SIP trunking lost the routing IN_GROUP

[macaruchi]

Hi!
I have a SIP trunking and I got a weird problem.
I configured a SIP trunking and this is the CARRIER

Code: Select all

[altice]
host=x.x.x.x
username=USER
secret=PASS
fromuser=8556667777
insecure=port,invite
disallow=all
allow=ulaw
allow=alaw
type=friend
canredirect=no
canreinvite=no
progressinband=never
dtmfmode=rfc2833
qualify=no
context=trunkinbound
trunk=yes

--DialPLan
exten => _9.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9.,2,Dial(${ALTICETRUNK}/${EXTEN:1},${CAMPDTO},To)
exten => _9.,3,Hangup


I have a DID to address to IN_GROUP the DID is 8556667777. The call in to asterisk but never arrives to IN_GROUP.
This is the output in console

Code: Select all

[Jul 16 11:30:15] VERBOSE[14665][C-000010d0] pbx.c: Executing [3131480258556667777@trunkinbound:1] AGI("SIP/altice-000003a2", "agi-DID_route.agi") in new stack
[Jul 16 11:30:15] VERBOSE[14665][C-000010d0] res_agi.c: Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jul 16 11:30:15] VERBOSE[14665][C-000010d0] res_agi.c: AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20240716113015_3131480258092950000_8098498087)
[Jul 16 11:30:15] VERBOSE[14665][C-000010d0] res_agi.c: <SIP/altice-000003a2>AGI Script agi-DID_route.agi completed, returning 0
[Jul 16 11:30:15] VERBOSE[14665][C-000010d0] pbx.c: Executing [99909*5***DID@default:1] Answer("SIP/altice-000003a2", "") in new stack
[Jul 16 11:30:16] VERBOSE[14665][C-000010d0] pbx.c: Executing [99909*5***DID@default:2] AGI("SIP/altice-000003a2", "agi-VDAD_ALL_inbound.agi") in new stack
[Jul 16 11:30:16] VERBOSE[14665][C-000010d0] res_agi.c: Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Jul 16 11:30:17] VERBOSE[14665][C-000010d0] res_agi.c: <SIP/altice-000003a2> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Jul 16 11:30:17] VERBOSE[14665][C-000010d0] res_agi.c: <SIP/altice-000003a2> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')


I dont know why asterisk plays the sip-silence but in the phone everythink is quit takes a few second and hangup after 5 o 6 seconds.
It seems how the call doesnt find the route to IN_GROUP. I tested the trunk in other asterisk server and works fine

I chnage the IP because I had other SIP trunk from other provider but how the calls in to asterisk I supose that was correct
Any cluees or any ideas ?


Thks In Advance

[williamconley]

You DID post the vicidial version and build, but not together (odd, lol). But you did not post your full installer version (Vicibox 11.X.X?). Otherwise great job.

Here's the pathway for an inbound call to an Ingroup:

1) Convince the Carrier to send the call to your Vicidial server. This has two basic options: You provide the carrier with your IP and port OR you Register to a SIP account which has that DID associated in the Carrier's system.
You appear to have managed this well. Moving on!

2) Configure that Carrier's account (or sip.conf's general context account if no carrier configured for this inbound IP) to send the call to "trunkinbound" via "context=trunkinbound".
You appear to have also managed this well. Moving on!

3) Once the call is in Vicidial's inbound DID control: Configure the DID in "Inbound->Show DIDs" as its own specific DID matching the DID your carrier sends (perhaps you need to strip off extraneous digits such as "+" or "+1") OR configure the Default DID.

I do not see step three's information in your post.

Do you have the Vicidial Manager's Manual? And have you followed all steps in it without skipping any (from page one!)?

[macaruchi]

Hi!
I am here to post the solution if anybody can have this problem.
The problem was that Vicidial was putting in SIP.conf the parameter "externalip=x.x.x.x" the old IP that had this trunk.
We had changed the SIP trunk so for any reason the externalip doesnt change to new IP and vicidial kept the old IP so never heard the voice.
Bottom line, if you chnage your SIP trunk,private, check the externalip parameter in SIP.conf, we comment this and everything goes fine.

Just for anybody that has problem

TIA

[williamconley]

Excellent postback!

Yes, when a server's IP changes there are many modifications that must be made. There's also a script for the IP update, but it does not hit ... Everything: Especially the "externip" value.