[SOLVED] AMI originate call drops after 32 seconds

FreePBX Development
1

I have a strange problem using AMI to originate a call.
If I do

Action:originate
Channel:SIP/203
Exten:202
Priority:1
Context:from-internal
Account:203

my softphone (extension 203) rings and when I answer I’m connected to extension 202. I can keep the conversation as long as I want.

But if I

Action:originate
Channel:SIP/203
Exten: xxxxxxxxxx
Priority:1
Context:from-internal
Account:203

where xxxxxxxxxx is my mobile phone number then
my softphone (extension 203) rings and when I answer my mobile rings. If I answer the call the line drops exactly after 32 seconds.
The outbound route goes to a DAHDI channel (isdn line).

If I manually dial my mobile phone number in the softphone (extension 203) I can keep on talking as long as I want.
I know 32 seconds in the SIP timer B standard timeout but I can’t understand what is happening.
Any help would be greatly appreciated.

Andrea

2

I solved the problem but I don’t really understand what’s happening,
The softphone I use is counterpath x-lite. If I go to options, advanced, network and disable “in times of network disruption, automatically hang up calls after:” the communication isn’t dropped anymore.
The call drop is surely related to RTCP timeout. If I shorten the timeout to 10 seconds the call is dropped after 10 seconds.If I disable RTCP on the softphone the call is not dropped. I issued a rtp debug from asterisk’s CLI but everything seems normal. I can see rtcp packets going both ways.
The problem seems related to bridging. If I originate the call from AMI then bridging occours and connection drops il RTCP is enabled on the softphine, If I dial the number on the softphone no bridging occours ad the call does’nt drop even with RTCP enabled.
Any ideas?