Re: SIP Trunk Problems

MartyCullen · ‎04-23-2007

I have CCM 4.1 and connected a SIP trunk from an Asterisk server. Inbound and outbound calls work great... until some one wants to transfer or conference.

If a CCM client tries to transfer the call that came in over the SIP trunk, the CCM Client loses the sip client, and the SIP client just gets silence. Could I be missing something in the "Route Pattern Configuration?"

I would be more than happy to post any config information you might think is helpful.

thanks

MarkMonteleone · ‎04-23-2007

Please reply with screen shot of SIP trunk setup and explanation of Asterisk server network layout. I can see if I can help.

I am interested in seeing how you successfully setup the two callagents in the SIP config. So any config setting from Asterisk would help.

Concerning your calling plan as it pertains to the devices associated with that trunk..

What are you using for Route Patterns?

MartyCullen · ‎04-24-2007

Summary:

My HQ is here in Raleigh, NC

It has the Cisco Call manger

It has phones with extensions 40XX 80XX 81XX

My Remote office is in Moultrie, GA

It has the TrixBox / Asterisk PBX

It has phones with extensions 66XX

thanks for looking at this.

call manger

MarkMonteleone · ‎04-24-2007

Strange as it looks to be right config and your phone to phone calls work fine. I would suggest making sure that the SIP trunk is Ulaw and make the Asterisk Ulaw as well, not optional of either or.

The only other thing is to make sure that the phones that are transfering are using 711 codec as well as the phones the call is going to, unless you have a DSP farm.

MartyCullen · ‎04-24-2007

Thanks Mark,

The CCM trunk is set to 711ulaw

The Asterisk trunk was either and I have changed it to ulaw only.

Problem persists.

We are currently using all Cisco 7960G phones.

Skinny in the CCM environement and SIP in Asterisk. No problems transferring on the Asterisk side.

In SIP mode I can see that the phones are configured with NO preferred codec (Settings-4-8). Where is that setting in SCCP?

MarkMonteleone · ‎04-24-2007

Try setting one of the phones on Asterisk side to 711. See if you can transfer to that phone.

Also, as you know SCCP is Cisco only protocol. I doubt Asterisk supports any SCCP functionality.

Thirdly, do you have a DSPfarm or Codec transformation in place for the SIP trunk calls that the transfer may not be using during transfer?

MartyCullen · ‎04-24-2007

I am not currently having any transfer problems on the asterisk side. They receive trunk calls from our CCM and transfer them with no problems.

There are people doing SCCP in Asterisk.

It is the only way to use the sidecars (7914) currently.

It was not stable enough for me to use it though, and requires alot of manual configuration (bypassing TrixBox).

I have not heard of a DSPfarm or Codec transformation. I am pretty sure the only config we had to modify was the Trunks and Routes on the 2 boxes.

thanks again

kulagowskir · ‎04-24-2007

Not exactly the same situation in my instance, but similar issues.

Remote site 2851 (12.4.9T) using SIP trunk back to CM 4.1.3sr2. When a call is placed on hold, transferred, etc, it no longer can be accessed. If it's on hold, it can't be resumed. If it's transferred, the destination phone rings, but if you pickup, there's no audio path. The caller hears silence the entire time until the call is dropped.

I've got transcoders, MTP resources, conference bridges all configured and available.

I have an open TAC case, so if they come back with anything I'll post to this thread. Might help.

MartyCullen · ‎04-24-2007

Robert,

Thank you very much.

Marty

MartyCullen · ‎05-08-2007

Have you gotten anything from TAC?

I have since found that I am also getting the following errors:

Subject: [RTMT-ALERT] MediaListExhausted

At 15:15:40 on 05/07/2007 on cluster BELL-Cluster.

Number of MediaListExhausted events exceed 0 within 60 minutes.

Our "support" company is pointing the finger at Asterisk. They borrowed resources from our secondary CCM, but the problem persists. They recommend buying a seperate server to handle this (and conf calls, etc). But that aint gonna happen.

thanks

kulagowskir · ‎05-08-2007

No; I've been asking for a few weeks now for an update:

*** Service Request LOG 2007-04-23 16:20:26.0 GMT, RREICHAR, Action Type: Phone Log ***

On the phone with Robert.

*** Service Request LOG 2007-04-25 18:42:19.0 GMT, KULAGOWSKIR, Action Type: Web Update ***

Can I have an update please?

*** Service Request LOG 2007-05-01 13:46:44.0 GMT, KULAGOWSKIR, Action Type: Web Update ***

Any additional information? This would be a big feature "win" for

our company if we can get this operational.

*** Service Request LOG 2007-05-03 22:54:02.0 GMT, RREICHAR, Action Type: Email Out ***

I'm on Holiday

In an effort to provide quality support on your Service Request (SR), I...

I've called his supervisor and I'm waiting for an update on requeuing, or something.

MartyCullen · ‎05-10-2007

I just had a meeting with my Avvid support group. We have been experiencing errors on our call manager since we installed Asterisk. Erorrs like this:

Subject: [RTMT-ALERT] MediaListExhausted

At 15:15:40 on 05/07/2007 on cluster BELL-Cluster.

Number of MediaListExhausted events exceed 0 within 60 minutes.

They explained the problem as SIP not being native to CCM 4.1. So the server has to convert it and that uses "transcoding" resources, which causes these errors.

They laid out the options as:

1) Upgrade to CCM 5.1 (in which SIP is native)

2) Add an MTP Gateway

a) Software running on a server

b) firmware on a router

2b) is not an option because IOS does not support SIP

So, for us, since we have smartnet (for the software) and are allowed one free version upgrade per year (for the labor), we plan to do the upgrade. They can not guarantee that Asterisk would work, but it has to be better than what I have now. Calls dropping into the ether...

Best of luck.

antonio.fragoso · ‎02-27-2012

Hi

kulagowskir

Did you solve this problem ?

Regards

connect101 · ‎05-12-2007

Marty,

We have the same installation on our network.

Transfers on both sides works well.

First, I think that Trixbox is not appropriate for these advanced features use. You cannot master configurations.

Take the time to learn and install a full Asterisk version.

Otherwise, I think, it's a matter of RTP codecs.

When CCM client tries to transfer he sends a G729br8 codec to Asterisk. But Asterisk doesn't support G729br8. So, you need to configure all codecs to G711 on both sides.

And think to buy a G729 licence for your Asterisk.