UC500 SIP Trunk

[davidfield]

Hello All,

Apologies in advance for the long winded post. helping with the config woudl be awesome, but I'm not expecitng that and some concise links would be aprreciated.

I've been asked to setup a SIP trunk on the office UC540.   Using CCA I thought it would not be a problem, but suffice it to say that when I applied the settings through CCA it broke a lot of stuff that one of the other guys had been doing through CLI - He has left so need to work out on my own.

In summary we have a SIP account with a provider and I've been doing a lot of reading but confusion has set in.  Can anyone point me to a good CLI example of the following:

- Setup of SIP trunk

- Dial Peers.  In summary I need to be able to press say 8....... to get a Line through SIP and 9...... to get a line through FXO

- An example of how to Map SIP DDI's to internal Extensions.

This is what I have from the Provider:

Number of Channels: 4
IP Address: 178.AA.42.156

SIP signalling gateway address: 88.BB.61.195

- UDP port 5060 egress/ingress to 88.BB.61.195 (IPDC SIP signalling gateways).
- All UDP ports between 6000 - 40000 egress/ingress to 88.BB.61.196 (IPDC Media gateway) Omission of this setting will result in one-way speech

DDI 02034478900- 02034478910

From what I can gather they haven't provided a Username/password and are authenticating my use of the service via my source IP so I assume something like as follows:

Internal Extensions are 2000-2100

voice translation-profile PROVIDER

translate calling 2010

translate called 2011

translate redirect-called 2011

dial-peer voice 888 voip

description SIP to Provider

translation-profile outgoing PROVIDER

preference 1

destination-pattern [8]...

rtp payload-type comfort-noise 13

session protocol sipv2

session target ipv4:88.BB.61.195

session transport udp

voice-class codec 2

dtmf-relay rtp-nte

fax-relay ecm disable

no vad

I assume I have to create a copy of above for every destination pattern that is present for the FXO to go out the SIP when preceeding 8 is pressed.

No idea on how to map the DDI's to an internal extension.

Am I missing anything?

I should also mention there is another SIP trunk to another UC520 over a VPN.  This is what gets messed up when I use the CCA.

Thanks in advance

Dave

[Tagir Temirgaliyev]

voice service voip

allow−connections sip to sip

no supplementary−service sip moved−temporarily

!−−−Disable 302 sending

no supplementary−service sip refer

!−−−Disable REFER sending

sip

registrar server expires max 3600 min 3600

localhost dns:domain.test.com

!

!

voice class codec 1

codec preference 1 g711ulaw

!

!

!

!

!

!

!−−− SIP UA Configuration −−−

sip−ua

authentication username 5123781000 password 075A701E1D5E415447425B

no remote−party−id

retry invite 2

retry register 10

retry options 0

timers connect 100

registrar dns:domain.test.com expires 3600

sip−server dns:domain.test.com

host−registrar

!

!

!−−− CME Telephony Service Configuration −−−

telephony−service

no auto−reg−ephone

load 7960−7940 P0030702T023

max−ephones 168

max−dn 500

ip source−address 172.22.1.107 port 2000

calling−number initiator

!−−− Preserves the caller−id of a call when transferred or forwarded

dialplan−pattern 1 51237812.. extension−length 3 extension−pattern 2.. no−reg

voicemail 600

max−conferences 12 gain −6

call−forward pattern .T

call−forward system redirecting−expanded

!−−− Enables translation rule features for call−forwarding

moh music−on−hold.au

transfer−system full−consult dss

transfer−pattern 9.T

secondary−dialtone 9

create cnf−files version−stam

[davidfield]

Hello and thankyou for the above sample and it has helped.

I'm still having some issues but at least now I can get a handset to ring internally when I ring the SIP trunk number. When the call is anwsered I just jet continuous ringing and no call connect.   From the SIP trace it looks like I get a 403 Forbidden but from what I can see the SIP trunk is up and running.

The only reason we are using the SIP trunk is to allow some staff to had Direct DDI numbers but primarily for a DDI for the Auto Attendant on 2888.  So I only really need inbound calling ans the ability to translate to their internal extension. All outbound calling is via the analogue lines.

PS: I've disabled trust list and access list at the moment on the Public interface for testing.  So really the only config on the router is the NAT translations below.

Can anyone point me in the right direction?

Thanks in advance Dave

SIP trunk config

voice translation-rule 4

rule 1 /020366XXXXX/ /2888/

sip-ua

keepalive target ipv4:88.BB.61.195:5060

no remote-party-id

retry invite 2

retry register 10

timers connect 100

timers keepalive active 100

sip-server ipv4:88.BB.61.195:5060

host-registrar

g729-annexb override

ephone-dn  64

number 020366XXXXX

description SIP Main Number registration

preference 10

dial-peer voice 3000 voip

description DDIRange

translation-profile incoming DDIRange_Called_4

incoming called-number 020366XXXXX

!

dial-peer voice 1000 voip

permission term

description ** Incoming call from SIP trunk (Gamma) **

session protocol sipv2

session target sip-server

incoming called-number .%

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

Router config

ip nat pool SIPPORTFWD 10.44.1.254 10.44.1.254 netmask 255.255.255.0 type rotary

ip nat inside source list 1 interface Dialer0 overload

ip nat inside source static udp 10.44.1.254 5060 interface Dialer0 5060

ip nat inside source static tcp 10.44.1.254 5060 interface Dialer0 5060

ip nat inside destination list 100 pool SIPPORTFWD

ip route 0.0.0.0 0.0.0.0 Dialer0

access-list 100 permit udp any any range 4000 6000

Stip Status

UC520London#sh sip-ua status

SIP User Agent Status

SIP User Agent for UDP : ENABLED

SIP User Agent for TCP : ENABLED

SIP User Agent for TLS over TCP : ENABLED

SIP User Agent bind status(signaling): DISABLED

SIP User Agent bind status(media): DISABLED

SIP early-media for 180 responses with SDP: ENABLED

SIP max-forwards : 70

SIP DNS SRV version: 2 (rfc 2782)

NAT Settings for the SIP-UA

Role in SDP: NONE

Check media source packets: DISABLED

Maximum duration for a telephone-event in NOTIFYs: 2000 ms

SIP support for ISDN SUSPEND/RESUME: ENABLED

Redirection (3xx) message handling: ENABLED

Reason Header will override Response/Request Codes: DISABLED

Out-of-dialog Refer: DISABLED

Presence support is DISABLED

protocol mode is ipv4

SDP application configuration:

Version line (v=) required

Owner line (o=) required

Timespec line (t=) required

Media supported: audio video image

Network types supported: IN

Address types supported: IP4 IP6

Transport types supported: RTP/AVP udptl

SIP trace:

eceived:

INVITE sip:020366XXXXX@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>

From: Anonymous ;tag=3567595023-902814

Call-ID: 2052182-3567595023-902808@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK91317f87f846673dc8a1e9f22f018b64

Contact: <>anonymous@88.BB.61.195:5060>

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 291

v=0

o=MSX27 8020304351181920704 1 IN IP4 88.BB.61.195

s=sip call

c=IN IP4 88.BB.61.196

t=0 0

m=audio 37052 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

035630: //33514/86BC294EA21E/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK91317f87f846673dc8a1e9f22f018b64

From: Anonymous ;tag=3567595023-902814

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>

Date: Sat, 19 Jan 2013 14:37:03 gmt

Call-ID: 2052182-3567595023-902808@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

035631: //33514/86BC294EA21E/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK91317f87f846673dc8a1e9f22f018b64

From: Anonymous ;tag=3567595023-902814

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>;tag=3DB43710-1D63

Date: Sat, 19 Jan 2013 14:37:03 gmt

Call-ID: 2052182-3567595023-902808@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=21

Content-Length: 0

035632: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:020366XXXXX@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>;tag=3DB43710-1D63

From: Anonymous ;tag=3567595023-902814

Call-ID: 2052182-3567595023-902808@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK91317f87f846673dc8a1e9f22f018b64

Contact: <>anonymous@88.BB.61.195:5060>

P-Charging-Vector: icid-value=00001mqPBF;icid-generated-at=10.120.0.0;orig-ioi=tes001-tss-pst.gamma.uktel.org.uk

Content-Length: 0

035633: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:020366XXXXX@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>

From: Anonymous ;tag=3567595026-635196

Call-ID: 2052186-3567595026-635188@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK5d9d0ab5eb372d2af55964025249a27e

Contact: <>anonymous@88.BB.61.195:5060>

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 290

v=0

o=MSX27 355439370013519270 1 IN IP4 88.BB.61.195

s=sip call

c=IN IP4 88.BB.61.196

t=0 0

m=audio 37060 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

035634: //33515/885E44DFA223/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK5d9d0ab5eb372d2af55964025249a27e

From: Anonymous ;tag=3567595026-635196

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>

Date: Sat, 19 Jan 2013 14:37:06 gmt

Call-ID: 2052186-3567595026-635188@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

035635: //33515/885E44DFA223/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK5d9d0ab5eb372d2af55964025249a27e

From: Anonymous ;tag=3567595026-635196

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>;tag=3DB441C4-1F8A

Date: Sat, 19 Jan 2013 14:37:06 gmt

Call-ID: 2052186-3567595026-635188@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=21

Content-Length: 0

035636: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:020366XXXXX@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>;tag=3DB441C4-1F8A

From: Anonymous ;tag=3567595026-635196

Call-ID: 2052186-3567595026-635188@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK5d9d0ab5eb372d2af55964025249a27e

Contact: <>anonymous@88.BB.61.195:5060>

P-Charging-Vector: icid-value=00001JOBYy;icid-generated-at=10.121.0.0;orig-ioi=tes001-tss-tra.gamma.uktel.org.uk

Content-Length: 0

035637: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:020366XXXXX@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>

From: <>+21268947446@88.BB.61.195>;tag=3567595028-778280

P-Asserted-Identity: <>+21268947446@88.BB.61.195;user=phone>

Call-ID: 2052190-3567595028-778272@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK8d95134536808daa650be8e082389233

Contact: <>+21268947446@88.BB.61.195:5060>

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 291

v=0

o=MSX27 8052248780475719226 1 IN IP4 88.BB.61.195

s=sip call

c=IN IP4 88.BB.61.196

t=0 0

m=audio 37068 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

035638: //33516/89A43523A228/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK8d95134536808daa650be8e082389233

From: <>+21268947446@88.BB.61.195>;tag=3567595028-778280

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>

Date: Sat, 19 Jan 2013 14:37:08 gmt

Call-ID: 2052190-3567595028-778272@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

035639: //33516/89A43523A228/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK8d95134536808daa650be8e082389233

From: <>+21268947446@88.BB.61.195>;tag=3567595028-778280

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>;tag=3DB44A1C-1672

Date: Sat, 19 Jan 2013 14:37:08 gmt

Call-ID: 2052190-3567595028-778272@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=21

Content-Length: 0

035640: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:020366XXXXX@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>;tag=3DB44A1C-1672

From: <>+21268947446@88.BB.61.195>;tag=3567595028-778280

Call-ID: 2052190-3567595028-778272@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK8d95134536808daa650be8e082389233

Contact: <>+21268947446@88.BB.61.195:5060>

P-Charging-Vector: icid-value=00001mqPDa;icid-generated-at=10.120.0.0;orig-ioi=tes001-tss-pst.gamma.uktel.org.uk

Content-Length: 0

035641: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:020366XXXXX@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>

From: <>+21268947446@88.BB.61.195>;tag=3567595030-728142

P-Asserted-Identity: <>+21268947446@88.BB.61.195;user=phone>

Call-ID: 2052192-3567595030-728133@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK18dafbd1d89e8995616aeecba29f6fe5

Contact: <>+21268947446@88.BB.61.195:5060>

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 290

v=0

o=MSX27 805097837701719365 1 IN IP4 88.BB.61.195

s=sip call

c=IN IP4 88.BB.61.196

t=0 0

m=audio 37072 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

035642: //33517/8AD082ADA22D/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK18dafbd1d89e8995616aeecba29f6fe5

From: <>+21268947446@88.BB.61.195>;tag=3567595030-728142

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>

Date: Sat, 19 Jan 2013 14:37:10 gmt

Call-ID: 2052192-3567595030-728133@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

035643: //33517/8AD082ADA22D/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK18dafbd1d89e8995616aeecba29f6fe5

From: <>+21268947446@88.BB.61.195>;tag=3567595030-728142

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>;tag=3DB451CC-649

Date: Sat, 19 Jan 2013 14:37:10 gmt

Call-ID: 2052192-3567595030-728133@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=21

Content-Length: 0

035644: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:020366XXXXX@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>;tag=3DB451CC-649

From: <>+21268947446@88.BB.61.195>;tag=3567595030-728142

Call-ID: 2052192-3567595030-728133@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK18dafbd1d89e8995616aeecba29f6fe5

Contact: <>+21268947446@88.BB.61.195:5060>

P-Charging-Vector: icid-value=00001mqPER;icid-generated-at=10.120.0.0;orig-ioi=tes001-tss-pst.gamma.uktel.org.uk

Content-Length: 0

035645: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:020366XXXXX@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <>020366XXXXX@88.BB.61.195:5060;user=phone>

From: Anonymous ;tag=3567595033-871432

Call-ID: 2052196-3567595033-871426@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.BB.61.195:5060;branch=z9hG4bK5b0510fde6c0568e953b811145d6a896

Contact: <>anonymous@88.BB.61.195:5060>

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 291

[Ayodeji Okanlawon]

Reason why call is disconnected is this

Sent:

SIP/2.0 403 Forbidden

Reason: Q.850;cause=21

To resolve this..

You need to configure the ip address of your sip providedr under the ip address trusted list as follows:

voice service voip                                   

    ip address trusted list                             

     ipv4 []          

Please rate all useful posts

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

[Ayodeji Okanlawon]

Reason why call is disconnected is this

Sent:

SIP/2.0 403 Forbidden

Reason: Q.850;cause=21

To resolve this..

You need to configure the ip address of your sip providedr under the ip address trusted list as follows:

voice service voip                                   

    ip address trusted list                             

     ipv4 []          

Please rate all useful posts

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

[davidfield]

Thanks for hte quick repsonse.  For testing purposes I applied accept all

voice service voip

ip address trusted list

  ipv4 0.0.0.0 0.0.0.0

The 402 forbiden is gone now but I still dont get a call to the autoattendant.  I think my dial peers are a bit messed up adn I'm also getting 503 service unavailable. I can dial Auto Attendant on 2888 ok from inside.

voice translation-rule 4

rule 1 /02036678903/ /2888/

sip-ua

keepalive target ipv4:88.BB.61.195:5060

no remote-party-id

retry invite 2

retry register 10

timers connect 100

timers keepalive active 100

sip-server ipv4:88.BB.61.195:5060

host-registrar

g729-annexb override

ephone-dn  64

number 02036678903

description SIP Main Number registration

preference 10

voice translation-profile DDIRange_Called_4

translate calling 3265

translate called 4

!

dial-peer voice 3000 voip

description DDIRange

translation-profile incoming DDIRange_Called_4

incoming called-number 020366XXXX

!

dial-peer voice 1000 voip

permission term

description ** Incoming call from SIP trunk (Gamma) **

session protocol sipv2

session target sip-server

incoming called-number .%

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

Kindest Regards

David

Sip messages updated

UC520London#

038231: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:02036678903@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <02036678903>

From: Anonymous ;tag=3567602887-218887

Call-ID: 2074374-3567602887-218880@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK0ba712fb0a799c8991707f8325564588

Contact:

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 290

v=0

o=MSX27 805031372727595675 1 IN IP4 88.215.61.195

s=sip call

c=IN IP4 88.215.61.196

t=0 0

m=audio 12572 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

038232: //34293/D5A19D6DA81F/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK0ba712fb0a799c8991707f8325564588

From: Anonymous ;tag=3567602887-218887

To: <02036678903>

Date: Sat, 19 Jan 2013 16:48:07 gmt

Call-ID: 2074374-3567602887-218880@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

038233: //34293/D5A19D6DA81F/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK0ba712fb0a799c8991707f8325564588

From: Anonymous ;tag=3567602887-218887

To: <02036678903>;tag=3E2C3220-9AA

Date: Sat, 19 Jan 2013 16:48:07 gmt

Call-ID: 2074374-3567602887-218880@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

Content-Length: 0

038234: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:02036678903@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <02036678903>;tag=3E2C3220-9AA

From: Anonymous ;tag=3567602887-218887

Call-ID: 2074374-3567602887-218880@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK0ba712fb0a799c8991707f8325564588

Contact:

P-Charging-Vector: icid-value=00001mrGo0;icid-generated-at=10.120.0.0;orig-ioi=tes001-tss-pst.gamma.uktel.org.uk

Content-Length: 0

038235: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:02036678903@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <02036678903>

From: Anonymous ;tag=3567602887-320123

Call-ID: 2074375-3567602887-320117@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK63c5132a9094e9192af12818c740f8c4

Contact:

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 289

v=0

o=MSX27 80201582476225700 1 IN IP4 88.215.61.195

s=sip call

c=IN IP4 88.215.61.196

t=0 0

m=audio 12574 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

038236: //34296/D5B0DFB7A826/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK63c5132a9094e9192af12818c740f8c4

From: Anonymous ;tag=3567602887-320123

To: <02036678903>

Date: Sat, 19 Jan 2013 16:48:07 gmt

Call-ID: 2074375-3567602887-320117@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

038237: //34296/D5B0DFB7A826/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK63c5132a9094e9192af12818c740f8c4

From: Anonymous ;tag=3567602887-320123

To: <02036678903>;tag=3E2C3288-1119

Date: Sat, 19 Jan 2013 16:48:07 gmt

Call-ID: 2074375-3567602887-320117@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

Content-Length: 0

038238: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:02036678903@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <02036678903>;tag=3E2C3288-1119

From: Anonymous ;tag=3567602887-320123

Call-ID: 2074375-3567602887-320117@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK63c5132a9094e9192af12818c740f8c4

Contact:

P-Charging-Vector: icid-value=00001mrGo0;icid-generated-at=10.120.0.0;orig-ioi=tes001-tss-pst.gamma.uktel.org.uk

Content-Length: 0

038239: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:02036678903@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <02036678903>

From: Anonymous ;tag=3567602887-420677

Call-ID: 2074376-3567602887-420670@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK4d4f7748163fc2804f0ad064bd3fe33b

Contact:

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 291

v=0

o=MSX27 8020234744357331980 1 IN IP4 88.215.61.195

s=sip call

c=IN IP4 88.215.61.196

t=0 0

m=audio 12576 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

038240: //34299/D5C02167A82D/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK4d4f7748163fc2804f0ad064bd3fe33b

From: Anonymous ;tag=3567602887-420677

To: <02036678903>

Date: Sat, 19 Jan 2013 16:48:07 gmt

Call-ID: 2074376-3567602887-420670@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

038241: //34299/D5C02167A82D/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK4d4f7748163fc2804f0ad064bd3fe33b

From: Anonymous ;tag=3567602887-420677

To: <02036678903>;tag=3E2C32F0-248F

Date: Sat, 19 Jan 2013 16:48:07 gmt

Call-ID: 2074376-3567602887-420670@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

Content-Length: 0

[Ayodeji Okanlawon]

Where is the dial-peer for the auto attendant? 2888..I cant see it...

can you also send the debug voip ccapi inout

Please rate all useful posts

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

[Ayodeji Okanlawon]

Where is the dial-peer for the auto attendant? 2888..I cant see it...

can you also send the debug voip ccapi inout

Please rate all useful posts

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

[davidfield]

Thanks Aokanlawan for helping  I have attached the UC520 config, debug Voip Ccapi & debug translations

I think my issue is the translations but the error 503 from what I've read is indicative of a more fundamental issue.  As menitoned above I just need this one number ot land on the Auttoattendant and for the caller to be able to menu their way through to someone onsite.

I tried to use the CCA to configure the SIP trunk and it messed up all my call handling with my other sites.  So rolled back and trying to configure via CLI and its proving to be quite a mission to put it together.

The pressure is on me to have the inbound calling running by Monday so I really appreciate your input.

Regards

David

[davidfield]

I just noticed that there is a Dial Peer 2001 being referenced in the Ccapi which relates to the Autoattendant.  2001 has a translation profile of PSTN_Call_forwarding which is not in the config.  when I purged the CCA config I took all the translations out as they were causing issues as mentioned above.  So when a call comes in do I also need a Dial peer for the outbound response also?

This what was removed.

voice translation-rule 410

rule 1 /^9\(.*\)/ /\1/

rule 15 /^....$/ /02036678903/

!

voice translation-rule 411

rule 1 /^9\(.*\)/ /ABCD9\1/

!

voice translation-rule 412

rule 1 /^ABCD\(.*\)/ /\1/

!

voice translation-rule 422

rule 1 /^ABCD909[01]......../ //

rule 2 /^ABCD9090[89]......./ //

rule 3 /^ABCD9098\(.*\)/ //

rule 15 /^ABCD\(.*\)/ /\1/

voice translation-profile PSTN_CallForwarding

translate redirect-target 410

translate redirect-called 410

!

voice translation-profile PSTN_Outgoing

translate calling 1111

translate called 1112

translate redirect-target 410

translate redirect-called 410

!

voice translation-profile SIP_Incoming

translate called 411

!

voice translation-profile SIP_Passthrough

translate called 412

!

voice translation-profile SIP_Passthrough_CallBlocking

translate called 422

[Ayodeji Okanlawon]

Hi,

Please configure your dial-peer as follows:

dial-peer voice 3000 voip

description DDIRange

translation-profile incoming DDIRange_Called_4

incoming called-number 02036678903

codec g711ulaw

session protocol sipv2

dtmf-relay rtp-nte

no vad

Then configure this..

voice service voip

allow connections sip to sip

Test again and send debug ccsip messages

Please rate all useful posts

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

[Ayodeji Okanlawon]

Hi,

Please configure your dial-peer as follows:

dial-peer voice 3000 voip

description DDIRange

translation-profile incoming DDIRange_Called_4

incoming called-number 02036678903

codec g711ulaw

session protocol sipv2

dtmf-relay rtp-nte

no vad

Then configure this..

voice service voip

allow connections sip to sip

Test again and send debug ccsip messages

Please rate all useful posts

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

[davidfield]

I get a carrier message saying the service cannot be connected. below is also CCSIP for codec G729r8 and get busy tone from carrier.

Codec g711ulaw

C520London#

043832: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:02036678903@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <02036678903>

From: Anonymous ;tag=3567610710-66444

Call-ID: 2094307-3567610710-66437@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK641c34fb3ca4f8206c4bd5272547073d

Contact:

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 291

v=0

o=MSX27 3586393515818613715 1 IN IP4 88.215.61.195

s=sip call

c=IN IP4 88.215.61.196

t=0 0

m=audio 17596 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

043833: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 488 Not Acceptable Media

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK641c34fb3ca4f8206c4bd5272547073d

From: Anonymous ;tag=3567610710-66444

To: <02036678903>;tag=3EA38EFC-8EB

Date: Sat, 19 Jan 2013 18:58:30 gmt

Call-ID: 2094307-3567610710-66437@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Warning: 304 10.44.1.254 "Media Type(s) Unavailable"

Reason: Q.850;cause=65

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Codec for dial peer 3000  set to g729r8 - get busy signal

UC520London#

043845: Jan 19 19:03:30.297: %SYS-5-CONFIG_I: Configured from console by cisco on vty0 (10.44.1.2)

043846: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:02036678903@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <02036678903>

From: Anonymous ;tag=3567611021-709418

Call-ID: 2095053-3567611021-709411@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKd2196116f7d14da022ab0ea32b42a839

Contact:

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 291

v=0

o=MSX27 8020103151071525935 1 IN IP4 88.215.61.195

s=sip call

c=IN IP4 88.215.61.196

t=0 0

m=audio 17758 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

043847: //35289/C62A3BC3AF87/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKd2196116f7d14da022ab0ea32b42a839

From: Anonymous ;tag=3567611021-709418

To: <02036678903>

Date: Sat, 19 Jan 2013 19:03:41 gmt

Call-ID: 2095053-3567611021-709411@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

043848: //35289/C62A3BC3AF87/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKd2196116f7d14da022ab0ea32b42a839

From: Anonymous ;tag=3567611021-709418

To: <02036678903>;tag=3EA85068-2332

Date: Sat, 19 Jan 2013 19:03:41 gmt

Call-ID: 2095053-3567611021-709411@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

Content-Length: 0

043849: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:02036678903@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <02036678903>;tag=3EA85068-2332

From: Anonymous ;tag=3567611021-709418

Call-ID: 2095053-3567611021-709411@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKd2196116f7d14da022ab0ea32b42a839

Contact:

P-Charging-Vector: icid-value=00001mryTP;icid-generated-at=10.120.0.0;orig-ioi=tes001-tss-pst.gamma.uktel.org.uk

Content-Length: 0

043850: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:02036678903@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <02036678903>

From: Anonymous ;tag=3567611021-825058

Call-ID: 2095054-3567611021-825050@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKf9daa50b7e6c072d22ec9e37a17cc767

Contact:

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 290

v=0

o=MSX27 801812905228314342 1 IN IP4 88.215.61.195

s=sip call

c=IN IP4 88.215.61.196

t=0 0

m=audio 17760 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

043851: //35292/C63E607DAF8E/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKf9daa50b7e6c072d22ec9e37a17cc767

From: Anonymous ;tag=3567611021-825058

To: <02036678903>

Date: Sat, 19 Jan 2013 19:03:41 gmt

Call-ID: 2095054-3567611021-825050@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

043852: //35292/C63E607DAF8E/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKf9daa50b7e6c072d22ec9e37a17cc767

From: Anonymous ;tag=3567611021-825058

To: <02036678903>;tag=3EA850EC-64B

Date: Sat, 19 Jan 2013 19:03:41 gmt

Call-ID: 2095054-3567611021-825050@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

Content-Length: 0

043853: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:02036678903@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <02036678903>;tag=3EA850EC-64B

From: Anonymous ;tag=3567611021-825058

Call-ID: 2095054-3567611021-825050@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKf9daa50b7e6c072d22ec9e37a17cc767

Contact:

P-Charging-Vector: icid-value=00001mryTP;icid-generated-at=10.120.0.0;orig-ioi=tes001-tss-pst.gamma.uktel.org.uk

Content-Length: 0

043854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:02036678903@10.44.1.254;user=phone SIP/2.0

Max-Forwards: 69

Session-Expires: 3600;refresher=uac

Min-SE: 600

Supported: timer, 100rel

To: <02036678903>

From: Anonymous ;tag=3567611021-964588

Call-ID: 2095055-3567611021-964582@MSX27.gammatelecom.com

CSeq: 1 INVITE

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKee873fc0b7285e08252bd8346e061349

Contact:

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 290

v=0

o=MSX27 805270580154421636 1 IN IP4 88.215.61.195

s=sip call

c=IN IP4 88.215.61.196

t=0 0

m=audio 17762 RTP/AVP 8 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no;mr=0;mg=0;jmdelay=no;mrmods=

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

043855: //35295/C653BD87AF95/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKee873fc0b7285e08252bd8346e061349

From: Anonymous ;tag=3567611021-964588

To: <02036678903>

Date: Sat, 19 Jan 2013 19:03:41 gmt

Call-ID: 2095055-3567611021-964582@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

043856: //35295/C653BD87AF95/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKee873fc0b7285e08252bd8346e061349

From: Anonymous ;tag=3567611021-964588

To: <02036678903>;tag=3EA85178-1D5E

Date: Sat, 19 Jan 2013 19:03:41 gmt

Call-ID: 2095055-3567611021-964582@MSX27.gammatelecom.com

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

Content-Length: 0

043857: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:02036678903@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <02036678903>;tag=3EA85178-1D5E

From: Anonymous ;tag=3567611021-964588

Call-ID: 2095055-3567611021-964582@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bKee873fc0b7285e08252bd8346e061349

Contact:

P-Charging-Vector: icid-value=00001mryTP;icid-generated-at=10.120.0.0;orig-ioi=tes001-tss-pst.gamma.uktel.org.uk

Content-Length: 0

043834: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:02036678903@10.121.30.4;user=phone SIP/2.0

Max-Forwards: 70

To: <02036678903>;tag=3EA38EFC-8EB

From: Anonymous ;tag=3567610710-66444

Call-ID: 2094307-3567610710-66437@MSX27.gammatelecom.com

CSeq: 1 ACK

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

Via: SIP/2.0/UDP 88.215.61.195:5060;branch=z9hG4bK641c34fb3ca4f8206c4bd5272547073d

Contact:

P-Charging-Vector: icid-value=00001JPi6i;icid-generated-at=10.121.0.0;orig-ioi=tes001-tss-tra.gamma.uktel.org.uk

Content-Length: 0

[Ayodeji Okanlawon]

Ok..I see whats going on now..

Your provider are only sending G711alaw and G729. These are the only two codecs they can support. This is going to be an issue because the auto attendant only supports g711ulaw i believe..So we are going to need some MTP resource...

Do you have any dsp or pvdm in your gateway...

Please rate all useful posts

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

[davidfield]

ok.   I'm still a bit new to the SIP/Cisco Voice.  So before probing the DSP/PVDM requirements, would this cause a problem with assigning the SIP inbound to say an extension and not the Auto-Attendant?

I ran the following re DSP. Not sure if this helps with what is/isn't presnet.  I inherited the system.

dsp 1:

State: UP, firmware: 28.3.5

Max signal/voice channel: 16/16

Max credits: 240

num_of_sig_chnls_allocated: 14

Transcoding channels allocated: 0

Group: FLEX_GROUP_VOICE, complexity: FLEX

Shared credits: 225, reserved credits: 0

Signaling channels allocated: 14

Voice channels allocated: 1

Credits used (rounded-up): 15

Voice channels:

Ch01: voice port: 0/4/0, codec: g711ulaw, credits allocated: 15

dsp 2:

State: UP, firmware: 28.3.5

Max signal/voice channel: 16/16

Max credits: 240

num_of_sig_chnls_allocated: 0

Transcoding channels allocated: 0

Group: FLEX_GROUP_CONF, complexity: CONFERENCE

Shared credits: 0, reserved credits: 240

Codec: CONF_G711, maximum participants: 32

Sessions per dsp: 2

0 DSP resource allocation failure

UC520London#sh dspfarm all

Dspfarm Profile Configuration

Profile ID = 1, Service = CONFERENCING, Resource ID = 1

Profile Description : DO NOT MODIFY, active CCA conference profile - CCA2.0 codec711

Profile Service Mode : Non Secure

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SCCP   Status : ASSOCIATED

Resource Provider : FLEX_DSPRM   Status : UP

Number of Resource Configured : 2

Number of Resource Available : 2

Maximum conference participants : 32

Codec Configuration: num_of_codecs:2

Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required

Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required

SLOT DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0    2   28.3.5   UP     N/A  FREE  conf    1      -         -         -

0    2   28.3.5   UP     N/A  FREE  conf    1      -         -         -

Total number of DSPFARM DSP channel(s) 2

Cisco UC520W-16U-4FXO-K9 (MPC8358) processor (revision 0x202) with 235520K/26624K bytes of memory.

Processor board ID FTX1236Z1BW

MPC8358 CPU Rev: Part Number 0x804A, Revision ID 0x20

22 User Licenses

10 FastEthernet interfaces

2 terminal lines

1 Virtual Private Network (VPN) Module

4 Voice FXO interfaces

8 Voice FXS interfaces

1 Voice MoH interface

1 802.11 Radio

1 cisco service engine(s)

128K bytes of non-volatile configuration memory.

125440K bytes of ATA CompactFlash (Read/Write)