Am new to Asterisk. We’re using Elastix. We’ve created SIP trunk to one of our branch office using PBX (Shoretel) using config below via our intranet line and both incoming and outgoing calls are ok. We’ve created another SIP trunk to another branch office who is using Cisco and used the same config, but of course, different trunk name. Outgoing calls are ok but incoming calls to Asterisk goes intermittent. It goes to “not in service” voice prompt, but after once, twice of trice redial, the call goes in. There are times when incoming call to Asterisk is ok but more instances when it goes to “not in service” before call is received thru the Asterisk.
– Executing [s@from-sip-external:8] Playback(“SIP/x.x.x.x-000cf426”, “ss-noservice”) in new stack
– <SIP/x.x.x.x-000cf426> Playing ‘ss-noservice.gsm’ (language ‘en’)
Not sure why the config below is working ok to one of our SIP trunk but not with Cisco. From forums, there are mention of context=from-pstn, context=from-trunk, context=from-internal, etc. What are the differences between these and if this line is necessary?
Need assistance on how to resolve intermittent incoming call to Asterisk.
------oOo------
SIP Trunk
Outgoing Settings
PEER Details
host=x.x.x.x
port=5060
type=peer
disallow=all
allow=ulaw&alaw
dtmfmode=rfc2833
reinvite=no
canreinvite=no
thanks,
gunbladex31
You should ask in the forum of your gui in this case in elastix forums since they manage many macros to do the stuff.
To expand on the first response, the decision to output the voice message is being made by the GUI, not by Asterisk itself, so without knowledge of the internals of the GUI, one cannot know what events might cause its production.
It might be possible to debug this in pure Asterisk terms, but you would need to provide adequate debugging information (sip debug, core debug 5 and core verbose 5). However, any subsequent advice would be in terms of direct configuration of Asterisk, not in terms of using the GUI, and it might conflict with subsequent configuration via the GUI.
It may be that you have to get a reason for the error from the GUI people, and instructions on how to enable debugging, then come back here for a diagnosis, and finally go back to the GUI people for how to implement the solution.
Thanks navaismo and david55 for your quick replies. Appreciate it.
Hi,
The usual reason for intermittent incoming calls is the Asterisk Box losing it’s registration with IP Service provider.
Do you have qualify=yes in the trunk config?
Can you check the /var/log/asterisk full files for instances of UNREACHABLE or LAGGING messages.
Can you provide a log of a failed call. I.e. is the ISP sending an Invite packet to your box.
Hope this helps.
Ian
Just re-read your post. do you have a default incoming rule. i.e any CID/DID can you see a DID in the Log.
Ian