[cisco-voip] SIP provider woes.
Mark Holloway
mh at markholloway.com
Wed Jul 8 12:35:57 EDT 2009
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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