cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1241
Views
0
Helpful
12
Replies

CME and a third-partty SIP Application

biesselillo
Level 1
Level 1

Hello to everyone,

I'm new in Cisco world.

I have installed into my office a CME and configured it to communicate with SIP protocol with its ip phones (7961, 7911, 7941).

Then I also have registered on the CME a third-party application which simply recevies a call from the sip phone model 7961 and interconnect it to the other sip phone 7941.

The registration is ok, the application is correctly registered on the CME.

When I make a call from the 7961 I can see that the application answers but on the wireshark traces I see that there is a 488 Not Acceptable Here coming from my Application.

So in order to investigate I also tried to register at first an Eyebeam, the registration is ok, but when I try to call it when I answer the line is hunged-up. If I make a call from the Eyebeam to call the cisco ip phone, I see the message on the Eyebeam saying "Unsopported media type".

So I tried to investigate the codecs, and I configured on the cisco ip phone to use the G.711Alaw, and the same on the eybeam, but the resutl is the same.

So I tried to configure the codec G.729 on both and the problem is still the same.

So in conclusion I'm afraid that my Application when answers the call coming from the cisco ip phone, hs the same behaviour, and so the line is hunged-up.

Please help me!

Thank you very much in advance!

12 Replies 12

paolo bevilacqua
Hall of Fame
Hall of Fame

Hi, please collect "debug ccsip message" with "term mon" when trying a call and send it here.

Thank you very much!

I've done the trace trying to call from a cisco ip phone 7961 to a Aastra ip sip phone 480i which is correctly registered on the CME.

Attached you'll find the trace.

Anyway, I'm trying to implement this because my task is to integrate out little callcenter solution on the CME.

So I'm not sure that this is the correct way to proceed, but I'm sure that the sip signalling and the sip voice stream should work, then I have to understand the best way to attach the CallCenter Application to the CME, because I need that 10-20-30 or more calls must be routed from the CME to the SIP Callcneter Application which should route this requests to the appropriate available cisco sip phones registered on the CME.

But this is the second step, at now I have to set up a correct audio flow.

Hi, your call fails with cause code 65 (codec mismatch) because the aastra phone wants g729 and the router evidently is not configured for that. Configure g711u everywhere and you should be fine.

Regarding the call center I don't know if you knew that, cme does it very well, you can have multiple groups etc, it will also give statistics, I have that in many customers and they're happy. However I would recommend use the phones in SCCP for that.

Hi, thank you very much, but now I tried to call the eyebeam with only codec G.711uLaw enabled and the result is the same.

Please see the attached trace.

Regarding the CallCenter, I need to use the SIP Protocol, because our CallCenter Solution uses SIP Protocol.

So I need to understand how to let it communicate in the best way with the CME.

Hi, you still have g.729 configure somewhere, look below.

The problem with third-party SIP and CME is that is not officially supported, so when something doesn;t work, you're on your own.

Instead the call center with cme only is simple and self-contained.

Feb 11 13:06:59: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:105@192.168.10.16:58260 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.57:5060;branch=z9hG4bKFB9D0

Remote-Party-ID: "Prova 3" <103>;party=calling;screen=yes;priv

acy=off

From: "Prova 3" <103>;tag=F04C64C-F1A

To: <105>

Date: Mon, 11 Feb 2008 12:06:59 GMT

Call-ID: AE827403-D7D011DC-92F6D76C-6E60F463@192.168.10.57

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 2922144784-3620737500-2465257324-1851847779

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF

Y, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1202731619

Contact: <103>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 240

v=0

o=CiscoSystemsSIP-GW-UserAgent 8720 280 IN IP4 192.168.10.57

s=SIP Call

c=IN IP4 192.168.10.57

t=0 0

m=audio 19388 RTP/AVP 18 19

c=IN IP4 192.168.10.57

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:19 CN/8000

a=ptime:20

Hello, you are right, the codec requested is still the g.729, but reading the traces I think that it is the CME which sets this codec.

Because the INVITE message is sent from "Prova 3" which is the extension 103 of the Cisco.

So is not the Eyebeam configured for g.729.

What do you think?

If you agree, where can I set the codec for the extension 103 into the CME ?

Under voice register pool XX, configure "codec g711ulaw".

Ok, I'll try to set the codec g.711aLaw

Anyway, at now I have registered the Application on the CME, and I also tried to start threee contemporary calls to the Application. ( at now the audio does not work, but I see on the application that the three calls arrives).

But I'm not sure that this is the correct way, because how many contemporary calls can I make to the application extension ?

This could be a problem.

I'm also thinking to SIP Trunking, but our application is not a registrar.

What do you think?

The task is that for example 30 calls arrive on the pstn e1 trunk connected to the Cisco, these calls must be re-directed automatically to the application which should manage them in order to connect these arriving calls to the extensions of the cisco.

So I have to receive incoming calls via ip from cisco and send back to the cisco via ip the calls in order to let cisco extensions ring.

What do you think is the correct way to implement this knowing that our application is SIP ?

Thank you very much in advance!

Hi, I think the router can do what you want, however is a bit risky due to interaction with a non-cisco device, as I said if all what you need to do is a call center, the simplest way to do that is SCCP on the phones, then the router has ephone-hunt for sequential, peer or least-idle to the phones, also you have a button on the phone to logout the agent when needed, finally as I mentioned before you have statistics to tell you average answer time, call duration and all the things associated with a call center.

Yes, ok I've understood, but I need to integrate our application with Cisco CME, so I can't use SCCP on the cisco phones.

Our application can operate as a callcenter, unified messaging, ivr, and much more and it is completely SIP compliant.

So can you suggest me the correct way to integrate it with the CME, please?

Or eventually make me in contact with a person who can help me doing this.

It is very important for us to understand the starting architecture, ho to connect the application on the CME and how to send back to the CME the calls.

This is the first step, after that we have to understand how to retrieve the status info of the extensions, I mean if they are busy, talking, registered, ecc...

Please help me!

Hi, if you want to use the router as a SIP GW for you application, that is fine and will work ok with a simple configuration.

The thing is than when you use a cisco phone in SIP with a third-party call control system, you are likely to have many issues, because the phone software is actually desinged for cm and cme.

Look at voip-info.org for suggestions on how to integrate cisco phones with a popular SIP server.

mazloumi.arash
Level 1
Level 1

Dear biesselillo

I have two Aastra phone (57i,31i), i tried everything  that i know about sip phones but i cant register that in both CUCM and  CME, if you do that i appericite your help, everytime i get 408 error on phones

Thanks

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: