[Help]: Bridge Event Missing

Asterisk Asterisk Support
1

Hi there,

When a call comes into my PBX through my SIP line and the call is answered I recieve three bridge events on the AMI:

Event: Bridge
Privilege: call,all
Bridgestate: Link
Bridgetype: core
Channel1: Local/SIP-3901@MemberConnectorQ-00000625;2
Channel2: SIP/3901-00001294
Uniqueid1: 1391716255.8219
Uniqueid2: 1391716255.8220
CallerID1: 077xxxxxxxx
CallerID2: 014xxxxxxxx

Event: Bridge
Privilege: call,all
Bridgestate: Link
Bridgetype: core
Channel1: SIP/443xxxxxxxx-00001293
Channel2: Local/SIP-3901@MemberConnectorQ-00000625;1
Uniqueid1: 1391716232.8217
Uniqueid2: 1391716255.8218
CallerID1: 077xxxxxxxx
CallerID2: 014xxxxxxxx

Event: Bridge
Privilege: call,all
Bridgestate: Link
Bridgetype: core
Channel1: SIP/443xxxxxxxx-00001293
Channel2: SIP/3901-00001294
Uniqueid1: 1391716232.8217
Uniqueid2: 1391716255.8218
CallerID1: 077xxxxxxxx
CallerID2: 014xxxxxxxx

Which I am quite happy with, I ignore the bridges with the ‘Local/’ channels and process the other bridge which is suitable for my needs.

However when a call comes into my PBX via my PSTN I only recieve 2 bridge events (The ‘Local/’ channels’) The third Bridge event seems to be missing… Or is it supposed to do this?

Event: Bridge
Privilege: call,all
Bridgestate: Link
Bridgetype: core
Channel1: Local/SIP-3901@MemberConnectorQ-00000624;2
Channel2: SIP/3901-00001290
Uniqueid1: 1391716021.8212
Uniqueid2: 1391716021.8213
CallerID1: 77xxxxxxxx
CallerID2: 014xxxxxxxx

Event: Bridge
Privilege: call,all
Bridgestate: Link
Bridgetype: core
Channel1: DAHDI/i2/77xxxxxxxx-108
Channel2: Local/SIP-3901@MemberConnectorQ-00000624;1
Uniqueid1: 1391715999.8210
Uniqueid2: 1391716021.8211
CallerID1: 77xxxxxxxx
CallerID2: 014xxxxxxxx

Is there a reason for this?

Thanks for any help.

Chad

2

Do you by any chance have the “n” option on the Local channel address in the Dial.

Note that you only get the multiple links because you, or the writer of your dial plan, has made a positive decision to use Local channels.

The lack of the last link for PSTN means that the Local channel is not being optimised out.

3

Thanks for the reply David,

The local channels were being used as a means to check a DND status I set in the astDB for each sip device. So when a call was in a queue I could check this value to see if the were to recieve the call or not.

There was no ‘n’ option present as you had asked.

But your reply has made me rethink using the locals channels and just use the dynamic ability to login and out of queues. In turn allowing the agents to pause and unpause their status in the queue.

A bit more work but less of a headache :smile:

Thanks again

Chad