSTATE_NONE (0)

Calling Number : 0015195132407

Called Number : 8686170390

Bit Flags : 0xC04018 0x10000100 0x80

CC Call ID : 22122

Source IP Address (Sig ): 10.0.249.6

Destn SIP Req Addr:Port : [10.0.6.30]:5060

Destn SIP Resp Addr:Port: [10.0.6.30]:5060

Destination Name : 10.0.6.30

Number of Media Streams : 1

Number of Active Streams: 1

RTP Fork Object : 0x0

Media Mode : flow-through

Media Stream 1

State of the stream : STREAM_ACTIVE

Stream Call ID : 22122

Stream Type : voice+dtmf (1)

Stream Media Addr Type : 1

Negotiated Codec : Passthrough (0 bytes)

Codec Payload Type : 255 (None)

Negotiated Dtmf-relay : inband-voice

Dtmf-relay Payload Type : 0

QoS ID : -1

Local QoS Strength : BestEffort

Negotiated QoS Strength : BestEffort

Negotiated QoS Direction : None

Local QoS Status : None

Media Source IP Addr:Port: [10.0.249.6]:17140

Media Dest IP Addr:Port : [10.0.249.101]:22858

Options-Ping ENABLED:NO ACTIVE:NO

Number of SIP User Agent Client(UAC) calls: 1

SIP UAS CALL INFO

Call 1

SIP Call ID : 550e9d824501dfa76c0d2c977aa4dcde@85.88.5.229:5060

State of the call : STATE_ACTIVE (7)

Substate of the call : SUBSTATE_NONE (0)

Calling Number : 0015195132407

Called Number : 8686170390

Bit Flags : 0xC0401C 0x10000100 0x4

CC Call ID : 22121

Source IP Address (Sig ): 10.0.249.6

Destn SIP Req Addr:Port : [85.88.5.252]:5060

Destn SIP Resp Addr:Port: [85.88.5.252]:5060

Destination Name : 85.88.5.252

Number of Media Streams : 1

Number of Active Streams: 1

RTP Fork Object : 0x0

Media Mode : flow-through

Media Stream 1

State of the stream : STREAM_ACTIVE

Stream Call ID : 22121

Stream Type : voice+dtmf (1)

Stream Media Addr Type : 1

Negotiated Codec : Passthrough (0 bytes)

Codec Payload Type : 255 (None)

Negotiated Dtmf-relay : inband-voice

Dtmf-relay Payload Type : 0

QoS ID : -1

Local QoS Strength : BestEffort

Negotiated QoS Strength : BestEffort

Negotiated QoS Direction : None

Local QoS Status : None

Media Source IP Addr:Port: [10.0.249.6]:17138

Media Dest IP Addr:Port : [85.88.5.240]:21266

Ayodeji Okanlawon · ‎05-24-2013

Steve,

You need to look at your call routing..Somethings dotn seem to add up...

On the NA CUBE..here is the rtp flow..I assume that this call went from your NA CUBE directly to an IP Phone in germany...Please clarify

Media Source IP Addr:Port: [10.0.250.4]:27046

Media Dest IP Addr:Port : [10.0.6.30]:30010

Media Source IP Addr:Port: [10.0.1.67]:27132

Media Dest IP Addr:Port : [173.46.30.202]:20766

I assume that 173.x.x.x. is your ITSP...

ITSP(173.46.30.202)----rtp---CUBE(10.0.1.67)

CUBE(10.0.250.4)--rtp----(10.0.6.30)

From that flow we can see that CUBE in NA is sending its rtp to CUCM, again this suggests that CUBE is sending its media to MTP

For the call to the CUBE in germany...

Media Source IP Addr:Port: [10.0.249.6]:17140

Media Dest IP Addr:Port : [10.0.249.101]:22858

Media Source IP Addr:Port: [10.0.249.6]:17138

Media Dest IP Addr:Port : [85.88.5.240]:21266

ITSP(85.88.5.240)----rtp---CUBE(10.0.249.6)

CUBE(10.0.249.6)---rtp---ipphone(10.0.249.101)

Questions...

1. How were these calls placed? The call to gernamy phone..via the germany cube? Did you dial in via the germany DDI..

2. What are these IPs can you clarify. The CUBE in germany only has one interface...10.0.249.6..The ITSP is on 88.88.x.x How are you connecting to the ITSP on this ip address?

3. If you dial from NA the DDI number of the germany phone, these call should go to the NA CUBE. It should go to the germany CUBE, then to CUCM, then to the IP Phone

I need clarifications so I can understand whats going on.

It is also important to know why media for calls in NA are going to the CUCM. I bet it is DTMF related. What model of IP phones are you using..Can you also send the sh run of your NA CUBE.

Finally to get a clearer picture I am going to need the CUCM SDI traces.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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Matt Smith · ‎05-24-2013

We have a dedicated fiber connection to our ITSP in NA (ie: no WAN traffic)

*** Calling from NA to Germany (DID) ***

On the NA CUBE..here is the rtp flow..I assume that this call went from your NA CUBE directly to an IP Phone in germany...Please clarify

Media Source IP Addr:Port: [10.0.250.4]:27046 <--- NA CUBE

Media Dest IP Addr:Port : [10.0.6.30]:30010 <--- NA CUCM

Media Source IP Addr:Port: [10.0.1.67]:27132 <--- NA Router to ITSP

Media Dest IP Addr:Port : [173.46.30.202]:20766 <--- Upstream ITSP SIP server (not WAN routed)

I assume that 173.x.x.x. is your ITSP...

ITSP(173.46.30.202)----rtp---CUBE(10.0.1.67) <--- this is ITSP to Router(10.0.1.67)

CUBE(10.0.250.4)--rtp----(10.0.6.30) <--- this is CUBE to CUCM (...to my internal phone in NA)

From that flow we can see that CUBE in NA is sending its rtp to CUCM, again this suggests that CUBE is sending its media to MTP

For the call to the CUBE in germany...

Media Source IP Addr:Port: [10.0.249.6]:17140 <--- German CUBE

Media Dest IP Addr:Port : [10.0.249.101]:22858 <--- German phone

Media Source IP Addr:Port: [10.0.249.6]:17138

Media Dest IP Addr:Port : [85.88.5.240]:21266 <--- German ITSP (WAN routed)

ITSP(85.88.5.240)----rtp---CUBE(10.0.249.6)

CUBE(10.0.249.6)---rtp---ipphone(10.0.249.101)

Questions...

1. How were these calls placed? The call to gernamy phone..via the germany cube? Did you dial in via the germany DDI..

Correct, DID to DID.

Ext to Ext calls are traversing a VPN connection.

2. What are these IPs can you clarify. The CUBE in germany only has one interface...10.0.249.6..The ITSP is on 88.88.x.x How are you connecting to the ITSP on this ip address?

Correct, one interface, NAT'ing.

3. If you dial from NA the DDI number of the germany phone, these call should go to the NA CUBE. It should go to the germany CUBE, then to CUCM, then to the IP Phone

Yes, I believe this is happening, except there is no audio so the call will connect, but the call can't reach an end point (IVR).

I need clarifications so I can understand whats going on.

It is also important to know why media for calls in NA are going to the CUCM. I bet it is DTMF related. What model of IP phones are you using..Can you also send the sh run of your NA CUBE.

Media to CUCM is likely because Media Termination Point Required is checked - I can play with this this weekend when we're outside of business hours, but could this contribute to one-way audio for the German branch when we're calling DID to DID? I don't have one-way audio with any other calls to NA.

Phones:

Germany 7961

NA 7965 (currently testing with) + 7961

NA CUBE config

!

! Last configuration change at 17:02:58 UTC Wed May 15 2013 by root

version 15.2

service timestamps debug datetime msec

service timestamps log datetime msec

service password-encryption

no service dhcp

!

hostname 2951-01

!

boot-start-marker

boot-end-marker

!

logging buffered 51200 warnings

enable secret 4

!

aaa new-model

aaa local authentication attempts max-fail 3

!

aaa authentication login root local

!

aaa session-id common

!

crypto pki trustpoint TP-self-signed-2512309000

enrollment selfsigned

subject-name cn=IOS-Self-Signed-Certificate-2512309000

revocation-check none

rsakeypair TP-self-signed-2512309000

!

crypto pki certificate chain TP-self-signed-2512309000

certificate self-signed 01

quit

ip cef

!

ip dhcp excluded-address 10.10.10.1

!

no ip domain lookup

ip domain name miovision.corp

no ipv6 cef

!

multilink bundle-name authenticated

!

voice-card 0

!

voice service voip

ip address trusted list

ipv4 173.46.30.218

ipv4 173.46.30.202

ipv4 10.0.6.30

ipv4 10.0.6.31

ipv4 10.0.6.33

ipv4 10.0.6.32

ipv4 10.1.1.4

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

no call service stop

!

voice class codec 1

codec preference 1 g711ulaw

!

license udi pid CISCO2951/K9 sn FGL170810HD

hw-module pvdm 0/0

!

username root privilege 15 secret 4

username admin privilege 15 secret 4

!

redundancy

!

ip ssh time-out 60

ip ssh source-interface GigabitEthernet0/0.42

ip ssh version 2

ip ssh pubkey-chain

username root

zone security Trusted

csdb tcp synwait-time 30

csdb tcp idle-time 3600

csdb tcp finwait-time 5

csdb tcp reassembly max-memory 1024

csdb tcp reassembly max-queue-length 16

csdb udp idle-time 30

csdb icmp idle-time 10

csdb session max-session 65535

!

interface Embedded-Service-Engine0/0

no ip address

shutdown

!

interface GigabitEthernet0/0

description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$

no ip address

no ip route-cache

duplex auto

speed auto

!

interface GigabitEthernet0/0.41

encapsulation dot1Q 41

ip address 10.0.1.67 255.255.248.0

no ip route-cache

!

interface GigabitEthernet0/0.42

encapsulation dot1Q 42

ip address 10.0.250.4 255.255.255.0

zone-member security Trusted

no ip route-cache

!

interface GigabitEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface GigabitEthernet0/2

no ip address

shutdown

duplex auto

speed auto

!

ip forward-protocol nd

!

ip http server

ip http access-class 99

ip http authentication local

ip http secure-server

ip http timeout-policy idle 60 life 86400 requests 10000

!

ip route 0.0.0.0 0.0.0.0 10.0.250.1

ip route 10.0.0.0 255.255.255.0 10.0.250.1

ip route 10.0.6.0 255.255.255.0 10.0.250.1

ip route 10.0.250.0 255.255.255.0 10.0.250.1

ip route 10.1.1.0 255.255.255.0 10.0.250.1

ip route 173.46.30.0 255.255.255.0 10.0.1.65

ip route 173.46.30.202 255.255.255.255 10.0.1.65

ip route 173.46.30.218 255.255.255.255 10.0.1.65

!

access-list 23 permit 10.10.10.0 0.0.0.7

access-list 99 permit 10.0.0.0

access-list 99 permit 10.0.250.0 0.0.0.255

access-list 99 permit any

!

nls resp-timeout 1

cpd cr-id 1

!

snmp-server community cis1localkit RO 99

snmp-server enable traps entity-sensor threshold

!

control-plane

!

mgcp profile default

!

dial-peer voice 10 voip

description ** Incoming call from Rogers trunk main line/main line TF **

destination-pattern 5195132407

session protocol sipv2

session target ipv4:10.0.6.30

session transport udp

incoming called-number 5195132407

voice-class codec 1

dtmf-relay rtp-nte

dtmf-interworking standard

no vad

!

dial-peer voice 11 voip

description ** Incoming call from Rogers trunk to Marketing TF **

destination-pattern 5195149988

session protocol sipv2

session target ipv4:10.0.6.30

session transport udp

incoming called-number 5195149988

voice-class codec 1

dtmf-relay rtp-nte

dtmf-interworking standard

no vad

!

dial-peer voice 12 voip

description ** Incoming call from Rogers trunk to Support TF **

destination-pattern 5195149993

session protocol sipv2

session target ipv4:10.0.6.30

session transport udp

incoming called-number 5195149993

voice-class codec 1

dtmf-relay rtp-nte

dtmf-interworking standard

no vad

!

dial-peer voice 13 voip

description ** Incoming call from Cologne Trunk **

destination-pattern 2...

session protocol sipv2

session target ipv4:10.0.6.30

session transport udp

incoming called-number 2...

voice-class codec 1

dtmf-relay rtp-nte

dtmf-interworking standard

no vad

!

dial-peer voice 21 voip

destination-pattern 4..

session protocol sipv2

session target ipv4:10.1.1.4

session transport udp

incoming called-number 4..

voice-class codec 1

dtmf-interworking standard

no vad

!

dial-peer voice 14 voip

description ** Incoming call from Cologne Trunk **

destination-pattern 2..

session protocol sipv2

session target ipv4:10.0.6.30

session transport udp

incoming called-number 2..

voice-class codec 1

dtmf-relay rtp-nte

dtmf-interworking standard

no vad

!

dial-peer voice 20 voip

description ** Outbound call to Rogers trunk **

destination-pattern ....T

session protocol sipv2

session target ipv4:173.46.30.202

session transport udp

incoming called-number ....T

voice-class codec 1

dtmf-interworking standard

no vad

!

sip-ua

no remote-party-id

retry invite 10

retry register 10

host-registrar

!

gatekeeper

shutdown

!

line con 0

login authentication local

line aux 0

line 2

no activation-character

no exec

transport preferred none

transport input all

transport output pad telnet rlogin lapb-ta mop udptn v120 ssh

stopbits 1

line vty 0 4

access-class 99 in

exec-timeout 60 0

privilege level 15

transport input ssh

line vty 5 15

access-class 23 in

privilege level 15

transport input telnet ssh

!

scheduler allocate 20000 1000

!

end

I'm not sure how to take that trace, looking into it right now.

Ayodeji Okanlawon · ‎05-24-2013

Okay..So from your responses look like the problem you have is in germany..

Yes, I believe this is happening, except there is no audio so the call will connect, but the call can't reach an end point (IVR).

The statement above suggest that your problem is with an IVR not a phone...

More questions

1. Did you dial the IVR from a cell phone or you dialled the DID of the german IVR from a cisco phone so the call went from NA CUBE to Germany CUBE?

Successfully dial out from NA to DID of German phone. Audio in both directions.

2. Does this work when you dial an IP phone in germany directly

No issues dialing in either direction internally.

3. Does this work when you dial from a mobile to an ip phone in germany. or to the IVR

Any calls inbound to the German DID work as expected. Its only outbound calls from Germany to any destination that do not have inbound audio.

Ayodeji Okanlawon · ‎05-24-2013

Steve can you send me CUCM sdi traces (ensure the trace setting is set to detailed)

Also send me debug ccsip messages from the germany CUBE

Do an outbound test call..include the calling and called number.

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

Matt Smith · ‎05-24-2013

Really appreciate your time on this.

I'll preface this info with: the invalid IP address messages are related to our trunk provider in Germany, it has not had an affect on calls before.

2901-01.miovision.corp#debug ccsip messages
SIP Call messages tracing is enabled
2901-01.miovision.corp#terminal monitor
2901-01.miovision.corp#
*May 24 17:38:16.365: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:216.16.231.94:26664 SIP/2.0
Via: SIP/2.0/UDP 85.88.5.252:5060;branch=0
From: sip:pinger@sip.dcalling.de;tag=37146386
To: sip:216.16.231.94:26664
Call-ID: 4326ecd3-c4ed5583-3f4d1@85.88.5.252
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

*May 24 17:38:16.365: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid IP Address'
Via: SIP/2.0/UDP 85.88.5.252:5060;branch=0
From: sip:pinger@sip.dcalling.de;tag=37146386
To: sip:216.16.231.94:26664;tag=51F2AC80-1B6E
Date: Fri, 24 May 2013 17:38:16 GMT
Call-ID: 4326ecd3-c4ed5583-3f4d1@85.88.5.252
Server: Cisco-SIPGateway/IOS-15.2.4.M2
CSeq: 1 OPTIONS
Content-Length: 0

*May 24 17:38:46.541: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:216.16.231.94:26664 SIP/2.0
Via: SIP/2.0/UDP 85.88.5.252:5060;branch=0
From: sip:pinger@sip.dcalling.de;tag=8d146386
To: sip:216.16.231.94:26664
Call-ID: 4326ecd3-1bed5583-115d1@85.88.5.252
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

*May 24 17:38:46.545: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid IP Address'
Via: SIP/2.0/UDP 85.88.5.252:5060;branch=0
From: sip:pinger@sip.dcalling.de;tag=8d146386
To: sip:216.16.231.94:26664;tag=51F32260-14D8
Date: Fri, 24 May 2013 17:38:46 GMT
Call-ID: 4326ecd3-1bed5583-115d1@85.88.5.252
Server: Cisco-SIPGateway/IOS-15.2.4.M2
CSeq: 1 OPTIONS
Content-Length: 0

*May 24 17:38:50.445: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip.dcalling.de:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB4271D67
From: <>8686170390@sip.dcalling.de>;tag=51F331A0-1181
To: <>8686170390@sip.dcalling.de>
Date: Fri, 24 May 2013 17:38:50 GMT
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Max-Forwards: 70
Timestamp: 1369417130
CSeq: 45605 REGISTER
Contact: <8686170390>
Expires: 60
Supported: path
Content-Length: 0

*May 24 17:38:50.569: //23306/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB4271D67
From: <>8686170390@sip.dcalling.de>;tag=51F331A0-1181
To: <>8686170390@sip.dcalling.de>
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
CSeq: 45605 REGISTER
Server: DALASON GmbH SIP Proxy
Content-Length: 0

*May 24 17:38:50.569: //23306/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB4271D67
From: <>8686170390@sip.dcalling.de>;tag=51F331A0-1181
To: <>8686170390@sip.dcalling.de>;tag=c0b858c265d3be265092494badc524c3.3355
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
CSeq: 45605 REGISTER
WWW-Authenticate: Digest realm="sip.dcalling.de", nonce="519fa6310000912d823357785b1b1c9721abaa732c690ff0"
Server: DALASON GmbH SIP Proxy
Content-Length: 0

*May 24 17:38:50.569: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip.dcalling.de:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB428873
From: <>8686170390@sip.dcalling.de>;tag=51F331A0-1181
To: <>8686170390@sip.dcalling.de>
Date: Fri, 24 May 2013 17:38:50 GMT
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Max-Forwards: 70
Timestamp: 1369417130
CSeq: 45606 REGISTER
Contact: <8686170390>
Expires: 60
Authorization: Digest username="8686170390",realm="sip.dcalling.de",uri="sip:sip.dcalling.de:5060",response="422c59af859900af4ab56d3f3d61f460",nonce="519fa6310000912d823357785b1b1c9721abaa732c690ff0",algorithm=md5
Content-Length: 0

*May 24 17:38:50.693: //23306/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB428873
From: <>8686170390@sip.dcalling.de>;tag=51F331A0-1181
To: <>8686170390@sip.dcalling.de>
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
CSeq: 45606 REGISTER
Server: DALASON GmbH SIP Proxy
Content-Length: 0

*May 24 17:38:50.693: //23306/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB428873
From: <>8686170390@sip.dcalling.de>;tag=51F331A0-1181
To: <>8686170390@sip.dcalling.de>;tag=c0b858c265d3be265092494badc524c3.172e
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
CSeq: 45606 REGISTER
Contact: <8686170390>;q=0.5;expires=60;received="sip:216.16.231.94:26664"
Server: DALASON GmbH SIP Proxy
Content-Length: 0

*May 24 17:39:16.577: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:216.16.231.94:26664 SIP/2.0
Via: SIP/2.0/UDP 85.88.5.252:5060;branch=0
From: sip:pinger@sip.dcalling.de;tag=93246386
To: sip:216.16.231.94:26664
Call-ID: 4326ecd3-21fd5583-f25d1@85.88.5.252
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

*May 24 17:39:16.577: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid IP Address'
Via: SIP/2.0/UDP 85.88.5.252:5060;branch=0
From: sip:pinger@sip.dcalling.de;tag=93246386
To: sip:216.16.231.94:26664;tag=51F397B4-59B
Date: Fri, 24 May 2013 17:39:16 GMT
Call-ID: 4326ecd3-21fd5583-f25d1@85.88.5.252
Server: Cisco-SIPGateway/IOS-15.2.4.M2
CSeq: 1 OPTIONS
Content-Length: 0

*May 24 17:39:22.897: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0015195132407@10.0.249.6:5060 SIP/2.0
Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bKad284fdd42cb
From: "TESTING BRANCH" <500>;tag=71642~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19789738
To: <0015195132407>
Date: Fri, 24 May 2013 17:41:08 GMT
Call-ID: 1c5ed780-19f1a634-3937-1e06000a@10.0.6.30
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <10.0.6.30:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0475977600-0000065536-0000012227-0503709706
Session-Expires: 1800
P-Asserted-Identity: "TESTING BRANCH" <500>
Remote-Party-ID: "TESTING BRANCH" <500>;party=calling;screen=yes;privacy=off
Contact: <500>
Max-Forwards: 70
Content-Length: 0

*May 24 17:39:22.905: //23307/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bKad284fdd42cb
From: "TESTING BRANCH" <500>;tag=71642~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19789738
To: <0015195132407>
Date: Fri, 24 May 2013 17:39:22 GMT
Call-ID: 1c5ed780-19f1a634-3937-1e06000a@10.0.6.30
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Content-Length: 0

*May 24 17:39:23.017: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0015195132407@sip.dcalling.de:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB42971F
Remote-Party-ID: "TESTING BRANCH" <500>;party=calling;screen=yes;privacy=off
From: "TESTING BRANCH" <>8686170390@sip.dcalling.de>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>
Date: Fri, 24 May 2013 17:39:23 GMT
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0475977600-0000065536-0000012227-0503709706
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1369417163
Contact: <500>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0

*May 24 17:39:23.141: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB42971F
From: "TESTING BRANCH" <>8686170390@sip.dcalling.de>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>;tag=c0b858c265d3be265092494badc524c3.7f4c
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.dcalling.de", nonce="519fa6520000918f1307836b0bc16b901952bd5dcf08b348"
Server: DALASON GmbH SIP Proxy
Content-Length: 0

*May 24 17:39:23.141: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:0015195132407@sip.dcalling.de:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB42971F
From: "TESTING BRANCH" <500>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>;tag=c0b858c265d3be265092494badc524c3.7f4c
Date: Fri, 24 May 2013 17:39:23 GMT
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

*May 24 17:39:23.141: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0015195132407@sip.dcalling.de:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB42A1BE
Remote-Party-ID: "TESTING BRANCH" <500>;party=calling;screen=yes;privacy=off
From: "TESTING BRANCH" <>8686170390@sip.dcalling.de>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>
Date: Fri, 24 May 2013 17:39:23 GMT
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0475977600-0000065536-0000012227-0503709706
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1369417163
Contact: <500>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="8686170390",realm="sip.dcalling.de",uri="sip:0015195132407@sip.dcalling.de:5060",response="5bc57ea0e5b98e15c7f2c4e1deeab02a",nonce="519fa6520000918f1307836b0bc16b901952bd5dcf08b348",algorithm=md5
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0

*May 24 17:39:23.269: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB42A1BE
From: "TESTING BRANCH" <>8686170390@sip.dcalling.de>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
CSeq: 102 INVITE
Server: DALASON GmbH SIP Proxy
Content-Length: 0

*May 24 17:39:23.601: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.249.6:5060;rport=26664;received=216.16.231.94;branch=z9hG4bKB42A1BE
Record-Route: <85.88.5.252>
From: "TESTING BRANCH" <>8686170390@sip.dcalling.de>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>;tag=as51c2f9be
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
CSeq: 102 INVITE
Server: DCALLING PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <0015195132407>
Content-Type: application/sdp
Content-Length: 308

v=0
o=root 1794050140 1794050140 IN IP4 85.88.5.233
s=Asterisk PBX 1.6.2.6
c=IN IP4 85.88.5.240
t=0 0
m=audio 21984 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

*May 24 17:39:23.613: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bKad284fdd42cb
From: "TESTING BRANCH" <500>;tag=71642~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19789738
To: <0015195132407>;tag=51F3B328-E30
Date: Fri, 24 May 2013 17:39:22 GMT
Call-ID: 1c5ed780-19f1a634-3937-1e06000a@10.0.6.30
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <0015195132407>;party=called;screen=no;privacy=off
Contact: <0015195132407>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 307

v=0
o=root 1794050140 1794050140 IN IP4 85.88.5.233
s=Asterisk PBX 1.6.2.6
c=IN IP4 10.0.249.6
t=0 0
m=audio 17172 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

*May 24 17:39:23.669: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0015195132407@10.0.249.6:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bKad2937dc17c7
From: "TESTING BRANCH" <500>;tag=71642~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19789738
To: <0015195132407>;tag=51F3B328-E30
Date: Fri, 24 May 2013 17:41:08 GMT
Call-ID: 1c5ed780-19f1a634-3937-1e06000a@10.0.6.30
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 233

v=0
o=CiscoSystemsCCM-SIP 71642 1 IN IP4 10.0.6.30
s=SIP Call
c=IN IP4 10.0.249.101
b=TIAS:64000
b=AS:64
t=0 0
m=audio 17576 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

*May 24 17:39:23.677: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:0015195132407@85.88.5.233 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB42B18F5
From: "TESTING BRANCH" <500>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>;tag=as51c2f9be
Date: Fri, 24 May 2013 17:39:23 GMT
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
Route: <85.88.5.252>
Max-Forwards: 70
CSeq: 102 ACK
Proxy-Authorization: Digest username="8686170390",realm="sip.dcalling.de",uri="sip:0015195132407@sip.dcalling.de:5060",response="5bc57ea0e5b98e15c7f2c4e1deeab02a",nonce="519fa6520000918f1307836b0bc16b901952bd5dcf08b348",algorithm=md5
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 231

v=0
o=CiscoSystemsCCM-SIP 71642 1 IN IP4 10.0.6.30
s=SIP Call
c=IN IP4 10.0.249.6
b=TIAS:64000
b=AS:64
t=0 0
m=audio 17170 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

*May 24 17:39:44.149: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:0015195132407@10.0.249.6:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bKad2a27e009b9
From: "TESTING BRANCH" <500>;tag=71642~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19789738
To: <0015195132407>;tag=51F3B328-E30
Date: Fri, 24 May 2013 17:41:08 GMT
Call-ID: 1c5ed780-19f1a634-3937-1e06000a@10.0.6.30
User-Agent: Cisco-CUCM9.1
Max-Forwards: 70
P-Asserted-Identity: "TESTING BRANCH" <500>
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0

*May 24 17:39:44.153: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bKad2a27e009b9
From: "TESTING BRANCH" <500>;tag=71642~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19789738
To: <0015195132407>;tag=51F3B328-E30
Date: Fri, 24 May 2013 17:39:44 GMT
Call-ID: 1c5ed780-19f1a634-3937-1e06000a@10.0.6.30
Server: Cisco-SIPGateway/IOS-15.2.4.M2
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=0,OS=0,PR=1023,OR=175956,PL=0,JI=0,LA=0,DU=20
Content-Length: 0

*May 24 17:39:44.153: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:0015195132407@85.88.5.233 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB42C17D0
From: "TESTING BRANCH" <500>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>;tag=as51c2f9be
Date: Fri, 24 May 2013 17:39:23 GMT
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Max-Forwards: 70
Route: <85.88.5.252>
Timestamp: 1369417184
CSeq: 103 BYE
Proxy-Authorization: Digest username="8686170390",realm="sip.dcalling.de",uri="sip:0015195132407@85.88.5.233",response="8d2c6d97ccbddaa1e012b0caee51832d",nonce="519fa6520000918f1307836b0bc16b901952bd5dcf08b348",algorithm=md5
Reason: Q.850;cause=16
P-RTP-Stat: PS=1023,OS=163680,PR=0,OR=0,PL=0,JI=0,LA=0,DU=20
Content-Length: 0

*May 24 17:39:44.653: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:0015195132407@85.88.5.233 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB42C17D0
From: "TESTING BRANCH" <500>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>;tag=as51c2f9be
Date: Fri, 24 May 2013 17:39:23 GMT
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Max-Forwards: 70
Route: <85.88.5.252>
Timestamp: 1369417184
CSeq: 103 BYE
Proxy-Authorization: Digest username="8686170390",realm="sip.dcalling.de",uri="sip:0015195132407@85.88.5.233",response="8d2c6d97ccbddaa1e012b0caee51832d",nonce="519fa6520000918f1307836b0bc16b901952bd5dcf08b348",algorithm=md5
Reason: Q.850;cause=16
P-RTP-Stat: PS=1023,OS=163680,PR=0,OR=0,PL=0,JI=0,LA=0,DU=20
Content-Length: 0

*May 24 17:39:44.781: //23308/1C5ED7800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.249.6:5060;rport=26664;received=216.16.231.94;branch=z9hG4bKB42C17D0
Record-Route: <85.88.5.252>
From: "TESTING BRANCH" <500>;tag=51F3B0DC-1F3A
To: <>0015195132407@sip.dcalling.de>;tag=as51c2f9be
Call-ID: B488E039-C3CF11E2-B856E3B7-54AA1348@10.0.249.6
CSeq: 103 BYE
Server: DCALLING PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

*May 24 17:39:46.213: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:216.16.231.94:26664 SIP/2.0
Via: SIP/2.0/UDP 85.88.5.252:5060;branch=0
From: sip:pinger@sip.dcalling.de;tag=c9246386
To: sip:216.16.231.94:26664
Call-ID: 4326ecd3-57fd5583-d45d1@85.88.5.252
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

*May 24 17:39:46.217: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid IP Address'
Via: SIP/2.0/UDP 85.88.5.252:5060;branch=0
From: sip:pinger@sip.dcalling.de;tag=c9246386
To: sip:216.16.231.94:26664;tag=51F40B78-65B
Date: Fri, 24 May 2013 17:39:46 GMT
Call-ID: 4326ecd3-57fd5583-d45d1@85.88.5.252
Server: Cisco-SIPGateway/IOS-15.2.4.M2
CSeq: 1 OPTIONS
Content-Length: 0

*May 24 17:39:50.693: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip.dcalling.de:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB42D13D9
From: <>8686170390@sip.dcalling.de>;tag=51F41CF8-BC1
To: <>8686170390@sip.dcalling.de>
Date: Fri, 24 May 2013 17:39:50 GMT
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Max-Forwards: 70
Timestamp: 1369417190
CSeq: 45607 REGISTER
Contact: <8686170390>
Expires: 60
Supported: path
Content-Length: 0

*May 24 17:39:50.817: //23309/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB42D13D9
From: <>8686170390@sip.dcalling.de>;tag=51F41CF8-BC1
To: <>8686170390@sip.dcalling.de>
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
CSeq: 45607 REGISTER
Server: DALASON GmbH SIP Proxy
Content-Length: 0

*May 24 17:39:50.817: //23309/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB42D13D9
From: <>8686170390@sip.dcalling.de>;tag=51F41CF8-BC1
To: <>8686170390@sip.dcalling.de>;tag=c0b858c265d3be265092494badc524c3.bdb7
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
CSeq: 45607 REGISTER
WWW-Authenticate: Digest realm="sip.dcalling.de", nonce="519fa66d000091d5b01b77da7ab436d0fb7dbfd83d5d60e9"
Server: DALASON GmbH SIP Proxy
Content-Length: 0

*May 24 17:39:50.817: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:sip.dcalling.de:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.6:5060;branch=z9hG4bKB42E254E
From: <>8686170390@sip.dcalling.de>;tag=51F41CF8-BC1
To: <>8686170390@sip.dcalling.de>
Date: Fri, 24 May 2013 17:39:50 GMT
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Max-Forwards: 70
Timestamp: 1369417190
CSeq: 45608 REGISTER
Contact: <8686170390>
Expires: 60
Authorization: Digest username="8686170390",realm="sip.dcalling.de",uri="sip:sip.dcalling.de:5060",response="c8eb4da55fd75388e9297c01c8dce65f",nonce="519fa66d000091d5b01b77da7ab436d0fb7dbfd83d5d60e9",algorithm=md5
Content-Length: 0

*May 24 17:39:50.941: //23309/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB42E254E
From: <>8686170390@sip.dcalling.de>;tag=51F41CF8-BC1
To: <>8686170390@sip.dcalling.de>
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
CSeq: 45608 REGISTER
Server: DALASON GmbH SIP Proxy
Content-Length: 0

*May 24 17:39:50.941: //23309/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.249.6:5060;received=216.16.231.94;rport=26664;branch=z9hG4bKB42E254E
From: <>8686170390@sip.dcalling.de>;tag=51F41CF8-BC1
To: <>8686170390@sip.dcalling.de>;tag=c0b858c265d3be265092494badc524c3.4aac
Call-ID: 8BA784B2-B74E11E2-8002E3B7-54AA1348
CSeq: 45608 REGISTER
Contact: <8686170390>;q=0.5;expires=60;received="sip:216.16.231.94:26664"
Server: DALASON GmbH SIP Proxy
Content-Length: 0

2901-01.miovision.corp#terminal no monitor
2901-01.miovision.corp#

Trace:

https://dl.dropboxusercontent.com/u/21916057/calllogs_00000019.txt.gz

https://dl.dropboxusercontent.com/u/21916057/SDL001_100_001156.txt.gz

Ayodeji Okanlawon · ‎05-24-2013

From the logs I can see that you are doing DO to DO. That implies that you are not offering any SDP to your ITSP for outbound calls...

Configure this and test again on the german cube

conf t

voice service voip

sip

early-offer forced

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

Matt Smith · ‎05-25-2013

Ok, so early-offer forced was already enabled.

I also had pass-thru content sdp enabled. And once that was removed, I have two-way audio! Appreciate you getting me on the right track with SDP.

Back to the topic of MTP's:

On CUCM's trunk configuration for the NA CUBE I've unchecked "Media Termination Point Required". I've checked the voice service voip|sip settings and they have early-offer forced as well, and no sdp pass-through but I have one issue:

Inbound calls have a very short life. I can call in (from my German trunk using NA DID) get the IVR, enter an extension, and the call will briefly display on my phone before it drops. If I call in from my cell phone to NA DID, I can get through the IVR, to my extension and connect but have no audio (connection stays up).

I'm guessing the German trunk provider isn't seeing any media so it drops the connection. I recieve this message on the Germany CUBE:

*May 25 15:53:10.871: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:500@216.16.231.94:39019 SIP/2.0

Record-Route: <85.88.5.252>

Via: SIP/2.0/UDP 85.88.5.252;branch=z9hG4bK1d75.6aa30751.0

Via: SIP/2.0/UDP 85.88.5.233:5060;received=85.88.5.233;branch=z9hG4bK21064c67;rport=5060

Max-Forwards: 69

From: <>0015195132407@sip.dcalling.de>;tag=as476855bd

To: "TESTING BRANCH" <>8686170390@sip.dcalling.de>;tag=56B8A2F4-2F5

Call-ID: 1C15D66-C48A11E2-8345E3B7-54AA1348@10.0.249.6

CSeq: 102 BYE

User-Agent: DCALLING PBX

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

Content-Length: 0

This seems like a NA CUBE -> IP phone issue now, seeing as I can get audio from the IVR, and send DTMF to interact with it.