Autopatch in ASL3 with callcentric

App_rpt-users
1

Hey all,

I am new to ASL and am running an ASL3 node and its working fine. Its a simple USB version with an AIOC device on a RPI zero 2W.

I have my callcentric endpoint setup and its registered with them fine.I can see it both thru pjsip show registrations and on the callcentric site. But when I go to make a call I get the following error. The URL is just a phone number.

Can someone please point me in the right directions…

025-01-16 11:04:32.712] WARNING[66091]: app_rpt/rpt_bridging.c:567 rpt_play_tone: Cannot start tone on DAHDI/pseudo-480768157
– Executing [19417737000@autopatch:1] NoOp(“DAHDI/pseudo-723621484”, “Dialing 19417737000 via Callcentric”) in new stack
– Executing [19417737000@autopatch:2] Dial(“DAHDI/pseudo-723621484”, “PJSIP/callcentric/19417737000,20”) in new stack
[2025-01-16 11:04:32.714] ERROR[66020]: res_pjsip.c:993 ast_sip_create_dialog_uac: Endpoint ‘callcentric’: Could not create dialog to invalid URI ‘19417737000’. Is endpoint registered and reachable?
[2025-01-16 11:04:32.714] ERROR[66020]: chan_pjsip.c:2702 request: Failed to create outgoing session to endpoint ‘callcentric’
[2025-01-16 11:04:32.715] NOTICE[66092][C-0000000c]: app_dial.c:2768 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)

2

I use CallCentric but not for autopatch. Make sure you’re following the Autopatch manual for how to set it up. The endpoint should be sip.callcentric.com and make sure that your number is actually your Callcentric number + extension, not the phone number. I believe all the Callcentric numbers are still starting with 1777. For example, your number should be 1777nnnnnnn followed by your extension number. For example, if you’ve created extension 105 for AllStarLink, then it’d be 1777nnnnnnn105.

3

Yes I followed the callcentric directions for that in the pjsip.conf file.

4

Update:

I got the autopatch working by using chatgtp to examine my configs and it suggested a single line change that seemed to fix the issue. The only problem I have now is that the transmit audio from the radio works sometimes on ptt and other times it does not work, seems intermittent. Is there a setting for that somewhere ?

The one line fix for above was:

original line:
same => n,Dial(PJSIP/callcentric/${EXTEN},20)

working line:
same => n,Dial(PJSIP/${EXTEN}@callcentric,20)