Solved: Problem Configuring Incoming SIP calls CUCM7

nanosynth · ‎11-23-2012

After self teaching myself and help from others here, I have learned CME 4.1 & Unity Express, I bought a server and loaded CUCM7 and Unity on it. This is my first crack at Publisher, Subscriber and Unity. So far I have gotten two 7912 phones to call eachother and setup the voicmail on each. I have the server LAN'd to a 2621XM acting as an h323 gateway. I have two SIP numbers comming into it from the internet. I have successfully made calls from each extension out to the world. I can't call into the CUCM7, get fast busy. Just need to be pointed in the right direction and I will figure out the rest. My incoming dial peer, don't know if it's correct for CUCM7. Phones are SCCP.

dial-peer voice 1 voip

description To/From CUCM

destination-pattern .T

voice-class codec 1

session protocol sipv2

session target ipv4:10.10.210.10 <--- IP of CUCM7

incoming called-number .%

dtmf-relay rtp-nte

no vad

Debug CCSIP Messages from 2621XM when call placed to CUCM7

INVITE sip:17275652954@10.10.210.10:5060 SIP/2.0

Via: SIP/2.0/UDP 10.10.210.1:5060;branch=z9hG4bKA32017

From: <sip:17272026330@216.115.69.144>;tag=439558-E40

To: <sip:17275652954@10.10.210.10>

Date: Fri, 23 Nov 2012 22:43:53 GMT

Call-ID: 1758E39F-34F611E2-8102E0F1-87FDF4FE@sip.flowroute.com:5060

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 390061477-888541666-2164121841-2281567486

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, R

CSeq: 101 INVITE

Timestamp: 1353710633

Contact: <sip:17272026330@10.10.210.1:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 94

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 278

v=0

o=CiscoSystemsSIP-GW-UserAgent 5826 170 IN IP4 10.10.210.1

s=SIP Call

c=IN IP4 10.10.210.1

t=0 0

m=audio 17608 RTP/AVP 0 18 101

c=IN IP4 10.10.210.1

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

Nov 23 22:43:53.315: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Date: Fri, 23 Nov 2012 22:43:53 GMT

From: <sip:17272026330@216.115.69.144>;tag=439558-E40

Allow-Events: presence

Content-Length: 0

To: <sip:17275652954@10.10.210.10>

Call-ID: 1758E39F-34F611E2-8102E0F1-87FDF4FE@sip.flowroute.com:5060

Via: SIP/2.0/UDP 10.10.210.1:5060;branch=z9hG4bKA32017

CSeq: 101 INVITE

Nov 23 22:43:53.319: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 503 Service Unavailable

Date: Fri, 23 Nov 2012 22:43:53 GMT

Warning: 399 "Routing failed: ccbid=23 socket=10.10.210.1:5060"

From: <sip:17272026330@216.115.69.144>;tag=439558-E40

Allow-Events: presence

Content-Length: 0

To: <sip:17275652954@10.10.210.10>;tag=941901863

Call-ID: 1758E39F-34F611E2-8102E0F1-87FDF4FE@sip.flowroute.com:5060

Via: SIP/2.0/UDP 10.10.210.1:5060;branch=z9hG4bKA32017

CSeq: 101 INVITE

Of course I see the routing failed. What I don't understand is how to setup the phone in CUCM7 so that when it's called from the outside world, it recognizes the SIP message to ring it. In regular CME 4.1 my incoming dial peer setup was this, and it worked.

dial-peer voice 2 voip

destination-pattern .T

redirect ip2ip

voice-class codec 1

voice-class sip localhost dns:sip.flowroute.com

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target dns:sip.flowroute.com

incoming called-number .%

dtmf-relay rtp-nte

no vad

I believe in CME4.1, how the phone knew to ring when called was this:

ephone-dn 1

number 17275652954 secondary 3001 no-reg both

How do I have to identify the 7912 phone in CUCM7 so it can respond to an incomming call, and do I need to modify my incoming dial peer?

Thank YOU!

Ayodeji Okanlawon · ‎11-24-2012

Can you send your full configuration..Send us a show run.

Also It looks like you have a h323 to sip setup

CUCM-------h323----->Gateway-------SIP------>ITSP

If this is correct then you need to make config changes...The dial-peer that sends call to your CUCM is dial-peer 1 and its configured to use SIP...Can you configure a second dial-peer as follows

dial-peer voice 1 voip

destination-pattern 17275652954

session target ipv4:10.10.210.10

no vad

dtmf-relay h245-alphanumeric

dial-peer voice 1 voip

translation-profile incoming PSTN-IN

incoming called-number 17275652954

session protocol sipv2

voice-class codec 1

no vad

dtmf-relay rtp-nte

dial-peer voice 3 voip

destination-pattern 3001

session target ipv4:10.10.210.10

no vad

dtmf-relay h245-alphanumeric

Now you will need to create a translation rule to change 17275652954 to 3001 (assuming 3001 is the internal extension in cucm)

voice translation-rule 1

rule 1 /^17275652954/ /3001/

voice translation-profile PSTN-IN

translate called 1

So this is what this config does...

1. Incoming calls to Number 17275652954 is mapped to a sip leg/sip dial-peer

2. A translation prifle is applied to this leg and the called number is changed to 3001

3. The extension 3001 is then routed to cucm on a h323 leg

You also need to ensure that you have CUBE functionality enabled..because by default the gateway is not going to route incoming voip leg to outgoing voip leg...

voice service voip

allow-connections h323 to sip

allow-connections sip to h323

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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View solution in original post

will.alvord · ‎11-24-2012

It depends on how (or where) you want to handle the digit manipulation. You can do it on the gateway as described or on cucm. Just so you have both options -- under the 323 trunk incoming settings there's a digit length setting. Adjust or leave as is but then create a translation pattern with that length of your DID as the pattern with a mask of your extension.

There are multiple ways to skin the cat so to speak. It mostly comes down to where you're most comfortable - ios or cucm.

Sent from Cisco Technical Support Android App

View solution in original post

Ayodeji Okanlawon · ‎11-24-2012

Usually one way voice is a routing issue...But I had like us to do something first. Lets create a sip end to end connection.

So we will change your call flow to this:

CUCM--------sip-------->gateway--------sip-------->ITSP

To do this, you need to create a sip trunk to your gateway. You then do the ff:

1. You need to creata an inbound dial-peer for both cucm and sip provider to use sip as follows

dial-peer voice 1 voip

incoming called number .

session protocol sipv2

dtmf-relay rtp-nte

no vad

This dial-peer will be used to route calls from both sip provider and cucm (inbound to the gateway)

2. You then need to change the dial-peer that route calls to cucm from h323 to sip

dial-peer voice 4 voip

session protocol sipv2

destination-pattern 1001

session target ipv4:10.10.210.10

dtmf-relay rtp-nte

no vad

3. You need to enable sip to sip calls on the gateway

voice service voip

allow-connections sip to sip

Now do a test call..let me know what the result is

send debug ccsip messages

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

View solution in original post

will.alvord · ‎11-23-2012

Way to go! Great initiative.

Within cucm you'll need to add a connection to your gateway be it an h.323 gateway or sip trunk. You could go with an mgcp gateway, but it looks like you're looking to learn gateway and cucm configs so you're left with the other 2. Within either set of configs you'll see an inbound routing section. Normally, you'd provision the lines of your phones in a partition. Then you'd set the (h323 gateway or sip trunk) inbound css to a css containing that partition.

That should get you headed in the right direction.

will

nanosynth · ‎11-24-2012

I do have a connection to my gatway from CUCM and it is h323. I have outbound numbers mapped in that partition as well. Thats how I am able to dial out to the world on the two 7912 phones. I have only one CSS and one parititon in that CSS. Very basic. The phones are in that one partition in that one CSS. SInce my phones are in that partition with extension numbers 1001 and 1002, shouldn't an incoming dial peer in the 2621 written like this ring the one extension, or am I still missing something, like do I need to do incoming trranslations in the 2621XM to translate my incoming SIP number of 17275652954 to my extension number of 1001?

dial-peer voice 1 voip

description To CUCM

destination-pattern 1001

voice-class codec 1

session protocol sipv2

session target ipv4:10.10.210.10 <--- IP of CUCM7

incoming called-number 17275652954

dtmf-relay rtp-nte

no vad

Ayodeji Okanlawon · ‎11-24-2012

Can you send your full configuration..Send us a show run.

Also It looks like you have a h323 to sip setup

CUCM-------h323----->Gateway-------SIP------>ITSP

If this is correct then you need to make config changes...The dial-peer that sends call to your CUCM is dial-peer 1 and its configured to use SIP...Can you configure a second dial-peer as follows

dial-peer voice 1 voip

destination-pattern 17275652954

session target ipv4:10.10.210.10

no vad

dtmf-relay h245-alphanumeric

dial-peer voice 1 voip

translation-profile incoming PSTN-IN

incoming called-number 17275652954

session protocol sipv2

voice-class codec 1

no vad

dtmf-relay rtp-nte

dial-peer voice 3 voip

destination-pattern 3001

session target ipv4:10.10.210.10

no vad

dtmf-relay h245-alphanumeric

Now you will need to create a translation rule to change 17275652954 to 3001 (assuming 3001 is the internal extension in cucm)

voice translation-rule 1

rule 1 /^17275652954/ /3001/

voice translation-profile PSTN-IN

translate called 1

So this is what this config does...

1. Incoming calls to Number 17275652954 is mapped to a sip leg/sip dial-peer

2. A translation prifle is applied to this leg and the called number is changed to 3001

3. The extension 3001 is then routed to cucm on a h323 leg

You also need to ensure that you have CUBE functionality enabled..because by default the gateway is not going to route incoming voip leg to outgoing voip leg...

voice service voip

allow-connections h323 to sip

allow-connections sip to h323

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

nanosynth · ‎11-24-2012

Thank you very much aokanlowon. The light came on when you said I had the setup of CUCM-->h323-->SIP-->ITSP and not the other way around. This is my config below, with the second two of your dial peers. The first dial peer listed messed up call procesing. Everything works now, but had a bug. When I called the CUCM from a landline, through my SIP provider to my CUCM, it rang the 7912 and when I answered the pots line kept ringing in my ear. I fixed that by enabling fast start incoming on the gateway CUCM page. Phones can call eachother now.

Still have two issues. When I use the landline to call into the CUCM 7912 phone, I cannot hear any speech on the landline phone when talking into the CUCM 7912 phone. If I speak into the landline phone, speech comes out of the 7912, so I have a one way audio issue here. No audio from CUCM to landline phone.

The DTMF tones do not pass to either phone. Of course I can't hear anything on the landline phone, so I don't know if the DTMF tones generated from the CUCM 7912 would make it though.

hostname CUCM_GATEWAY

!

boot-start-marker

boot-end-marker

!

no logging console

enable secret 5 $1$uNta$f23yMXIpygHlTVta8z7Jz.

!

no aaa new-model

clock timezone EST -5

no network-clock-participate slot 1

no network-clock-participate wic 0

ip cef

!

no ip dhcp use vrf connected

ip dhcp excluded-address 192.168.20.1 192.168.20.10

!

ip dhcp pool VOICE_20

network 192.168.20.0 255.255.255.0

default-router 192.168.20.1

option 150 ip 10.10.210.10

!

no ip domain lookup

ip name-server 208.67.222.222

ip name-server 208.67.220.220

multilink bundle-name authenticated

!

voice service voip

allow-connections h323 to sip

allow-connections sip to h323

h323

sip

registrar server expires max 160 min 160

localhost dns:sip.flowroute.com:5060

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

!

voice translation-rule 1

rule 1 /^9$..........$$/ /1\1/

!

voice translation-rule 2

rule 2 /^17275652954/ /1001/

!

voice translation-profile PSTN-IN

translate called 2

!

voice translation-profile outgoing

translate called 1

!

archive

log config

hidekeys

!

interface FastEthernet0/0

description To CUCM

ip address 10.10.210.1 255.255.255.0

duplex auto

speed auto

ntp broadcast

h323-gateway voip interface

h323-gateway voip h323-id HVoiceGW1

h323-gateway voip bind srcaddr 10.10.210.1

!

interface Ethernet1/0

description Telnet Management

ip address 192.168.2.115 255.255.255.0

half-duplex

!

interface Ethernet1/1

description Ethernet To Internet Home Router

ip address 192.168.1.100 255.255.255.0

half-duplex

!

interface Ethernet1/2

description To Phones

ip address 192.168.20.1 255.255.255.0

half-duplex

!

interface Ethernet1/3

no ip address

shutdown

half-duplex

!

ip forward-protocol nd

ip route 0.0.0.0 0.0.0.0 192.168.1.1

ip route 10.0.0.0 255.0.0.0 10.10.210.10

!

ip http server

no ip http secure-server

!

control-plane

!

dial-peer voice 2 voip

description Outbound To Flowroute

translation-profile outgoing outgoing

destination-pattern 9..........

voice-class codec 1

session protocol sipv2

session target ipv4:216.115.69.144

no vad

!

dial-peer voice 3 voip

translation-profile incoming PSTN-IN

voice-class codec 1

session protocol sipv2

incoming called-number 17275652954

dtmf-relay rtp-nte

no vad

!

dial-peer voice 4 voip

destination-pattern 1001

session target ipv4:10.10.210.10

dtmf-relay h245-alphanumeric

no vad

!

sip-ua

credentials username XXXX password XXXX realm sip.flowroute.com

authentication username XXXX password 7 XXXX

no remote-party-id

retry invite 2

retry register 10

timers connect 100

registrar ipv4:216.115.69.144 expires 60

connection-reuse

host-registrar

!

Ayodeji Okanlawon · ‎11-24-2012

Usually one way voice is a routing issue...But I had like us to do something first. Lets create a sip end to end connection.

So we will change your call flow to this:

CUCM--------sip-------->gateway--------sip-------->ITSP

To do this, you need to create a sip trunk to your gateway. You then do the ff:

1. You need to creata an inbound dial-peer for both cucm and sip provider to use sip as follows

dial-peer voice 1 voip

incoming called number .

session protocol sipv2

dtmf-relay rtp-nte

no vad

This dial-peer will be used to route calls from both sip provider and cucm (inbound to the gateway)

2. You then need to change the dial-peer that route calls to cucm from h323 to sip

dial-peer voice 4 voip

session protocol sipv2

destination-pattern 1001

session target ipv4:10.10.210.10

dtmf-relay rtp-nte

no vad

3. You need to enable sip to sip calls on the gateway

voice service voip

allow-connections sip to sip

Now do a test call..let me know what the result is

send debug ccsip messages

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

nanosynth · ‎11-24-2012

Ok, I completed what you asked. I am getting more proficient at this.. What we have now is the exact opposite of before. When I call the landline phone from the 7912, I get no voice on the landline phone but if I speak into it I can get voice on the 7912. I must confess the landline phone is a Brighthouse VIOP phone. Only one time out of like 10 test calls did it work voice both ways and DTMF both ways. It's not really that stable. Here is the ccsip's

SIP Call messages tracing is enabled

CUCM_GATEWAY#

Nov 24 19:32:06.351: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:17272026330@216.115.69.144:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK3181F08

From: <17275652954>;tag=124477C-1459

To: <17272026330>

Date: Sat, 24 Nov 2012 19:32:06 GMT

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 2151945376-3055554827-167780865-3232240652

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1353785526

Contact: <17275652954>

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

Nov 24 19:32:06.439: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK3181F08

From: <17275652954>;tag=124477C-1459

To: <17272026330>

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

CSeq: 101 INVITE

Content-Length: 0

Nov 24 19:32:06.443: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK3181F08

From: <17275652954>;tag=124477C-1459

To: <17272026330>;tag=da160a8e0534d7b0db4afd8620a22cb7.639f

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="ULEh4lCxILbi74Mhp2kX1LUm3Y3v/XPw", qop=

"auth"

Content-Length: 0

Nov 24 19:32:06.459: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:17272026330@216.115.69.144:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK3181F08

From: <17275652954>;tag=124477C-1459

To: <17272026330>;tag=da160a8e0534d7b0db4afd8620a22cb7.639f

Date: Sat, 24 Nov 2012 19:32:06 GMT

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Nov 24 19:32:06.463: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:17272026330@216.115.69.144:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK31914DF

From: <17275652954>;tag=124477C-1459

To: <17272026330>

Date: Sat, 24 Nov 2012 19:32:06 GMT

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 2151945376-3055554827-167780865-3232240652

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 102 INVITE

Max-Forwards: 70

Timestamp: 1353785526

Contact: <17275652954>

Expires: 180

Allow-Events: telephone-event

Proxy-Authorization: Digest username="86148921",realm="sip.flowroute.com",uri="sip:17272026330@216.1

15.69.144:5060",response="fc9b94fdefc76843bcd8b5e79c9e88bb",nonce="ULEh4lCxILbi74Mhp2kX1LUm3Y3v/XPw"

,cnonce="ABC8CB6C",qop="auth",algorithm=md5,nc=00000001

Content-Length: 0

Nov 24 19:32:06.939: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK31914DF

From: <17275652954>;tag=124477C-1459

To: <17272026330>

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

CSeq: 102 INVITE

Content-Length: 0

Nov 24 19:32:09.375: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

From: <17275652954>;tag=124477C-1459

To: <17272026330>;tag=gK0b851454

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK31914DF

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

CSeq: 102 INVITE

Record-Route: <216.115.69.133:5060>

Record-Route: <216.115.69.144:5060>

Contact: <>

Content-Length: 0

Nov 24 19:32:12.123: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

From: <17275652954>;tag=124477C-1459

To: <17272026330>;tag=gK0b851454

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK31914DF

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

CSeq: 102 INVITE

Record-Route: <216.115.69.133:5060>

Record-Route: <216.115.69.144:5060>

Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixe

d

Contact: <>

Content-Length: 216

Content-Type: application/sdp

v=0

o=- 30309 24983 IN IP4 4.55.22.66

s=-

c=IN IP4 4.55.22.66

t=0 0

m=audio 26144 RTP/AVP 0 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=maxptime:20

Nov 24 19:32:12.211: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:+17272026330@4.55.22.99:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK31A20C9

From: <17275652954>;tag=124477C-1459

To: <17272026330>;tag=gK0b851454

Date: Sat, 24 Nov 2012 19:32:06 GMT

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

Route: <216.115.69.144:5060>,<216.115.69.133:5060>

Max-Forwards: 70

CSeq: 102 ACK

Proxy-Authorization: Digest username="86148921",realm="sip.flowroute.com",uri="sip:17272026330@216.1

15.69.144:5060",response="fc9b94fdefc76843bcd8b5e79c9e88bb",nonce="ULEh4lCxILbi74Mhp2kX1LUm3Y3v/XPw"

,cnonce="ABC8CB6C",qop="auth",algorithm=md5,nc=00000001

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 181

v=0

o=CiscoSystemsSIP-GW-UserAgent 6017 638 IN IP4 192.168.1.100

s=SIP Call

c=IN IP4 192.168.1.100

t=0 0

m=audio 16406 RTP/AVP 0

c=IN IP4 192.168.1.100

a=rtpmap:0 PCMU/8000

Nov 24 19:32:16.835: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

OPTIONS sip:192.168.1.100:5060 SIP/2.0

Max-Forwards: 10

Record-Route: <216.115.69.144>

Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK0631.be4607b649049cd3002df07ca6a8fcca.0

Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0

Route: <216.115.69.144>

From: sip:ping@invalid;tag=42115a45

To: sip:192.168.0.3:5060

Call-ID: 4861c687-acf89827-0fb2be3@216.115.69.131

CSeq: 1 OPTIONS

Content-Length: 0

Nov 24 19:32:16.855: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK0631.be4607b649049cd3002df07ca6a8fcca.0,SIP/2.0/UDP 21

6.115.69.131:5060;branch=0

From: sip:ping@invalid;tag=42115a45

To: sip:192.168.0.3:5060;tag=1247080-227A

Date: Sat, 24 Nov 2012 19:32:16 GMT

Call-ID: 4861c687-acf89827-0fb2be3@216.115.69.131

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 OPTIONS

Supported: 100rel,resource-priority,replaces

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Accept: application/sdp

Content-Type: application/sdp

Content-Length: 170

v=0

o=CiscoSystemsSIP-GW-UserAgent 4536 3362 IN IP4 192.168.1.100

s=SIP Call

c=IN IP4 192.168.1.100

t=0 0

m=audio 0 RTP/AVP 18 0 8 4 2 15 3

c=IN IP4 192.168.1.100

Nov 24 19:32:17.091: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:+17272026330@4.55.22.99:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK31B9E2

From: <17275652954>;tag=124477C-1459

To: <17272026330>;tag=gK0b851454

Date: Sat, 24 Nov 2012 19:32:06 GMT

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Route: <216.115.69.144:5060>,<216.115.69.133:5060>

Timestamp: 1353785537

CSeq: 103 BYE

Proxy-Authorization: Digest username="86148921",realm="sip.flowroute.com",uri="sip:+17272026330@4.55

.22.99:5060",response="9227cec4f478fe09c64d0f97292db6e5",nonce="ULEh4lCxILbi74Mhp2kX1LUm3Y3v/XPw",cn

once="DEBF9368",qop="auth",algorithm=md5,nc=00000002

Reason: Q.850;cause=16

Content-Length: 0

Nov 24 19:32:17.219: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

From: <17275652954>;tag=124477C-1459

To: <17272026330>;tag=gK0b851454

Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK31B9E2

Call-ID: 7716E3BC-35A411E2-83E6BCE8-24D7501A@sip.flowroute.com:5060

CSeq: 103 BYE

Record-Route: <216.115.69.133:5060>

Record-Route: <216.115.69.144:5060>

Content-Length: 0

Ayodeji Okanlawon · ‎11-24-2012

Before troublshooting further, do you want to try another phone?

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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nanosynth · ‎11-24-2012

Funny you should ask that. I tried my Metro PCS cellphone. When I call into the CUCM from the cell phone, the call is perfect and full DTMF both ways. When I call the cell phone from the CUCM, it rings, I answer it and then I get a fast busy on the CUCM 7912. Weird.. I wish I had a POTS line to start off with. I have to work tonight so I am going to research this for 12 hours at work.

Ayodeji Okanlawon · ‎11-24-2012

Do a test call for the non working call..Please send the full debug ccsip messages. I need the debug to show the cucm side of the call please

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

nanosynth · ‎11-25-2012

Hi aokanlawon. Well, after woking 13 hours last ngiht in the NOC where I work at, I read ,researched, studied and created configs for my issues in 12.5 of those hours. What I came up with was a 100% SIP to SIP gateway configed in my 2621XM. The part I really had to learn was how to configure (correctly) the inbound and outbound SIP configs in the CUCM7. I got it all working rock solid. Calls in and out to both the Brighthouse VOIP phone and my Metro cell phone. No hangups, fast busys, one way audio...even the DTMF works both ways! I learned a lot from reading ccsip debugs too, probably cause you were asking form them all the time..haha..Well, thank you very much for all your help!

Ayodeji Okanlawon · ‎11-26-2012

I must say what a great job you did! Well done! Thats the way to learn. You may want to shar your findings with others on the forum. Its the way we give back to this excellent forum, so that when others come across your thread, they can also benefit.

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

will.alvord · ‎11-24-2012

It depends on how (or where) you want to handle the digit manipulation. You can do it on the gateway as described or on cucm. Just so you have both options -- under the 323 trunk incoming settings there's a digit length setting. Adjust or leave as is but then create a translation pattern with that length of your DID as the pattern with a mask of your extension.

There are multiple ways to skin the cat so to speak. It mostly comes down to where you're most comfortable - ios or cucm.

Sent from Cisco Technical Support Android App

nanosynth · ‎11-24-2012

I hear what your saying. I learned right from IOS and I am halfway through CCNA class (Not CCNA Voice, hence my questions here) now, so I have a good IOS base, but I'm learning the CUCM GUI. It's got a lot to offer..