[cisco-voip] [Bulk] RE: Calls not forwarding over SIP trunk to Cue
Jay Stants
jaystants at rogers.com
Sun Aug 14 00:23:18 EDT 2011
Interface Details
snet-wan2#sh ip int br
Interface IP-Address OK? Method Status Protocol
FastEthernet0/0 10.50.1.2 YES NVRAM up up
Service-Engine0/1 1.1.1.1 YES TFTP up up
FastEthernet0/1 74.74.255.254 YES DHCP up up
GigabitEthernet1/0 10.1.99.1 YES NVRAM up up
Loopback0 10.50.250.4 YES NVRAM up up
Loopback100 1.1.1.1 YES NVRAM up up
NVI0 10.50.1.2 YES unset up up
interface Service-Engine0/1
ip unnumbered Loopback100
service-module ip address 1.1.1.2 255.255.255.252
service-module ip default-gateway 1.1.1.1
Regards,
Jay Stants
jaystants at rogers.com
From: "jaystants at rogers.com" <jaystants at rogers.com>
>To: "Buchanan, James" <jbuchanan at presidio.com>; "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
>Sent: Sunday, August 14, 2011 12:03:36 AM
>Subject: Re: [cisco-voip] [Bulk] RE: Calls not forwarding over SIP trunk to Cue
>
>
>1.1.1.2 is bound to cue. I have a service mod with aim-cue. Give me a min and ill send the interface layout
>
>Sent from my HTC bolt of lightning...
>
>
>----- Reply message -----
>From: "Buchanan, James" <jbuchanan at presidio.com>
>To: "Jay Stants" <jaystants at rogers.com>, "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
>Subject: [Bulk] RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue
>Date: Sat, Aug 13, 2011 11:48 pm
>
>
>
>I assume 1.1.1.2 is your loopback. Have you tried the same interface to which your SIP is bound?
>
>James Buchanan| UC Technology Manager |Presidio South |Presidio Networked Solutions
>12 Cadillac Dr Ste 130 Brentwood, TN 37027 |jbuchanan at presidio.com
>D: 615-866-5729 |F:615-866-5781 www.presidio.com
>
>From:Jay Stants [mailto:jaystants at rogers.com]
>Sent: Saturday, August 13, 2011 10:24 PM
>To: Buchanan, James; cisco-voip at puck.nether.net
>Subject: Re: [cisco-voip] Calls not forwarding over SIP trunk to Cue
>
>debug ccsip messages output
>
>xx/SIP/Msg/ccsipDisplayMsg:
>Received:
>INVITE sip:5856786019 at 74.74.255.254:56665 SIP/2.0
>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
>From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
>To: <sip:5856786019 at 74.74.255.254:56665>
>Contact: <sip:5856784306 at 74.63.41.218>
>Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
>CSeq: 102 INVITE
>User-Agent: VoIPMS/SERAST
>Max-Forwards: 70
>Remote-Party-ID: "+15856784306" <sip:5856784306 at 74.63.41.218>;privacy=off;screen=no
>Date: Sun, 14 Aug 2011 03:21:35 GMT
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>Supported: replaces
>Content-Type: application/sdp
>Content-Length: 285
>v=0
>o=root 2851 2851 IN IP4 74.63.41.218
>s=session
>c=IN IP4 74.63.41.218
>t=0 0
>m=audio 14492 RTP/AVP 0 18 101
>a=rtpmap:0 PCMU/8000
>a=rtpmap:18 G729/8000
>a=fmtp:18 annexb=no
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-16
>a=silenceSupp:off - - - -
>a=ptime:20
>a=sendrecv
>Aug 14 03:21:35.836: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
>Sent:
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
>From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
>To: <sip:5856786019 at 74.74.255.254:56665>
>Date: Sun, 14 Aug 2011 03:21:35 GMT
>Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
>CSeq: 102 INVITE
>Allow-Events: telephone-event
>Server: Cisco-SIPGateway/IOS-12.x
>Content-Length: 0
>
>Aug 14 03:21:35.836: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
>Sent:
>SIP/2.0 180 Ringing
>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
>From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
>To: <sip:5856786019 at 74.74..255.254:56665>;tag=64143B74-13D4
>Date: Sun, 14 Aug 2011 03:21:35 GMT
>Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
>CSeq: 102 INVITE
>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
>Allow-Events: telephone-event
>Contact: <sip:5856786019 at 74.74.255.254:5060>
>Server: Cisco-SIPGateway/IOS-12.x
>Content-Length: 0
>
>Aug 14 03:21:47.324: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>Received:
>CANCEL sip:5856786019 at 74.74.255.254:56665 SIP/2.0
>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
>From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
>To: <sip:5856786019 at 74.74.255.254:56665>
>Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
>CSeq: 102 CANCEL
>User-Agent: VoIPMS/SERAST
>Max-Forwards: 70
>Remote-Party-ID: "+15856784306" <sip:5856784306 at 74.63.41.218>;privacy=off;screen=no
>Content-Length: 0
>
>Aug 14 03:21:47.356: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
>Sent:
>SIP/2.0 200 OK
>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
>From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
>To: <sip:5856786019 at 74.74.255.254:56665>
>Date: Sun, 14 Aug 2011 03:21:47 GMT
>Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
>CSeq: 102 CANCEL
>Content-Length: 0
>
>Aug 14 03:21:47.356: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
>Sent:
>SIP/2.0 487 Request Cancelled
>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
>From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
>To: <sip:5856786019 at 74.74.255.254:56665>;tag=64143B74-13D4
>Date: Sun, 14 Aug 2011 03:21:47 GMT
>Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
>CSeq: 102 INVITE
>Allow-Events: telephone-event
>Server: Cisco-SIPGateway/IOS-12.x
>Reason: Q.850;cause=16
>Content-Length: 0
>
>Aug 14 03:21:47.400: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>Received:
>ACK sip:5856786019 at 74.74.255.254:56665 SIP/2.0
>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
>From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
>To: <sip:5856786019 at 74.74.255.254:56665>;tag=64143B74-13D4
>Contact: <sip:5856784306 at 74.63.41.218>
>Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218
>CSeq: 102 ACK
>User-Agent: VoIPMS/SERAST
>Max-Forwards: 70
>Remote-Party-ID: "+15856784306" <sip:5856784306 at 74.63.41.218>;privacy=off;screen=no
>Content-Length: 0
>
>
>
>
>Regards,
>Jay Stants
>jaystants at rogers.com
>
>
>
>From:"Buchanan, James" <jbuchanan at presidio.com>
>To: Jay Stants <jaystants at rogers.com>; "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
>Sent: Saturday, August 13, 2011 11:11:50 PM
>Subject: RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue
>Can you send a debug ccsip messages?
>
>James Buchanan| UC Technology Manager |Presidio South |Presidio Networked Solutions
>12 Cadillac Dr Ste 130 Brentwood, TN 37027 |jbuchanan at presidio.com
>D: 615-866-5729 |F:615-866-5781 www.presidio.com
>
>From:Jay Stants [mailto:jaystants at rogers.com]
>Sent: Saturday, August 13, 2011 10:02 PM
>To: Buchanan, James; cisco-voip at puck.nether.net
>Subject: Re: [cisco-voip] Calls not forwarding over SIP trunk to Cue
>
>no dice - still just rings infinantly ..
>
>Regards,
>Jay Stants
>jaystants at rogers.com
>From:"Buchanan, James" <jbuchanan at presidio.com>
>To: Jay Stants <jaystants at rogers.com>; "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
>Sent: Saturday, August 13, 2011 10:55:22 PM
>Subject: RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue
>Try adding b2bua onto your voicemail dial peer.
>
>James Buchanan| UC Technology Manager |Presidio South |Presidio Networked Solutions
>12 Cadillac Dr Ste 130 Brentwood, TN 37027 |jbuchanan at presidio.com
>D: 615-866-5729 |F:615-866-5781 www.presidio.com
>
>From:cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jay Stants
>Sent: Saturday, August 13, 2011 9:49 PM
>To: cisco-voip at puck.nether.net
>Subject: [cisco-voip] Calls not forwarding over SIP trunk to Cue
>
>Having an issue where calls coming in from ITSP ring and never fwd to voicemail. Internally voicemail works. either by dialing the vm number or by pushing the messages button which essentially just speed dials the 4000 extension used for voicemail. Can someone take a look at the included config and maybe point out what i'm missing to allow for calls to be fwd'ed to Cue after the specified time.
>
>voice service voip
> ip address trusted list
> ipv4 0.0.0.0 0.0.0.0
> allow-connections h323 to h323
> allow-connections h323 to sip
> allow-connections sip to h323
> allow-connections sip to sip
> supplementary-service h450.12
> sip
> registrar server expires max 600 min 60
>!
>!
>!
>!
>voice translation-rule 1
> rule 1 /^9/ //
>!
>voice translation-rule 3
> rule 1 /4.../ /5856786019/
>!
>!
>voice translation-profile voip.ms
> translate calling 3
> translate called 1
>dial-peer voice 1 voip
> description **SIP Trunk to newyork.voip.ms**
> translation-profile outgoing voip.ms
> destination-pattern 9[2-9].[2-9].......
> session protocol sipv2
> session target dns:newyork.voip.ms
> dtmf-relay rtp-nte sip-notify
> codec g711ulaw
> no vad
>!
>dial-peer voice 2 voip
> description **Incoming SIP Trunk - Voip.ms**
> translation-profile incoming voip.ms
> session protocol sipv2
> session target ipv4:10.50.1.2
> incoming called-number 5856786019
> dtmf-relay sip-notify
> codec g711ulaw
> no vad
>!
>dial-peer voice 20 pots
> destination-pattern 5.T
> direct-inward-dial
> no sip-register
>!
>dial-peer voice 3 voip
> description ** Voicemail **
> destination-pattern 4000
> session protocol sipv2
> session target ipv4:1.1.1.2
> dtmf-relay sip-notify
> codec g711ulaw
> no vad
>
>ephone-dn-template 1
> call-forward busy 4000
> call-forward noan 4000 timeout 18
>
>ephone-dn 1 dual-line
> number 4005 secondary 5856786019 no-reg
> label 4005
> name Jay Stants
> ephone-dn-template 1
>
>
>Regards,
>Jay Stants
>jaystants at rogers.com
>
>
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>
>
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