SIP Trunk to H.323 Phone

[travlries]

Does anyone know if any of the releases of CM, does a call from a SIP trunk to an H.323 phone shuffle off and not use a MedPro resource?

I always thought it goes through the TDM bus because of the different call signalling but my engineer is saying it does shuffle. If so does anyone have any documentation stating either way?

 

[sjforcum]

When we looked into SIP Trunking from our carriers, the hold up seemed to be on the carrier side, not the ACM side.

An RTP stream is an RTP stream after all....

 

[travlries]

Does anyone have a packet capture of this happening? Or show an Avaya SIP call to go with this? Anyone out there using SIP trunking with H.323 phones?

 

[gblucas]

I am using SIP trunking with H.323 phones, but I have mine going through an appliance at the moment. I have the SES server installed, just haven't had time to get everything up and running to port over the trunks. We plan to have 96 SIP trunks when all is said and done. If anyone knows of any issues, I am interested also...

gblucas

 

[travlries]

Have you done any testing with the trunks? I would really love to see a packet capture of the call if you have one. H.323 to Sip trunk.

 

[gblucas]

My current setup is an Adtran unit that is doing the SIP stuff, connected PRI to a CM4.0 system. I am getting pressure to get the new SIP trunks setup to the SES server, so it could be done by the end of the month. I will see what I can do...

I do have an EMMC server as well, running SIP, and I use a H.323 phone on my desk everyday. I really haven't used EMMC since I set it up. I could do some testing on it possibly...

We are still running mostly 2 wire digital phones everywhere. We started with IP phones about a year ago to deploy for At-Home Agents. It works great!!!!

gblucas

 

[Midstream]

I've got a customer with an S8720 3.1, SES server, SIP trunks and H.323 phones. We're having issues with Meet-Me Conferencing. A call comes in the SIP trunks, hits the vector, caller gets the code in and is joined to the conference. About 7 seconds later the caller is prompted for the code again and then dropped from the call. If a call comes in the ISDN-PRI trunks to the same vector everything works well. Can't put my finger on where things are going wrong.

 

[travlries]

rockspop,

Do you think you could send me a packet capture of a call from a Avaya H.323 phone to SIP trunk?

travisriesenberg@gmail.com

 

[jimbojimbo]

Verify DTMF set for rtp-payload on IP Options form.

Set the conference bridge in a separate network region and turn off hair pinning and shuffling.

 

[travlries]

gblucas,

Could you send me a capture of a call on the SIP trunk?