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No calls from CME registered phones to a SIP provider trunk

Hello everyone,

I'm testing a new voice connection to a SIP provider.

I've enabled the following:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

The sip-ua configuration is has follows:

sip-ua

credentials username XXXXX password YYYY realm sip.XXX.com

authentication username XXXXX password YYYY

registrar dns:sip.XXX.com expires 3600

sip-server dns:sip.XXX.com

From the "debug ccsip messages" i can see the Register request being sent and the 200 OK being received and running a "show sip-ua register status" i can see the XXXXX as registered.

The problem is, when i try to call a specific test number that i want to go out through this SIP provider, i don't see the INVITE packet being sent, altough in the "debug voice dialpeer" i can see the following dial-peer being matched:

dial-peer voice 11 voip

destination-pattern 123456789

session protocol sipv2

session target dns:sip.XXX.com

dtmf-relay sip-notify

codec transparent

no vad

Any idea why?

Thanks

  • IP Telephony
7 REPLIES

No calls from CME registered phones to a SIP provider trunk

By the way:

Router 2911 running version 15.1

Telephony version 8.6

VIP Super Bronze

No calls from CME registered phones to a SIP provider trunk

Hi,

Can you try the ff:

Dial-peer voice 11 voip

Codec G711u

Can you send the output of your debug ccsip message here

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

No calls from CME registered phones to a SIP provider trunk

The codec doesn't seem to be the issue here, since even the control info is not going through.

The problem is theres no output of the debug ccsip message, apart from the regular registration:

May 25 10:56:18.294: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

REGISTER sip:sip.XXXXXX.com:5060 SIP/2.0

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKBB21A47

From: <>XXXXXXXX@sip.XXX.com>;tag=3892FB54-21B4

To: <>XXXXXXXXX@sip.XXX.com>

Date: Fri, 25 May 2012 10:56:18 GMT

Call-ID: A2F0AAE-A4F611E1-AFF5C3CC-899281C4

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

Max-Forwards: 70

Timestamp: 1337943378

CSeq: 299 REGISTER

Contact:

Expires: 3600

Authorization: Digest username="XXXXXXXXXXXXXX",realm="sip.XXX.com",uri="sip:sip.XXX.com:5060",response="1599e81611c486f6d01a900d7582bc36",nonce="4fbf6513000110ce2eec642e242d5c49228d65a55be69b0e",algorithm=MD5

Content-Length: 0

May 25 10:56:18.666: //8366/000000000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

From: <>XXXXXXXXXXX@sip.XXX.com>;tag=3892FB54-21B4

To: <>XXXXXXXXXXXX@sip.XXX.com>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.ef3b

Call-ID: A2F0AAE-A4F611E1-AFF5C3CC-899281C4

CSeq: 299 REGISTER

Via: SIP/2.0/UDP 172.26.12.253:5060;branch=z9hG4bKBB21A47

Contact: ;expires=300

Server: OpenSIPS

Expires: 300

Content-Length: 0

Cisco Employee

No calls from CME registered phones to a SIP provider trunk

Can the router resolve the sesion-target sip.XXX.com?

No calls from CME registered phones to a SIP provider trunk

yes.

And also with the session-target ipv4:x.x.x.x i still can't see the INVITE package, but the result is diferent. Instead of the call being dropped when i match the dial-peer, i get a fast busy signal.

No calls from CME registered phones to a SIP provider trunk

Solved.

Aparently my translation-profile wasn't working properly, and i wasn't announcing the correct user to my sip provider. I was expecting that if this was the problem i should at least see a Unauthorized packet coming from my SIP provider but that wasn't the case.

After changing my dial-peer:

dial-peer voice 11 voip

corlist outgoing Peer-Nacional

translation-profile outgoing Mask

destination-pattern 123456789

session protocol sipv2

session target dns:sip.XXX.com

dtmf-relay sip-notify

codec g711ulaw

clid network-number XXXXXXX

no vad  

Result:

May 25 11:37:39.330: //8511/DC5D325CB504/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:123456789@sip.XXX.com:5060 SIP/2.0

Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKBDA64B

From: "user" <>XXXXXXXX@sip.XXX.com>;tag=38B8D6EC-959

To: <>123456789@sip.XXX.com>

Date: Fri, 25 May 2012 11:37:39 GMT

Call-ID: DDAD8184-A59411E1-B509C3CC-899281C4@172.26.12.253

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

Min-SE:  1800

Cisco-Guid: 3697095260-2777944545-3036988364-2308080068

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: 1337945859

Contact:

Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=2000"

Expires: 180

Allow-Events: telephone-event

Proxy-Authorization: Digest username="XXXXXXXXXX",realm="sip.XXX.com",uri="sip:123456789@sip.XXX.com:5060",response="6f162daec8974185533a8a5699f5c57d",nonce="4fbf6ec400000279262314bacddf71622194962790d7152f",algorithm=MD5

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 194

v=0

o=CiscoSystemsSIP-GW-UserAgent 6922 6037 IN IP4 172.26.12.253

s=SIP Call

c=IN IP4 x.x.x.x

t=0 0

m=audio 17010 RTP/AVP 0

c=IN IP4 x.x.x.x

a=rtpmap:0 PCMU/8000

a=ptime:20

May 25 11:37:39.546: //8511/DC5D325CB504/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

...

May 25 11:37:42.394: //8511/DC5D325CB504/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

...

May 25 11:37:43.626: //8511/DC5D325CB504/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

...

v=0

o=CiscoSystemsSIP-GW-UserAgent 6922 6037 IN IP4 78.141.179.70

s=XXX call

c=IN IP4 78.141.179.70

t=0 0

m=audio 25872 RTP/AVP 0

a=rtpmap:0 pcmu/8000

a=ptime:20

VIP Super Bronze

No calls from CME registered phones to a SIP provider trunk

First of all glad to see you have resolved it.

Secondly, I strongly believe that you had an issue with your codec (i may also be wrong but we can confirm that). As I can see you are now using my suggested codec. So i am at a loss why you were quick to dismiss that suggestion even though you are now using it.

It will be intresting to see if your call will proceed if you revert back to codec transparent as you had previuosly. Do you wnat to give that a go and let me know what the result is..

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
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