I was thinking this, or else a dial-peer <br>Can you provide an example?<br><br>Regards,<br><br>Sent from my HTC bolt of lightning...<br><br><div id="htc_header" style="">----- Reply message -----<br>From: "ccieid1ot" <ccieid1ot@gmail.com><br>To: "Jay Stants" <jaystants@rogers.com><br>Cc: "cisco-voip@puck.nether.net" <cisco-voip@puck.nether.net>, "Buchanan, James" <jbuchanan@presidio.com><br>Subject: [cisco-voip] [Bulk] RE: Calls not forwarding over SIP trunk to Cue<br>Date: Sun, Aug 14, 2011 3:23 am<br><br></div><br><p>You might need a translation profile for redirecting number, unless you have your DID as an E164.</p>
<div class="gmail_quote">On Aug 13, 2011 11:23 PM, "Jay Stants" <<a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a>> wrote:<br type="attribution">> Interface Details<br>> <br>> snet-wan2#sh ip int br<br>
> Interface IP-Address OK? Method Status Protocol<br>> FastEthernet0/0 10..50.1.2 YES NVRAM up up<br>> Service-Engine0/1 1.1.1.1 YES TFTP up up<br>
> FastEthernet0/1 74.74.255.254 YES DHCP up up<br>> GigabitEthernet1/0 10.1.99.1 YES NVRAM up up<br>> Loopback0 10.50.250.4 YES NVRAM up up<br>
> Loopback100 1.1.1.1 YES NVRAM up up<br>> NVI0 10.50.1.2 YES unset up up<br>> <br>> <br>> interface Service-Engine0/1<br>
> ip unnumbered Loopback100<br>> service-module ip address 1.1.1.2 255.255.255.252<br>> service-module ip default-gateway 1.1.1.1<br>> <br>> <br>> Regards,<br>> Jay Stants<br>> <a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a><br>
> <br>> <br>> From: "<a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a>" <<a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a>><br>>>To: "Buchanan, James" <<a href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a>>; "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
>>Sent: Sunday, August 14, 2011 12:03:36 AM<br>>>Subject: Re: [cisco-voip] [Bulk] RE: Calls not forwarding over SIP trunk to Cue<br>>><br>>><br>>>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<br>
>><br>>>Sent from my HTC bolt of lightning...<br>>><br>>><br>>>----- Reply message -----<br>>>From: "Buchanan, James" <<a href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a>><br>
>>To: "Jay Stants" <<a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a>>, "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
>>Subject: [Bulk] RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue<br>>>Date: Sat, Aug 13, 2011 11:48 pm<br>>><br>>><br>>><br>>>I assume 1.1.1.2 is your loopback. Have you tried the same interface to which your SIP is bound?<br>
>> <br>>>James Buchanan| UC Technology Manager |Presidio South |Presidio Networked Solutions <br>>>12 Cadillac Dr Ste 130 Brentwood, TN 37027 |<a href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a><br>
>>D: 615-866-5729 |F:615-866-5781 <a href="http://www.presidio..com">www.presidio.com</a><br>>> <br>>>From:Jay Stants [mailto:<a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a>] <br>>>Sent: Saturday, August 13, 2011 10:24 PM<br>
>>To: Buchanan, James; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>>>Subject: Re: [cisco-voip] Calls not forwarding over SIP trunk to Cue<br>>> <br>>>debug ccsip messages output<br>
>> <br>>>xx/SIP/Msg/ccsipDisplayMsg:<br>>>Received:<br>>>INVITE <a href="http://sip:5856786019@74.74.255.254:56665">sip:5856786019@74.74.255.254:56665</a> SIP/2.0<br>>>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport<br>
>>From: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;tag=as29f9104d<br>>>To: <<a href="http://sip:5856786019@74.74.255.254:56665">sip:5856786019@74.74.255.254:56665</a>><br>
>>Contact: <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>><br>>>Call-ID: <a href="mailto:13b1851f78dee7904fd538ef4265d5a7@74.63.41.218">13b1851f78dee7904fd538ef4265d5a7@74.63.41.218</a><br>
>>CSeq: 102 INVITE<br>>>User-Agent: VoIPMS/SERAST<br>>>Max-Forwards: 70<br>>>Remote-Party-ID: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;privacy=off;screen=no<br>
>>Date: Sun, 14 Aug 2011 03:21:35 GMT<br>>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>>>Supported: replaces<br>>>Content-Type: application/sdp<br>>>Content-Length: 285<br>
>>v=0<br>>>o=root 2851 2851 IN IP4 74.63.41.218<br>>>s=session<br>>>c=IN IP4 74.63.41.218<br>>>t=0 0<br>>>m=audio 14492 RTP/AVP 0 18 101<br>>>a=rtpmap:0 PCMU/8000<br>>>a=rtpmap:18 G729/8000<br>
>>a=fmtp:18 annexb=no<br>>>a=rtpmap:101 telephone-event/8000<br>>>a=fmtp:101 0-16<br>>>a=silenceSupp:off - - - -<br>>>a=ptime:20<br>>>a=sendrecv<br>>>Aug 14 03:21:35.836: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:<br>
>>Sent:<br>>>SIP/2.0 100 Trying<br>>>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport<br>>>From: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;tag=as29f9104d<br>
>>To: <<a href="http://sip:5856786019@74.74.255.254:56665">sip:5856786019@74.74.255.254:56665</a>><br>>>Date: Sun, 14 Aug 2011 03:21:35 GMT<br>>>Call-ID: <a href="mailto:13b1851f78dee7904fd538ef4265d5a7@74.63.41.218">13b1851f78dee7904fd538ef4265d5a7@74.63.41.218</a><br>
>>CSeq: 102 INVITE<br>>>Allow-Events: telephone-event<br>>>Server: Cisco-SIPGateway/IOS-12.x<br>>>Content-Length: 0<br>>><br>>>Aug 14 03:21:35.836: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:<br>
>>Sent:<br>>>SIP/2.0 180 Ringing<br>>>Via: SIP/2.0/UDP 74..63.41.218:5060;branch=z9hG4bK5c9fe01a;rport<br>>>From: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;tag=as29f9104d<br>
>>To: <sip:5856786019@74.74..255.254:56665>;tag=64143B74-13D4<br>>>Date: Sun, 14 Aug 2011 03:21:35 GMT<br>>>Call-ID: <a href="mailto:13b1851f78dee7904fd538ef4265d5a7@74.63.41.218">13b1851f78dee7904fd538ef4265d5a7@74.63.41.218</a><br>
>>CSeq: 102 INVITE<br>>>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER<br>>>Allow-Events: telephone-event<br>>>Contact: <<a href="http://sip:5856786019@74.74.255.254:5060">sip:5856786019@74.74.255.254:5060</a>><br>
>>Server: Cisco-SIPGateway/IOS-12.x<br>>>Content-Length: 0<br>>><br>>>Aug 14 03:21:47.324: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>>>Received:<br>>>CANCEL <a href="http://sip:5856786019@74.74.255.254:56665">sip:5856786019@74.74.255.254:56665</a> SIP/2.0<br>
>>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport<br>>>From: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;tag=as29f9104d<br>>>To: <<a href="http://sip:5856786019@74.74.255.254:56665">sip:5856786019@74.74.255.254:56665</a>><br>
>>Call-ID: <a href="mailto:13b1851f78dee7904fd538ef4265d5a7@74.63.41.218">13b1851f78dee7904fd538ef4265d5a7@74.63.41.218</a><br>>>CSeq: 102 CANCEL<br>>>User-Agent: VoIPMS/SERAST<br>>>Max-Forwards: 70<br>
>>Remote-Party-ID: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;privacy=off;screen=no<br>>>Content-Length: 0<br>>><br>>>Aug 14 03:21:47.356: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:<br>
>>Sent:<br>>>SIP/2.0 200 OK<br>>>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport<br>>>From: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;tag=as29f9104d<br>
>>To: <<a href="http://sip:5856786019@74.74.255.254:56665">sip:5856786019@74.74.255.254:56665</a>><br>>>Date: Sun, 14 Aug 2011 03:21:47 GMT<br>>>Call-ID: <a href="mailto:13b1851f78dee7904fd538ef4265d5a7@74.63.41.218">13b1851f78dee7904fd538ef4265d5a7@74.63.41.218</a><br>
>>CSeq: 102 CANCEL<br>>>Content-Length: 0<br>>><br>>>Aug 14 03:21:47.356: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:<br>>>Sent:<br>>>SIP/2.0 487 Request Cancelled<br>>>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport<br>
>>From: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;tag=as29f9104d<br>>>To: <<a href="http://sip:5856786019@74.74.255.254:56665">sip:5856786019@74.74.255.254:56665</a>>;tag=64143B74-13D4<br>
>>Date: Sun, 14 Aug 2011 03:21:47 GMT<br>>>Call-ID: <a href="mailto:13b1851f78dee7904fd538ef4265d5a7@74.63.41.218">13b1851f78dee7904fd538ef4265d5a7@74.63.41.218</a><br>>>CSeq: 102 INVITE<br>>>Allow-Events: telephone-event<br>
>>Server: Cisco-SIPGateway/IOS-12.x<br>>>Reason: Q.850;cause=16<br>>>Content-Length: 0<br>>><br>>>Aug 14 03:21:47.400: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>>>Received:<br>>>ACK <a href="http://sip:5856786019@74.74.255.254:56665">sip:5856786019@74.74.255.254:56665</a> SIP/2.0<br>
>>Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport<br>>>From: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;tag=as29f9104d<br>>>To: <<a href="http://sip:5856786019@74.74.255.254:56665">sip:5856786019@74.74.255.254:56665</a>>;tag=64143B74-13D4<br>
>>Contact: <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>><br>>>Call-ID: <a href="mailto:13b1851f78dee7904fd538ef4265d5a7@74.63.41.218">13b1851f78dee7904fd538ef4265d5a7@74.63.41.218</a><br>
>>CSeq: 102 ACK<br>>>User-Agent: VoIPMS/SERAST<br>>>Max-Forwards: 70<br>>>Remote-Party-ID: "+15856784306" <<a href="mailto:sip%3A5856784306@74.63.41.218">sip:5856784306@74.63.41.218</a>>;privacy=off;screen=no<br>
>>Content-Length: 0<br>>><br>>><br>>> <br>>> <br>>>Regards,<br>>>Jay Stants<br>>><a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a><br>>><br>>><br>>><br>
>>From:"Buchanan, James" <<a href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a>><br>>>To: Jay Stants <<a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a>>; "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
>>Sent: Saturday, August 13, 2011 11:11:50 PM<br>>>Subject: RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue<br>>>Can you send a debug ccsip messages?<br>>> <br>>>James Buchanan| UC Technology Manager |Presidio South |Presidio Networked Solutions <br>
>>12 Cadillac Dr Ste 130 Brentwood, TN 37027 |<a href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a><br>>>D: 615-866-5729 |F:615-866-5781 <a href="http://www.presidio.com">www.presidio.com</a><br>
>> <br>>>From:Jay Stants [mailto:<a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a>] <br>>>Sent: Saturday, August 13, 2011 10:02 PM<br>>>To: Buchanan, James; <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
>>Subject: Re: [cisco-voip] Calls not forwarding over SIP trunk to Cue<br>>> <br>>>no dice - still just rings infinantly .. <br>>> <br>>>Regards,<br>>>Jay Stants<br>>><a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a><br>
>>From:"Buchanan, James" <<a href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a>><br>>>To: Jay Stants <<a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a>>; "<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>" <<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
>>Sent: Saturday, August 13, 2011 10:55:22 PM<br>>>Subject: RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue<br>>>Try adding b2bua onto your voicemail dial peer. <br>>> <br>>>James Buchanan| UC Technology Manager |Presidio South |Presidio Networked Solutions <br>
>>12 Cadillac Dr Ste 130 Brentwood, TN 37027 |<a href="mailto:jbuchanan@presidio.com">jbuchanan@presidio.com</a><br>>>D: 615-866-5729 |F:615-866-5781 <a href="http://www.presidio.com">www.presidio.com</a><br>
>> <br>>><a href="mailto:From%3Acisco-voip-bounces@puck.nether.net">From:cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Jay Stants<br>
>>Sent: Saturday, August 13, 2011 9:49 PM<br>>>To: <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>>>Subject: [cisco-voip] Calls not forwarding over SIP trunk to Cue<br>>> <br>
>>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.<br>
>> <br>>>voice service voip<br>>> ip address trusted list<br>>> ipv4 0.0.0.0 0.0.0.0<br>>> allow-connections h323 to h323<br>>> allow-connections h323 to sip<br>>> allow-connections sip to h323<br>
>> allow-connections sip to sip<br>>> supplementary-service h450.12<br>>> sip<br>>> registrar server expires max 600 min 60<br>>>!<br>>>!<br>>>!<br>>>!<br>>>voice translation-rule 1<br>
>> rule 1 /^9/ //<br>>>!<br>>>voice translation-rule 3<br>>> rule 1 /4.../ /5856786019/<br>>>!<br>>>!<br>>>voice translation-profile <a href="http://voip.ms">voip.ms</a><br>>> translate calling 3<br>
>> translate called 1<br>>>dial-peer voice 1 voip<br>>> description **SIP Trunk to <a href="http://newyork.voip.ms">newyork.voip.ms</a>**<br>>> translation-profile outgoing <a href="http://voip.ms">voip.ms</a><br>
>> destination-pattern 9[2-9].[2-9].......<br>>> session protocol sipv2<br>>> session target dns:<a href="http://newyork.voip.ms">newyork.voip.ms</a><br>>> dtmf-relay rtp-nte sip-notify<br>>> codec g711ulaw<br>
>> no vad<br>>>!<br>>>dial-peer voice 2 voip<br>>> description **Incoming SIP Trunk - Voip.ms**<br>>> translation-profile incoming <a href="http://voip.ms">voip.ms</a><br>>> session protocol sipv2<br>
>> session target ipv4:10.50.1.2<br>>> incoming called-number 5856786019<br>>> dtmf-relay sip-notify<br>>> codec g711ulaw<br>>> no vad<br>>>!<br>>>dial-peer voice 20 pots<br>>> destination-pattern 5.T<br>
>> direct-inward-dial<br>>> no sip-register<br>>>!<br>>>dial-peer voice 3 voip<br>>> description ** Voicemail **<br>>> destination-pattern 4000<br>>> session protocol sipv2<br>>> session target ipv4:1.1.1.2<br>
>> dtmf-relay sip-notify<br>>> codec g711ulaw<br>>> no vad<br>>> <br>>>ephone-dn-template 1<br>>> call-forward busy 4000<br>>> call-forward noan 4000 timeout 18<br>>> <br>
>>ephone-dn 1 dual-line<br>>> number 4005 secondary 5856786019 no-reg<br>>> label 4005<br>>> name Jay Stants<br>>> ephone-dn-template 1<br>>> <br>>> <br>>>Regards,<br>>>Jay Stants<br>
>><a href="mailto:jaystants@rogers.com">jaystants@rogers.com</a><br>>> <br>>> <br>>>_______________________________________________<br>>>cisco-voip mailing list<br>>><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
>><a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>>><br>>><br>>><br></div>