Hi,
INFO is generated in early stage.
The information I provided in my original post was not entirely accurate.
My setup was as follows:
Carrier PRI ----- Siemens PBX (4000) ---(tried qsiq and primary-ni) ---
Cisco 3845 ---- PROXY
After more debugging we found out that Siemens was not passing the correct
information in the ISDN setup messages. Specifically, it was missing the
following:
"Operation = InformationFollowing (calling_name)
Name information in subsequent FACILITY message”
We then connected the carrier PRI directly into the media gateway and
that resolved the issue. Cisco gateway was already configured to act
similar to how Jeff described the Adtran configuration. We started
seeing the correct calling name as soon as the PRI was connected
directly.
Thank you for the followup.
Best regards,
--
Zahid
On Wed, Jan 6, 2016 at 4:49 AM, Bogdan-Andrei Iancu <bog...@opensips.org>
wrote:
> Hi Zahid,
>
> When in the dialog is the INFO injected ? in the early stage ? or after
> the call is established ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 18.12.2015 17:12, Zahid Mehmood wrote:
>
> Hi Jeff,
> Thanks for your response. >From what I understand from Cisco
> documentation, gateway seems to act that way when using H323 but, sadly,
> not for SIP. I will push Cisco about this.
>
> Assuming the worst, Is there any thing I can try on the proxy side to get
> the desired results?
>
> Regards,
> --
> Zahid
>
>
> On Fri, Dec 18, 2015 at 9:44 AM, Jeff Pyle <jeff.p...@fidelityvoice.com>
> wrote:
>
>> In Adtran TA900 series gateways (very Cisco-like) I'm able to configure
>> the PRI interface to wait for the FACILITY message before sending the
>> initial INVITE. When the INVITE does leave the gateway towards the proxy,
>> it has full caller name information. Perhaps something like this is
>> available on the Cisco. I hope so, because if not, you're going to have a
>> difficult time integrating the INFO message.
>>
>>
>> - Jeff
>>
>>
>> On Thu, Dec 17, 2015 at 2:53 PM, Zahid Mehmood < <z...@columbia.edu>
>> z...@columbia.edu> wrote:
>>
>>> Hi,
>>> I am having trouble figuring out how to process the calling-name
>>> coming from the PRI. In my setup, PRI is connected to a Cisco media gateway
>>> which sends traffic to the proxy servers. Calling name is not coming in
>>> the ISDN setup message. It is actually provided in a separate facility
>>> message [1].
>>>
>>> Cisco gateway processes this secondary messages and generates a INFO
>>> message. Polycom phone sends the 200 ok message but there is no change in
>>> the visible caller id.
>>>
>>> Does anyone have a working example or suggestion of how this is supposed
>>> to work?
>>>
>>> Invite:
>>>
>>> U 2015/12/17 14:20:31.215540 10.10.1.1:50975 -> 10.10.2.2:5060
>>> INVITE sip:10301@10.10.2.2:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3A2284.
>>> Remote-Party-ID: "1112223333" < <sip%3A1112223333@10.10.1.1>
>>> sip:1112223333@10.10.1.1>;party=calling;screen=yes;privacy=off.
>>> From: "1112223333" < <sip%3A1112223333@10.10.1.1>
>>> sip:1112223333@10.10.1.1>;tag=5745CCC-1C72.
>>> To: <sip:10301@10.10.2.2>.
>>> Date: Thu, 17 Dec 2015 19:20:31 GMT.
>>> Call-ID: 12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1.
>>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
>>> Min-SE: 1800.
>>> Cisco-Guid: 0311776101-2754220517-2148597785-1445067520.
>>> 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: 1450380031.
>>> Contact: <sip:1112223333@10.10.1.1:5060>.
>>> Expires: 180.
>>> Allow-Events: telephone-event.
>>> Content-Type: application/sdp.
>>> Content-Disposition: session;handling=required.
>>> Content-Length: 279.
>>> .
>>> v=0.
>>> o=CiscoSystemsSIP-GW-UserAgent 3918 6190 IN IP4 10.10.1.1.
>>> s=SIP Call.
>>> c=IN IP4 10.10.1.1.
>>> t=0 0.
>>> m=audio 18854 RTP/AVP 0 18 101.
>>> c=IN IP4 10.10.1.1.
>>> 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.
>>>
>>> Invite messages:
>>>
>>> U 2015/12/17 14:20:31.546310 10.10.1.1:50975 -> 10.10.2.2:5060
>>> INFO sip:10301@10.219.136.69:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3C1EC0.
>>> From: "1112223333" < <sip%3A1112223333@10.10.1.1>
>>> sip:1112223333@10.10.1.1>;tag=5745CCC-1C72.
>>> To: <sip:10301@10.10.2.2>;tag=1768D8EC-CCDB1323.
>>> Date: Thu, 17 Dec 2015 19:20:31 GMT.
>>> Call-ID: 12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1.
>>> User-Agent: Cisco-SIPGateway/IOS-12.x.
>>> Max-Forwards: 70.
>>> Route: <sip:10.10.2.2;lr=on;ftag=5745CCC-1C72>.
>>> Timestamp: 1450380031.
>>> CSeq: 103 INFO.
>>> Contact: <sip:1112223333@10.10.1.1:5060>.
>>> Remote-Party-ID: "WIRELESS CALLER" < <sip%3A1112223333@10.10.1.1>
>>> sip:1112223333@10.10.1.1>;party=calling;screen=no;privacy=off.
>>> Content-Length: 0.
>>> .
>>>
>>>
>>> Best Regards,
>>>
>>> --
>>> Zahid
>>>
>>> [1]
>>> http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-isdn.html#GUID-53D5C9AB-AAC4-4178-8158-0DAEFB5BC33E
>>> (figure 2 is close to what we are seeing)
>>>
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