[cisco-voip] SIP provider woes.

Wed Jul 8 15:26:52 EDT 2009

The reason in the cancel message is interesting, "Reason: Q.850;cause=127", "Interworking, unspecified". I would recommend you have the provider check corresponding messaging beyond the device you are connecting to (i.e SS7 etc..,)

- Jim Pender

________________________________
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Holloway
Sent: Wednesday, July 08, 2009 12:36 PM
To: Jason Burton
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP provider woes.

Here is an example of what I am using for trunking CME to Broadworks.  Also, make sure your ASA Xlate is set to at least 3600.

ISR

voice service voip
 callmonitor
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol pass-through g711ulaw
 sip

sip-ua
 credentials username XXXXXXXXXX password 7 223234 realm BroadWorks
 authentication username XXXXXXXXXX password 7 253F36 realm BroadWorks
 no remote-party-id
 set pstn-cause 3 sip-status 486
 set pstn-cause 34 sip-status 486
 set pstn-cause 47 sip-status 486
 registrar dns:sip.provider.com expires 3600
 sip-server dns:sip.provider.com
 connection-reuse   <--- Reuses the same source port each time it refreshes SIP registration; good practice
 host-registrar

ASA

timeout xlate 3:00:00
timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 rpc 0:10:00 h225 1:00:00
timeout h323 0:05:00 mgcp 0:05:00 sip 3:00:00 sip_media 0:02:00
timeout sip-disconnect 0:02:00 sip-invite 0:03:00
timeout uauth 0:05:00 absolute

On Jul 8, 2009, at 7:50 AM, Jason Burton wrote:

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<mailto: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<mailto: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<mailto: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
t=0 0
m=audio 17670 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 13472 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv
*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<mailto: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<mailto: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<mailto: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<mailto: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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