vicidial.org

I configured my DIDs according to the manual but it's not going to the ingroup agents

specs:
Asterisk 1.8|GOAutoDial CE 3.0|Linux (CentOS) VPS|DAHDI

CLI log:
[Mar 24 20:36:06] -- Executing [18663237997@default:1] AGI("SIP/8001-0000000e", "agi-DID_route.agi") in new stack
[Mar 24 20:36:06] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Mar 24 20:36:06] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20150324203606_18663237997_0000000000)
[Mar 24 20:36:06] -- <SIP/8001-0000000e>AGI Script agi-DID_route.agi completed, returning 0
[Mar 24 20:36:06] -- Executing [99909*3***DID@default:1] Answer("SIP/8001-0000000e", "") in new stack
[Mar 24 20:36:06] -- Executing [99909*3***DID@default:2] AGI("SIP/8001-0000000e", "agi-VDAD_ALL_inbound.agi") in new stack
[Mar 24 20:36:06] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Mar 24 20:36:07] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 24 20:36:07] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 24 20:36:07] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 24 20:36:07] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 24 20:36:08] NOTICE[11599]: res_rtp_asterisk.c:2361 ast_rtp_read: Unknown RTP codec 126 received from '174.94.115.125:61562'
[Mar 24 20:36:12] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 24 20:36:12] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 24 20:36:12] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 24 20:36:12] -- Playing 'generic_hold' (escape_digits=) (sample_offset 0)
[Mar 24 20:36:18] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 24 20:36:18] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 24 20:36:18] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Mar 24 20:36:18] -- Playing 'queue-holdtime' (escape_digits=) (sample_offset 0)

Unknown RTP codec 126 received from '174.94.115.125:61562
Check with the carrier to find out what codec they are using or try sip debugging to see if you can get more details on the codec in question

Also, you should post the account entry for the carrier in question ... be sure "disallow=all" and "allow=ulaw" are the only two allow entries.

the carrier settings are disallow all and allow ulaw and alaw, should I enable only one?

[AlcazarNetIDID1]
context=inbound
type=peer
host=199.96.248.160
insecure=port
disallow=all
allow=ulaw
nat=no
qualify=yes
dtmfmode=rfc2833
canreinvite=no
sendrpid=yes

exten => _18663237997,1,AGI(agi-DID_route.agi)
exten => _18663237997,n,Dial(SIP/AlcazarNetIDID1,60)
exten => _18663237997,n,Hangup

they say everything works fine on their side

PerleG wrote:the carrier settings are disallow all and allow ulaw and alaw, should I enable only one?

if you're not in australia, skip alaw.

don't forget the sip debugging.

To: <sip:8001@192.168.2.16:14186;rinstance=b76f6d226958ae57>;tag=6ad3ab7d
From: "asterisk"<sip:asterisk@209.126.73.231>;tag=as4d30ea4d
Call-ID: 5450c91b60f77a0262676e892f0872c5@209.126.73.231:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces, eventlist
User-Agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0

<------------->
[Mar 24 21:27:36] --- (13 headers 0 lines) ---
[Mar 24 21:27:36] Really destroying SIP dialog '5450c91b60f77a0262676e892f0872c5@209.126.73.231:5060' Method: OPTIONS
[Mar 24 21:27:37]

<------------>
[Mar 24 21:27:37] Scheduling destruction of SIP dialog 'f6bf7c-2ac80ba094fe4f4-11a0cfc4@209.126.73.231' in 32000 ms (Method: REGISTER)
[Mar 24 21:27:37] Reliably Transmitting (NAT) to 174.94.115.125:25676:
OPTIONS sip:8002@174.94.115.125:25676;rinstance=ac3f5135aa038dce SIP/2.0
Via: SIP/2.0/UDP 209.126.73.231:5060;branch=z9hG4bK01df1da6;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@209.126.73.231>;tag=as67bbbded
To: <sip:8002@174.94.115.125:25676;rinstance=ac3f5135aa038dce>
Contact: <sip:asterisk@209.126.73.231:5060>
Call-ID: 42fd034e7e8c7d6202268943239be017@209.126.73.231:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Date: Wed, 25 Mar 2015 01:27:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

what do these sip debug messages mean?

I'm still having problems

PerleG wrote:I'm still having problems

I bet you are. Your SIP debug posts are not related to a phone call. Post the output from SIP debug resulting from an inbound phone call.

Also, post your DID and Ingroup settings relevant to the issue at hand: Specifically the inbound call route and Ingroup ID in the DID configuration. If you forget to specify an Ingroup (After choosing Ingroup as the route), the call will terminate because it has nowhere to go.