[cisco-voip] SIP provider woes.

Wed Jul 8 11:47:32 EDT 2009

You're getting a progress message from them which would suggest to me that
your part is working correctly.  I find it odd that a mere 8ms later their
side is cancelling the invite.

Unless there's something in the invite they don't like I can't see anything
out of the ordinary.  Have you verified the SDP capbilities with them in
that they wouldn't reject on anything listed there?  I've notated below what
each of the typically critical fields mean.

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Wednesday, July 08, 2009 9:50 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SIP provider woes.

Looking for some help.  I'm setting up a CUBE router to a Sip provider, but
am having issues getting calls placed.  The provider says the problem is on
my side, but I want to verify this.  Sip-ua register status shows as
registered.  Setup is UCM7.1=>h.323GW(CUBE)=>SIP to Provider.  Here is a
debug from ccsip messages:

*Jul  8 14:06:16.385: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent: 

INVITE sip:3172222222 at PROVIDERIP:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.3.253:5060;branch=z9hG4bK901E41

From: <sip:7655985006 at PROVIDERIP>;tag=9173C38-439

To: <sip:3172222222 at PROVIDERIP>

Date: Wed, 08 Jul 2009 14:06:16 GMT

Call-ID: 563ECD24-6AFF11DE-806A90D0-BD89AE8F at 192.168.3.253

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

Min-SE:  1800

Cisco-Guid: 2149292161-3701948837-1073764865-3232236289

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: 1247061976

Contact: <sip:7655985006 at 192.168.3.253:5060>

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

*Jul  8 14:06:16.409: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received: 

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.3.253:5060;branch=z9hG4bK901E41;received=OUTSIDEIP

From: <sip:7655985006 at PROVIDERIP>;tag=9173C38-439

To: <sip:3172222222 at PROVIDERIP>

Call-ID: 563ECD24-6AFF11DE-806A90D0-BD89AE8F at 192.168.3.253

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:3172222222 at PROVIDERIP>

Content-Length: 0

*Jul  8 14:06:16.637: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received: 

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.3.253:5060;branch=z9hG4bK901E41;received=OUTSIDEIP

From: <sip:7655985006 at PROVIDERIP>;tag=9173C38-439

To: <sip:3172222222 at PROVIDERIP>;tag=as5fb685bb

Call-ID: 563ECD24-6AFF11DE-806A90D0-BD89AE8F at 192.168.3.253

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:3172222222 at PROVIDERIP>

Content-Type: application/sdp

Content-Length: 361

v=0

o=root 3551 3551 IN IP4 PROVIDERIP

s=session

c=IN IP4 PROVIDERIP

b=CT:384  (bandwidth 384k) 

t=0 0

m=audio 17670 RTP/AVP 0 8 101  (audio services) 

a=rtpmap:0 PCMU/8000  (g.711 mu law) 

a=rtpmap:8 PCMA/8000  (g.711 a law) 

a=rtpmap:101 telephone-event/8000  (dynamic audio payload) 

a=fmtp:101 0-16  (dynamic payload spefication) 

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv  (can send and receive) 

m=video 13472 RTP/AVP 34 99  (video services) 

a=rtpmap:34 H263/90000  (h.263) 

a=rtpmap:99 H264/90000  (h.264) 

a=sendrecv  (can send and receive) 

*Jul  8 14:06:16.645: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent: 

CANCEL sip:3172222222 at PROVIDERIP:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.3.253:5060;branch=z9hG4bK901E41

From: <sip:7655985006 at PROVIDERIP>;tag=9173C38-439

To: <sip:3172222222 at PROVIDERIP>

Date: Wed, 08 Jul 2009 14:06:16 GMT

Call-ID: 563ECD24-6AFF11DE-806A90D0-BD89AE8F at 192.168.3.253

CSeq: 101 CANCEL

Max-Forwards: 70

Timestamp: 1247061976

Reason: Q.850;cause=127

Content-Length: 0

*Jul  8 14:06:16.665: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received: 

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 192.168.3.253:5060;branch=z9hG4bK901E41;received=OUTSIDEIP

From: <sip:7655985006 at PROVIDERIP>;tag=9173C38-439

To: <sip:3172222222 at PROVIDERIP>;tag=as5fb685bb

Call-ID: 563ECD24-6AFF11DE-806A90D0-BD89AE8F at 192.168.3.253

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

*Jul  8 14:06:16.669: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent: 

ACK sip:3172222222 at PROVIDERIP:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.3.253:5060;branch=z9hG4bK901E41

From: <sip:7655985006 at PROVIDERIP>;tag=9173C38-439

To: <sip:3172222222 at PROVIDERIP>;tag=as5fb685bb

Date: Wed, 08 Jul 2009 14:06:16 GMT

Call-ID: 563ECD24-6AFF11DE-806A90D0-BD89AE8F at 192.168.3.253

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

*Jul  8 14:06:16.669: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received: 

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.3.253:5060;branch=z9hG4bK901E41;received=OUTSIDEIP

From: <sip:7655985006 at PROVIDERIP>;tag=9173C38-439

To: <sip:3172222222 at PROVIDERIP>;tag=as5fb685bb

Call-ID: 563ECD24-6AFF11DE-806A90D0-BD89AE8F at 192.168.3.253

CSeq: 101 CANCEL

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:3172222222 at PROVIDERIP>

Content-Length: 0

Relevant CONFIG:

sip-ua 

 credentials username <USERNAME> password 7 PASSWORD realm asterisk

 authentication username <USERNAME> password 7 PASSWORD realm asterisk

 no remote-party-id

 retry invite 2

 retry register 2

 registrar ipv4:PROVIDERIP expires 3600

 sip-server ipv4:PROVIDERIP

 reason-header override

  host-registrar

voice service voip 

 allow-connections h323 to sip

 allow-connections sip to h323

 allow-connections sip to sip

 no supplementary-service sip moved-temporarily

 no supplementary-service sip refer

 sip

  bind control source-interface Vlan100

  bind media source-interface Vlan100

  registrar server

Also to complicate matters a bit the CUBE is sitting behind an ASA firewall.
The ASA does have a static NAT for SIP on the outside interface back into
the CUBE and I have SIP inspection enabled.

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