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BCM50 Asterisk digital phones not working

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Elendil326

IS-IT--Management
Jun 20, 2011
8
MX
Hi everybody

I've have been helped a lot by this forum. I'm actually in a project to implement a call center connecting Asterisk (installed on Red Hat Enterprise 5.6) with Nortel Meridian CS1000 option 11c, Nortel BCM50 and other Cisco systems.

I've traveled a long way connecting all this systems trough h323 trunks, that's why I used asterisk 1.8.4.

I can help if anyone needs help on configuration files or other troubleshooting, but right now I'm facing a big trouble. There are some digital phones (Nortel T7100 and T7316) behind the Nortel BCM50 that can't connect to Asterisk.

The IP phones under the BCM50 work fine both ways, but the digital ones(Nortel T7100 and T7316), when I call to them from an Asterisk extension it rings, but it just hangs up when picking up. Same when calling from the digital phones to Asterisk.

I've done some test and the way the IP phones can work is when I put this in the ooh323.conf file:

faststart=no
h245tunneling=yes
mediawaitforconnect=yes

My test showed that when faststart is enabled, there's no audio in the IP phones, but the communication is fine when an Asterisk extension calls to the digital phones, although the problem where the Nortel phones can't call to Asterisk still remains.

I've made some search and discover that digital phones cannot be connected to Asterisk directly unless using a citel gateway or something like that, is this the case even if the digital phones are passing trough the Nortel BCM50?

Also, the h323 debug shows messages like the following when the call hangs up

ooh323 indicate 20 on call
ooh323 indicate 3 on call
ooh323 indicate -1 on call

On the other hand, when a call is completed in an IP phone or having faststart enabled and calling to the digital phone it shows somethign diferent

ooh323 indicate 20 on call
update writeformat to ulaw
.
.
It doesn't shows 3 or -1 and it seems like both sides get an arrangement of which codec to use, unlike the hanging up call.

I would really appreciate some help in here

thanks in advance
 
you need to capture sniffer traces from BCM to see which side is dropping the call and what makes it to drop the call after call establishment. to me it seems like a codec negotiation problem. one of the sides may change the codec after call establishment.
 
Thanks for your answer


That would have been a great idea, unfortunately given the urgency of the project we had to do a work around, we put a Cisco router for the digital extensions and the others were left as they were.


If anyone has any doubt on how to connect all this equipment don't doubt asking me, I may be of help.
 
I am actually having a very similar problem to this between my BCM400 and BCM50 r6. I was not having this problem prior to upgrading my BCM50 r2 to the r6 with latest patches.

Basically, I have two H323 trunks linking the two systems. All the kinks were worked out of the systems prior to the upgrade. The BCM50 sits at a remote site with two employees and the BCM400 is at HQ. The 50 has 10 IP extensions and 2 digital extensions. The IP extensions are scattered throughout the state at unmanned sites. I also have one IP extension back at HQ for troubleshooting.

I can place and receiving calls from the IP extension no problem but the remote site users with the digital extensions report people can not hear them. It happens frequently but not always but only with the digital sets.

So I've thrown out all the codecs from the list except 911a. I also turned on voice activity detection on both sides. Haven't followed up with the issue any more since that so not sure if it's resolved. Just thought I would mention it here to add to the discussion.

A bit difficult to troubleshoot since the digital sets and BCM50 are 300 miles away. So I'm just slowly working through it all.
 
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