Is anyone familiar with setting up an AudioCode MP112 device? I have it set up in the sense that I can call the analog phone that’s connected to it from another extension. However I am unable to use the analog phone to call out (“your call can’t be completed as dialed” message after what feels like an eternity of silence), whether it’s to another extension or an external number (in the format 91[AreaCode][Number]).
Hope I’m doing this paste bin thing right, but here’s what I saw when I tried the call with asterisk -rvvv (extension I set for the AudioCode is 7449).
Is there something obvious I am missing? I saw the template had a default dial pattern of x+#|xx+* but I removed that, thinking it would make a difference. Alas no difference was made. I had followed a tutorial on YouTube (Audiocodes MP112 Gateway Setup | VoIP Supply - YouTube) and entered my FreePBX information.
-- Executing [[email protected]:7] GotoIf("PJSIP/7449-00000088", "1?restrictedroute-825989f0a48ada175aa43c68e29d84ee,99,2:outbound-allroutes,99,2") in new stack
-- Goto (restrictedroute-825989f0a48ada175aa43c68e29d84ee,99,2)
-- Channel 'PJSIP/7449-00000088' sent to invalid extension: context,exten,priority=restrictedroute-825989f0a48ada175aa43c68e29d84ee,99,2
Did you intend to call 99? Your system seems to have 99 listed as indicating a restricted route, a number pattern that you never let people call.
Yeah not intending to call it. I tried calling 2000 (one of the extensions I have) as well as my cell phone. Right now I’m running into something else, probably related. I have to see if I have a credential mismatch somewhere along the line. Though I’m not sure why it says “<sip:[email protected]…” I don’t have an extension 1001 and within the web configuration for the AudioCode I don’t see anything for that number.
Sorry by the way, I’m still familiarizing myself with FreePBX and Asterisk. I’ve come a long way since Marc when I started looking at it more, but still some what I’d imagine basic stuff still trips me up.
[2022-05-11 13:39:08] NOTICE res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:[email protected];user=phone>' failed for '10.60.190.6:5060' (callid: [email protected]) - No matching endpoint found
[2022-05-11 13:39:08] NOTICE res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '<sip:[email protected];user=phone>' failed for '10.60.190.6:5060' (callid: [email protected]) - Failed to authenticate
I know very little about AudioCodes but have one system that uses it.
On line 114 of your .ini file, there is a definition for TrunkGroup 1 with an empty phone number. I am guessing that this defaulted to 1001.
The authentication failure may be a mismatch between the SIP password set in the device and the value of Secret for the extension in FreePBX. Possibly, there is a limitation on length or character set – try a Secret consisting of no more than 12 letters and digits.
My system has an entry under [MEGACO Params] with something like: DIGITMAPPING = '911|9911|[2-7]xxx|91[2-9]xx[2-9]xxxxxx|9[2-9]xx[2-9]xxxxxx|*x.T'
which would allow dialing 4-digit extension numbers starting with 2 thru 7, as well as 10-digit external numbers prefixed by 9 or 91.
I’m making some progress. I removed the definition for TrunkGroup 1 (I think it was partly a copy-paste error on my part from an existing .ini file for an AudioCode in our Alcatel OXE). Now it definitely seems to be pre-pending a 9 to whatever I call out. It shows I’m trying to call 92000 and understandably fails since that extension doesn’t exist.
- Executing [[email protected]:7] GotoIf("PJSIP/7449-0000001b", "1?restrictedroute-825989f0a48ada175aa43c68e29d84ee,92000,2:outbound-allroutes,92000,2") in new stack
I don’t know if it makes a difference, but I put the dial plan you had both in the .ini file under [MEGACO Params] as well as in the Endpoint Manager for the AudioCode template. I’ll dig around some more in the configuration web interface for the AudioCode and see if I can find it adding the 9.