Outgoing call drop @ 30 seconds

Asterisk Asterisk Support
1

Hi,

I’m trying to configure NEC Univerge SV-8100, Asterisk 1.8.7.1 and Wellgate FXO 2540 (wellgate is connected to LG Nortel PBX) with SIP protocol.

The outgoing call from Wellgate FXO 2540 to NEC Univerge SV-8100 via Asterisk is working, but having problem if outgoing from NEC Univerge SV-8100 to Wellgate FXO 2540 via Asterisk.

The scenario is
The extension is 199 in NEC Univerge SV-8100
The extension is 490 in Wellgate FXO 2540

I’m able to make a conversation from 199 to 490, but it always drop the call @ 30 seconds.

I can see from sip debug the invite, ack status and the sip xxx/xxxx is answered sip xxx/xxxx

I have tried some suggestion regarding the retransmission thing, but no luck.

appreciate for your help

2

What retransmission thing? PLease provide the actual SIP traces.

Drops at 30 seconds with speech set up properly are typically due to broken handling of re-invites. This can be due to directmedia (turn off directmedia, and its older name, canreinvite), or connected line update (disable RPID). However, in both cases, you would see an outgoing INVITE after the ACK for the initial call set up.

3

This is an asterisk bug. See https://issues.asterisk.org/jira/browse/ASTERISK-25154