[cisco-voip] [Bulk] RE: Calls not forwarding over SIP trunk to Cue

Thu Aug 18 14:52:28 EDT 2011

In your cue gui, put in the number in the E164 field that is redirecting to
cue.  Do a deb ccsip events to see.
On Aug 18, 2011 1:04 PM, "Jay Stants" <jaystants at rogers.com> wrote:
> Good day everyone,
>
> I've made changes to timers but results are the same - call comes in from
outside across sip trunk / DID and just rings infenently could someone
possibly look over the dial-peers i have built / translation rules and see
if i may need to add something else to allow for inbound call to redirect to
CUE after no answer timer is triggered.
>
> Thanks to all that have helped so far  - I appreciate the knowledge that
everyone is aiding with as i take the endevour into the Voip world.
>
> Regards,
> Jay Stants
> jaystants at rogers.com
>
>
> From: Nick Matthews <matthnick at gmail.com>
>>To: "jaystants at rogers.com" <jaystants at rogers.com>
>>Sent: Wednesday, August 17, 2011 6:21:27 PM
>>Subject: Re: [cisco-voip] [Bulk] RE: Calls not forwarding over SIP trunk
to Cue
>>
>>
>>In the debugs there is a cancel message 12 seconds after the invite, but
the ring no answer is 18 seconds.  Try not hanging up or reducing the cal
forward timer.  If it still doesn't work send debugs.
>>On Aug 14, 2011 9:12 AM, "jaystants at rogers.com" <jaystants at rogers.com>
wrote:
>>> I was thinking this, or else a dial-peer
>>> Can you provide an example?
>>>
>>> Regards,
>>>
>>> Sent from my HTC bolt of lightning...
>>>
>>> ----- Reply message -----
>>> From: "ccieid1ot" <ccieid1ot at gmail.com>
>>> To: "Jay Stants" <jaystants at rogers.com>
>>> Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>,
"Buchanan, James" <jbuchanan at presidio.com>
>>> Subject: [cisco-voip] [Bulk] RE: Calls not forwarding over SIP trunk to
Cue
>>> Date: Sun, Aug 14, 2011 3:23 am
>>> You might need a translation profile for redirecting number, unless you
have your DID as an E164.
>>> On Aug 13, 2011 11:23 PM, "Jay Stants" <jaystants at rogers.com> wrote:>
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
>>>>>
>>>>>
>>>>>_______________________________________________
>>>>>cisco-voip mailing list
>>>>>cisco-voip at puck.nether.net
>>>
>>>>>https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
>>>>>
>>
>>
>>
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