[cisco-voip] SIP provider woes.

Nick Matthews matthnick at gmail.com
Wed Jul 8 15:40:36 EDT 2009 

This most likely is due to the SDP that the provider is sending the
gateway.  Cause 127 is generally an internal software error of some
variety.  A good command that sometimes offers more information is the
global config "voice iec syslog".  I would put that on and see if it
helps.

Otherwise, upgrade your CUBE version if you're not past 12.4(20)T, as
you may be on an older version hitting a bug.  If neither of those
work, get 'debug ccsip all' and TAC can take a look.


-nick

On Wed, Jul 8, 2009 at 3:26 PM, Pender, James<James.Pender at paetec.com> wrote:
> 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
> 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
> 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
> 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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