06-20-2012 06:31 AM - edited 03-16-2019 11:46 AM
We have a branch site with SIP trunk which connects back to HQ CUCM (6.0.1.3000-7) via SIP. (Upgrading to CUCM 8.3 btw )
There are also four PSTN lines at the site as a backup to be used in SRST fail-over mode. All Cisco phones on the site are registered back to HQ CUCM. Customers can make outbound calls and receive inbound calls via SIP trunk without any issue. However, forwarding calls are going out via backup PSTN lines matching one of those dial-peers although they have preference 2.
dial-peer voice 10 voip
description Inbound - SIP from CUCM
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 1
no voice-class sip outbound-proxy
dtmf-relay rtp-nte
fax protocol none
no vad
!
dial-peer voice 22 pots
trunkgroup ALL_FXO
description Outbound - POTS to PSTN
translation-profile outgoing OUTBOUND_STRIPZERO
preference 2
destination-pattern .T
!
dial-peer voice 20 voip
description Outbound - SIP to PSTN
preference 1
destination-pattern .T
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 1
dtmf-relay rtp-nte
fax protocol none
!
I got the following if I call to an extension on the site where I have CForwardAll on that extension to a mobile 0417171234.
SiteAr001#sh voice call status
CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dial-peers
0x7C282 3C2 0x3118B310 0/3/2 0/1:1 *417171234 g711ulaw 10/22
1 active call found
!
Can someone please advise if this can be dial-peer matching issue due to my limited knowledge on dial-peers or possible SIP provider issue on diversion? Thanks much in advance for your sharing.
Regards,
Lay
06-20-2012 06:49 AM
Please post "debug ccsip messages"
Make sure the carrier allows you to send calls with caller ID not belonging to the SIP trunk.
Chris
06-20-2012 06:52 AM
This is most likely as issue on diversion to your SIP provider, hence when that failed, the gateway routed the call through the other dial-peer.
Can you send a
debug ccsip messages.
Please let me know what the calling and called number is and possible the diverted number.
With diversion to sip providers, if the Mask of the diverted number is not in your DDI range your provider will reject the call. Funny enough I resolved a similar problem a few hours ago
Please rate useful posts
"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-20-2012 08:11 PM
Thanks guys for your replies. Yes, it makes sense to me that SIP provider is not allowing the diverted calls.
Here is output:
SiteAr001#debug ccsip message
SIP Call messages tracing is enabled
SiteAr001#
Jun 21 09:41:15 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0417171234@172.17.83.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d
Remote-Party-ID: "Alan Rhodes" <6122>;party=calling;screen=yes;privacy=off6122>
From: "Alan Rhodes" <6122>;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-271398306122>
To: <0417171234>0417171234>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 2f0eb500-fe217bbb-f967-32010a0a@10.10.1.50
Supported: timer,replaces
Min-SE: 1800
User-Agent: Cisco-CCM6.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH
CSeq: 101 INVITE
Contact: <6122>6122>
Expires: 180
Allow-Events: presence
Session-Expires: 1800
Max-Forwards: 70
Content-Length: 0
Jun 21 09:41:15 AWST: //509764/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d
From: "Alan Rhodes" <6122>;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-271398306122>
To: <0417171234>0417171234>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 2f0eb500-fe217bbb-f967-32010a0a@10.10.1.50
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Jun 21 09:41:15 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0417171234@as.nipt.telstra.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4
Remote-Party-ID: "Alan Rhodes" <>>6122@ourdomain.com>;party=calling;screen=yes;privacy=off
From: "Alan Rhodes" <>>6122@as.nipt.telstra.com>;tag=F8E756B0-11E7
To: <>>0417171234@as.nipt.telstra.com>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 5E6ED00-BA7911E1-B77C8642-AEA3559B@172.17.83.2
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0098940080-3128496609-3077998146-2929939867
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1340242875
Contact: <6122>6122>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
Jun 21 09:41:15 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0417171234@as.nipt.telstra.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4
Remote-Party-ID: "Alan Rhodes" <>>6122@ourdomain.com>;party=calling;screen=yes;privacy=off
From: "Alan Rhodes" <>>6122@as.nipt.telstra.com>;tag=F8E756B0-11E7
To: <>>0417171234@as.nipt.telstra.com>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 5E6ED00-BA7911E1-B77C8642-AEA3559B@172.17.83.2
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0098940080-3128496609-3077998146-2929939867
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1340242875
Contact: <6122>6122>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
Jun 21 09:41:16 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0417171234@as.nipt.telstra.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4
Remote-Party-ID: "Alan Rhodes" <>>6122@ourdomain.com>;party=calling;screen=yes;privacy=off
From: "Alan Rhodes" <>>6122@as.nipt.telstra.com>;tag=F8E756B0-11E7
To: <>>0417171234@as.nipt.telstra.com>
Date: Thu, 21 Jun 2012 01:41:16 GMT
Call-ID: 5E6ED00-BA7911E1-B77C8642-AEA3559B@172.17.83.2
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0098940080-3128496609-3077998146-2929939867
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1340242876
Contact: <6122>6122>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
Jun 21 09:41:18 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0417171234@as.nipt.telstra.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4
Remote-Party-ID: "Alan Rhodes" <>>6122@ourdomain.com>;party=calling;screen=yes;privacy=off
From: "Alan Rhodes" <>>6122@as.nipt.telstra.com>;tag=F8E7645C-2386
To: <>>0417171234@as.nipt.telstra.com>
Date: Thu, 21 Jun 2012 01:41:18 GMT
Call-ID: 5E6ED00-BA7911E1-B77C8642-AEA3559B@172.17.83.2
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0098940080-3128496609-3077998146-2929939867
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1340242878
Contact: <6122>6122>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
Jun 21 09:41:19 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0417171234@as.nipt.telstra.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4
Remote-Party-ID: "Alan Rhodes" <>>6122@ourdomain.com>;party=calling;screen=yes;privacy=off
From: "Alan Rhodes" <>>6122@as.nipt.telstra.com>;tag=F8E7645C-2386
To: <>>0417171234@as.nipt.telstra.com>
Date: Thu, 21 Jun 2012 01:41:19 GMT
Call-ID: 5E6ED00-BA7911E1-B77C8642-AEA3559B@172.17.83.2
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0098940080-3128496609-3077998146-2929939867
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1340242879
Contact: <6122>6122>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
Jun 21 09:41:20 AWST: //509765/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0417171234@as.nipt.telstra.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4
Remote-Party-ID: "Alan Rhodes" <>>6122@ourdomain.com>;party=calling;screen=yes;privacy=off
From: "Alan Rhodes" <>>6122@as.nipt.telstra.com>;tag=F8E7645C-2386
To: <>>0417171234@as.nipt.telstra.com>
Date: Thu, 21 Jun 2012 01:41:20 GMT
Call-ID: 5E6ED00-BA7911E1-B77C8642-AEA3559B@172.17.83.2
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0098940080-3128496609-3077998146-2929939867
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1340242880
Contact: <6122>6122>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
Jun 21 09:41:23 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:as.nipt.telstra.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838BA1AC3
From: <.>*@as.nipt.telstra.com>;tag=F8E776EC-A56
To: <.>*@as.nipt.telstra.com>
Date: Thu, 21 Jun 2012 01:41:23 GMT
Call-ID: B102CC6A-B9E111E1-B5048642-AEA3559B
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1340242883
CSeq: 363 REGISTER
Contact: <.>
Expires: 3600
Supported: path
Content-Length: 0
Jun 21 09:41:23 AWST: //509767/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838BA1AC3
From: <.>*@as.nipt.telstra.com>;tag=F8E776EC-A56
To: <.>*@as.nipt.telstra.com>;tag=454442420-1340242883452
Call-ID: B102CC6A-B9E111E1-B5048642-AEA3559B
Timestamp: 1340242883
CSeq: 363 REGISTER
Content-Length: 0
Jun 21 09:41:25 AWST: //509764/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d
From: "Alan Rhodes" <6122>;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-271398306122>
To: <0417171234>;tag=F8E77F70-1C980417171234>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 2f0eb500-fe217bbb-f967-32010a0a@10.10.1.50
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <417171234>;party=called;screen=no;privacy=off417171234>
Contact: <0417171234>0417171234>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 303
v=0
o=CiscoSystemsSIP-GW-UserAgent 3809 6909 IN IP4 172.17.83.2
s=SIP Call
c=IN IP4 172.17.83.2
t=0 0
m=audio 23036 RTP/AVP 8 0 18 101
c=IN IP4 172.17.83.2
a=rtpmap:8 PCMA/8000
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
Jun 21 09:41:25 AWST: //509764/05E5B4B0B776/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d
From: "Alan Rhodes" <6122>;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-271398306122>
To: <0417171234>;tag=F8E77F70-1C980417171234>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 2f0eb500-fe217bbb-f967-32010a0a@10.10.1.50
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <417171234>;party=called;screen=no;privacy=off417171234>
Contact: <0417171234>0417171234>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Require: timer
Session-Expires: 1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 303
v=0
o=CiscoSystemsSIP-GW-UserAgent 3809 6909 IN IP4 172.17.83.2
s=SIP Call
c=IN IP4 172.17.83.2
t=0 0
m=audio 23036 RTP/AVP 8 0 18 101
c=IN IP4 172.17.83.2
a=rtpmap:8 PCMA/8000
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
Jun 21 09:41:25 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0417171234@172.17.83.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a735933bd05
From: "Alan Rhodes" <6122>;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-271398306122>
To: <0417171234>;tag=F8E77F70-1C980417171234>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 2f0eb500-fe217bbb-f967-32010a0a@10.10.1.50
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 209
v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 10.10.1.50
s=SIP Call
c=IN IP4 10.6.120.17
t=0 0
m=audio 26048 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Jun 21 09:41:36 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:0417171234@172.17.83.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a7451c27407
From: "Alan Rhodes" <6122>;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-271398306122>
To: <0417171234>;tag=F8E77F70-1C980417171234>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 2f0eb500-fe217bbb-f967-32010a0a@10.10.1.50
User-Agent: Cisco-CCM6.0
Max-Forwards: 70
CSeq: 102 BYE
Content-Length: 0
Jun 21 09:41:36 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a7451c27407
From: "Alan Rhodes" <6122>;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-271398306122>
To: <0417171234>;tag=F8E77F70-1C980417171234>
Date: Thu, 21 Jun 2012 01:41:36 GMT
Call-ID: 2f0eb500-fe217bbb-f967-32010a0a@10.10.1.50
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=534,OS=85440,PR=534,OR=85440,PL=0,JI=0,LA=0,DU=10
Content-Length: 0
But I have tried
voice class sip-profiles 1
response ANY sip-header Diversion remove
request ANY sip-header Diversion remove
it didn't make any difference?
Thanks.
06-21-2012 12:20 AM
Looks like I am having a bigger issue, customers cannot make outbound calls via SIP trunk failing over to backup PSTN lines. Can someone please share what you think?
06-21-2012 01:01 AM
From your trace..Here is the analysis..
+++CUBE receives an Invite for 0417171234+++
INVITE sip:0417171234@172.17.83.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.50:5060;branch=z9hG4bK23a727214430d
Remote-Party-ID: "Alan Rhodes" <6122>;party=calling;screen=yes;privacy=off6122>
From: "Alan Rhodes" <6122>;tag=3cf9bdf1-32ba-4e0c-9392-584cf24ea6f2-271398306122>
To: <0417171234>0417171234>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 2f0eb500-fe217bbb-f967-32010a0a@10.10.1.50
+++After cube sent a trying to your ITSP++++
++CUBE sends an Invite to a device called as.nipt.telstra.com++
Sent:
INVITE sip:0417171234@as.nipt.telstra.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838B91EA4
Remote-Party-ID: "Alan Rhodes" <>>6122@ourdomain.com>;party=calling;screen=yes;privacy=off
From: "Alan Rhodes" <>>6122@as.nipt.telstra.com>;tag=F8E756B0-11E7
To: <>>0417171234@as.nipt.telstra.com>
Date: Thu, 21 Jun 2012 01:41:15 GMT
Call-ID: 5E6ED00-BA7911E1-B77C8642-AEA3559B@172.17.83.2
This invite was sent 6 times and CUBE never got a response.
after the invite sent a registration was also sent and there was an error 404 not found was received for the registration request
Sent:
REGISTER sip:as.nipt.telstra.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838BA1AC3
From: <.>*@as.nipt.telstra.com>;tag=F8E776EC-A56
To: <.>*@as.nipt.telstra.com>
Date: Thu, 21 Jun 2012 01:41:23 GMT
Received:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 172.17.83.2:5060;branch=z9hG4bK838BA1AC3
From: <.>*@as.nipt.telstra.com>;tag=F8E776EC-A56
Question is this what is as.nipt.telstra.com? From the trace it looks like its your ITSP.
is 0417171234 not a device on your cucm? Why are you sending the call back to your ITSP? For calls to your internal extension shouldnt you be sending it to your CUCM?
Please rate useful posts
"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-21-2012 02:02 AM
Thanks for your reply. Correction: Customers can make outbound calls via SIP trunk, I was having "
no voice-class sip outbound-proxy" on TPIT dial-peer.
However diversion is still an issue. as.nipt.telstra.com is our ITSP provider. 0417171234 is a mobile number that the customer want the incoming call forward to. Provider confirmed that they won't allow to send calls with caller ID not belonging to the SIP trunk. Can you pls advise how the caller ID can be set/modify on the diversion?
Cheers,
Lay
06-21-2012 04:29 AM
We cna use sip profiles to modify that. But before we do that the trace you sent me does not show the original number called. The trace with the 047 number does not show that this is a diversion as the diverison header is missing. Did you send this trace after you configured the headers to be removed. Can you please remove the config from the sip profile.
Please send a full trace. Let me know what the calling and called number is. For us to modify the diversion headers I also need to know what your DDI is and your internal DN range.
Finally can you send a sh run. Please put it in a text file and attach here. What type of gateway do you have configured in cucm? Is it a sip trunk or h323 gateway?
Please rate useful posts
"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-24-2012 05:53 PM
Thanks aoknlawon for your sharing. I have done debug ccsip message again without modifying anything. Diversion happens matching dial-peer that uses backup PSTN lines. Please have a look into attached outputs if it makes sense to you.
Calling Number: 04 6778 4687 (Mobile number - third party)
Called Number: 08 9290 6198 (One of DID belonged to this SIP Trunk - range 08 9290 61XX)
Forwarded Number: 04 4857 4470 (Mobile number - third party)
I have SIP trunk in CUCM for this site. Thanks again for your sharing.
Cheers,
Lay
06-25-2012 03:34 AM
Hi,
Add this to your voice class sip-profles 1 and then test again
voice class sip-profile 1
request INVITE sip-header From modify "<>>6198@as.nipt.telstra.com>" "<0892906198>"0892906198>
Send me a debug ccsip messages after adding the command. Let me know if it works. I belive it should work with this. If it works we will then need to configure a profile that will match all your DN for diverted calls.
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
06-26-2012 05:56 AM
Thanks very much for that, aokanlawon. We have recently resolved this with an additional translation rule in the dial-peer.
voice translation-rule 3
rule 1 /^\(61..\)$/ /89290\1/
rule 2 /^\(....\)$/ /892906100/
voice translation-profile INBOUND_FXS_N_DIVERSION
translate calling 3
dial-peer voice 20 voip
description Outbound - SIP to PSTN
translation-profile outgoing INBOUND_FXS_N_DIVERSION
preference 1
destination-pattern .T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte
fax protocol none
I have also tried what you advised adding request INVITE sip-header From modify "<>6198@as.nipt.telstra.com>" "<0892906198>" in voice class sip-profile since it seems to be one of better method to resolve this. It wasn't still working, please see attached output, I wonder if you have experienced a case the provider expects something elese, other than sip-header From? Thanks again for your sharing.0892906198>>
Cheers, Lay
06-26-2012 06:30 AM
Hi,
From your xaltion rule it showed that the number you send to your ITSP is 892906198 (without a 0 in front)
So to use my method change your profile config to the one below and test again
request INVITE sip-header From modify "<>6198@as.nipt.telstra.com>" "<892906198>"892906198>>
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"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"
09-17-2012 12:00 AM
Hi Guys,
i had the same issue and we are also on Telstras TIPT network. The TIPT network was just seeing our 4 digit extensions for calls forwarded from internal CUCM phones
In the end the translation rule worked for me, its funny in there deployment guide they dont tell you these things
//snip
voice translation-rule 1
rule 1 /^\(72..\)$/ /89218\1/
voice translation-profile Call_Forward_From_CUCM
translate calling 1
dial-peer voice 30 voip (on the outgoing dial peer if just copied the relevant bit)
description ## outgoing tipt ##
translation-profile outgoing Call_Forward_From_CUCM
//snip
Cheers,
Nick
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