SIP Line Questions

CUCM integration with Glassfish communication server

2013-01-02T07:38:16Z

Hi, Is it possible to integrate Glassfish Communication server with Cisco CUCM? With Thanks and Regards, Huthesh

Automatic reply: New Message from Sanjeev Thallikar in Cisco Unified Commun

2012-11-07T15:13:48Z

Am currently booked on a special assignment offsite untill Nov 15th. Expect some delays in responding to emails. Best Regards, Mohamed Lubbad

RE: SIP Dialer and Call Progress Analysis

2012-11-07T15:12:48Z

[quote=Huthesha K]Hi, I am still trying to identify a solution. Can anyone please help me in this regards? Thanks, Huthesh [/quote] Old thread but thought this will be useful for somebody in the future http://www.cisco.com/image/gif/paws/111980/cpa-00.pdf

RE: SIP Dialer and Call Progress Analysis

2012-09-22T05:55:09Z

Hi, I am still trying to identify a solution. Can anyone please help me in this regards? Thanks, Huthesh

SIP Dialer and Call Progress Analysis

2012-09-17T11:45:51Z

Hi All, I am trying to develop a SIP dialer. I want to implement Call Progress Analysis (CPA). Please anyone tell me how to implement CPA using SIP. With Regards, Huthesh

list of certified/tested 3rd party SIP Endpoints under CDN?

2012-08-13T08:21:17Z

What can I find a list of 3rd party SIP endpoints that are compatible/tested with the different CUCM releases? TekVizion does not have the latest list since the CDN program is the way to go. thanks Mohamed Lubbad

CUCM 7 MOH 3rd Party SIP MOH scenario

2012-05-09T09:19:00Z

Hi We are testing interoperation against both CUCM 7.1 and 8.5(but do not have 8.5 yet). But we are having some trouble invoking MOH when our gateway is registering as a 3rd party SIP phone and initating a hold (we have the same issue with snom 300s too). The same operation performed by a 7941/7960 (SIP) the held party receives music on hold from the CUCM. The call signalling does appear to operate correctly, in that the hold reinvite sent is responded to, and the held party is commanded by the CUCM to renegotiate the RTP behaviour, after which the resume is also processed correctly and the two RTP is renegotiated. But this is missing the music on hold that is present if a Cisco IP Phone intiates the hold. We would like to ask if there are any special configuration options that need to be enabled to allow 3rd party SIP Phones to intiate a hold and the held party receive MOH until resumed. Any help much appreciated!

Do the 29xx gateways/IOS 15.X support SIP RFC 5626

2012-03-20T04:16:49Z

Specifically section 4.4.1 about keep alives on a connection oriented protocol such as TCP? When I turn on SIP message debugging I see them show up (from my UA), but there is never a response to them. Do I need to configure something?

Triggering on hold request in TCL/IVR script

2012-03-06T19:37:19Z

I have a TCL IVR script (mostly default handling), and I want to trigger on the start and stop hold events (that is, when one of my SIP phones requests to put the call on hold). I thought it was just an ev_feature event, but my TCL script is not receiving that event. Does anyone have any idea if that is not the correct event, or any other issue I may be having? Thanks!

SIP contact header in REGISTER

2012-03-06T19:36:32Z

I am trying to set up forwarding of a SIP phone to an external (PSTN) phone number using the SIP REGISTER (rather than the usual 302 Moved Temporarily response) for a rather unconventional application. Is there a way to register an external phone number as a contact number on the CISCO 29xx series integrated router (IOS 15.1(4)M1)? I've tried using the #############Tel URI in the Contact Header field of the SIP REGISTER and received a 400 Bad Request - 'Malformed/Missing Contact field". (I've tried every variation of the number and get the same response.) I've tried using the SIP URI in the Contact Header field, using the following format: [url=https://supportforums.cisco.com/mailto:TelephoneNumber@IP_address_of_Cisco_gateway]TelephoneNumber@IP_address_of_Cisco_gateway[/url] and receive 404 Not Found. In the debug messages, I'll see, "contact address matches own ip address" (but what IP address can I use for an external phone number?). Any suggestions on the best way to set up a REGISTER for a SIP phone to have the Contact header field containing an external (PSTN) number?

RE: Virtual Cisco Unified Communications Manager V8.0 on VMWare

2012-01-04T17:54:45Z

[quote]Hi David, Thank you very much. Do you know where can I get the price for CUCM 8.0 and Cisco server for running CUCM 8.0 and also Cisco PBX? Thanks Steven[/quote] Hi Steven, You can get the pricing of CUCM and all Cisco Products at cisco.com/go/ordering and choose for Dynamic Configuration Tool (CCO Login Required). for CUCM, the part code would be CUCM-USR-LIC, Hope this helps. Thanks, Manish

BLF status monitor to get SIP call-id of outgoing leg

2011-10-22T17:10:03Z

Hello I did some test sniffing (with wireshark) the NOTIFY even received by a CIPC(SIP) configured with BLF for an extension assigned to a CTI Port placing an outgoing call to reach a SIP device connected vis SIP Trunk. BLF status monitor config: SIP IPC <--BLF DN 4900 status monitor -> CUCM <---> CTI Port DN 4901 How outbound call is placed: CTI Port DN 4901 call to 3357454317 ----> Route Pattern for 3! --> SIP Trunk ---> PSTN GW What I realized is the NOTIFY providing BLF status does not contain the dialog-info I need, and the dialog-info should contain the call-id of the INVITE placed by the CTI Port. Is there a way to get it? Thanks Marco

DSCP

2011-09-20T21:03:47Z

With SCCP, the CUCM communicates the DSCP value for the end point to use for the media streams via the DSCPValue field in relevant messages (StationStartMultiMediaTransmissionMessage, RSVP... etc.). How does the CUCM communicate this info when interfacing with a SIP endpoint? SDP? I cannot locate any info regarding DSCP/Q0S in the Cisco SIP messaging guides. Do SIP Cisco end points simply hardcode EF for audio streams? Received the following answer from Cisco Developer Services: [color=#1f497d]Your statements are correct. [/color] - SCCP devices receive DSCP info on a per call basis in the SCCP protocol messages - SIP devices receive DSCP info at boot up time via their TFTP config file, and this value is static for all calls

SUBSCRIBE to a SPAXXX phone

2011-07-08T12:37:09Z

Hello, I would like to know if it is possible to SUBSCRIBE to a SPAXXX phone in order to get notification of events from an external app. I need to capture the SIP NOTIFY received by the phone. Thanks Marco

Telematrix Phones with Cisco Call manager 7.1

2011-05-25T07:46:28Z

[color=#004080]Dear Team,[/color] [color=#004080][/color] [color=#004080]We have Telematrix SIP phones with Cisco Call Manager. Telematrix phones are wrongly responding to Cancel message from Call Manager.[/color] [color=#004080][/color] [color=#004080]Below is the email from Telematrix and in Black there is reply from TAC. I am wordering if you guy can help me to rectify the issue.[/color] [color=#004080][/color] [color=#004080]After further analysis of the captures I have concluded that the CUCM is sending an incorrect CANCEL message and that is why it is rejected. The rejection is proper according to the RFCs. I suggest you contact Cisco regarding this problem. We will talk with our contacts at Cisco as well.[/color] The INVITE method can establish a dialog when it succeeds with a 2xx response. After a 101-199 response, an ¿early dialog¿ can be established. When an UPDATE or a CANCEL message is sent, it must be sent within the dialog (or early dialog in the case of UPDATE.) A dialog ID consists of a Call ID, a local tag and a remote tag. The local and remote tags are taken from the From and To headers of the message. I have attached a filtered version of your capture file log1. Packet 1 is the INVITE to the TMX phone. It contains the Call ID and From tag. Packet 5 is the 180 Ringing response. It contains the To tag and establishes the ¿early dialog¿. Packet 8 is the UPDATE request to the TMX phone. You can see that it includes the Call ID, the From tag and the To tag, thus identifying the early dialog. The CANCEL message in packet 8 contains a Call ID and From tag, but no To tag. Without the To tag, the early dialog cannot be identified and the proper response is a 481 Call Leg/Transaction Does Not Exist. See sections 9.1 and 9.2 of RFC 3261. You may notice, as I did, that the 100 response did not contain a To tag, but the 180 response did. Section 12.1 states that only 2xx and 101-199 responses will create a dialog. Thus, this is normal. It is likely to take a long time to get Cisco to fix this problem. Perhaps there are unapplied patches that would solve this issue. Possibly an upgrade would fix it, but that is also a big effort. Since this appears to be a bug in CANCEL, I no longer see any likelihood that a configuration change will fix it. It still might be useful to try and find one though. Cheers, Bob Dear Muhammed, Bob, I have been reading in the RFC 3261 section 9.1 and I see: The Request-URI, Call-ID, To, the numeric part of CSeq, and From header fields in the CANCEL request MUST be identical to those in the request being cancelled, [b][u]including tags.[/u][/b] Which I can understand that the CANCEL sent by CallManager should have identical values/tags to that in the INVITE. I see that it¿s complying with the RFC and I cannot escalate the issue to change this. It must be a bug in the phone firmware. ---------------------- Thanks & Regards, Sultan

Deleting/recreating CTL in CUCM

2011-03-21T16:31:00Z

Hi, My Callmanager Cluster was broken and created with other secondary. Now i want to recreate the CTL files. I have lost the etokens used while CUCM installation. Now when i try to connect with the new a pair of etokens, the CTL Client in workstation gives error that it needs the etoken used while installation for adding/deleting/updating the new change information related to the secondary. So if any of you have run into similar problem or by experience, IS there any other way of deleting the CTL and recreating it for a updated callamanger cluster with out the actual CTL used while first time installation. Thanks to the community.

3rd Party SIP Endpoint as UCCX/E Agent Phone

2011-02-01T23:59:25Z

Does Cisco provide access to the SIP line side extensions allowing a third party SIP softphone to function as an agent phone.

Detailed description of the Cisco 79x2 series .conf file

2011-01-20T18:40:03Z

All, I have a partner looking to integrate the 79x2 series handsets with a third-party PBX solution. They know the "official" line is that the 79x2 series phones are only supported on the UCM platform. However, this partner is confident that, given the appropriate resources, they can provide interoperability with their call control platform. The file in question is the {MAC Address}.xml.conf file. They have plenty of examples of "working" files (as generated by UCM). What the partner is looking for is a detailed description of the different fields that can be edited in the file, to include: - field description - valid entries (type, length, etc) Thanks in advance for the help. This will help us grow our education business tremendously. Best Regards, Chris Marin chmarin@cisco.com 703 405 2932

RE: Does Cisco SPA7940/7941 IP Phone works with third party open SIP server

2010-09-23T20:38:45Z

Hi, AFAIK, integration of 7940 IP Phones with third party SIP call manager is possible. I believe, you would have to use SIP/MGCP firware version of IP phones to achieve this. Further reference on this from http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/mgcp/firmware/matrix/frmwrup.html The Cisco IP Phone 7960 and 7940 platform has the ability to support three protocol application firmware versions: >SCCP - Used with Cisco Call Manager. >SIP - Used with Cisco SIP Proxy Server (CSPS), base transceiver station (BTS), third-party servers, or as a standalone entity. >MGCP - Used with third-party call agents (CAs).

RE: CUCM 7.1.3 sends its IP in 200 OK response of Register.which Config doe

2010-09-19T12:52:06Z

Hi, This is not an configuration issue. The contact headder sent while registering has to be sent back by CUCM in 200 OK request. AFAIK there seems to be no know issue of this sort. For investigating this further, I would like you to open a Developer support case. Kindly send by the steps to recreate the issue or CCM/CTI logs for invetigation. The following link will detail you on the CDN Support Services program http://developer.cisco.com/web/devservices/alldevs