Outgoin calls issue Plz help

[shanjay86]

Hey i m not able to dial wid my dialer works wid other sip account but not with this one for some reason, n my VOIP people are little slow n dont understand my problem sayin we dont know wht you are asking...

My dial plan

Code: Select all

[Net4]
type=friend
username=Username
fromuser=Username
secret=Pass
host=Domain
fromdomain=Domain
dtmfmode=rfc2833
allow=g729
allow=ulaw
disallow=all
canreinvite=no

exten => _76X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _76X.,2,Dial(SIP/${EXTEN:2}@kdots,,tTo)
exten => _76.,3,Hangup



Putty Report

Code: Select all

Connected to Asterisk 1.4.27.1-1 RPM by demian@goautodial.com currently running on go (pid = 2052)
Verbosity is at least 21
[Apr  2 11:48:57] NOTICE[30196]: chan_sip.c:3115 auto_congest: Auto-congesting SIP/kdots-0000002f
[Apr  2 11:48:57]     -- SIP/kdots-0000002f is circuit-busy
[Apr  2 11:48:57]   == Everyone is busy/congested at this time (1:0/1/0)
[Apr  2 11:48:57]     -- Executing [7614045797903@default:3] Hangup("Local/8600052@default-5d4f,1", "") in new stack
[Apr  2 11:48:57]   == Spawn extension (default, 7614045797903, 3) exited non-zero on 'Local/8600052@default-5d4f,1'
[Apr  2 11:48:57]     -- Executing [h@default:1] DeadAGI("Local/8600052@default-5d4f,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------") in new stack
[Apr  2 11:48:57]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION---------- completed, returning 0
[Apr  2 11:48:57]   == Spawn extension (default, 8600052, 1) exited non-zero on 'Local/8600052@default-5d4f,2'
[Apr  2 11:48:57]     -- Executing [h@default:1] DeadAGI("Local/8600052@default-5d4f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr  2 11:48:57]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr  2 11:48:58]   == Refreshing DNS lookups.
[Apr  2 11:49:01]   == Parsing '/etc/asterisk/manager.conf': [Apr  2 11:49:01] Found
[Apr  2 11:49:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  2 11:49:01]   == Parsing '/etc/asterisk/manager.conf': [Apr  2 11:49:01] Found
[Apr  2 11:49:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  2 11:49:02] ERROR[5638]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer
[Apr  2 11:49:02] ERROR[5638]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Apr  2 11:49:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  2 11:49:07]   == Parsing '/etc/asterisk/manager.conf': [Apr  2 11:49:07] Found
[Apr  2 11:49:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  2 11:49:07] ERROR[5670]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer
[Apr  2 11:49:07] ERROR[5670]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Apr  2 11:49:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  2 11:49:07] ERROR[5640]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Apr  2 11:49:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  2 11:49:07]     -- Unregistered SIP 'cc100'
[Apr  2 11:49:15]   == Parsing '/etc/asterisk/manager.conf': [Apr  2 11:49:15] Found
[Apr  2 11:49:15]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  2 11:49:17]   == Parsing '/etc/asterisk/manager.conf': [Apr  2 11:49:17] Found
[Apr  2 11:49:17]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  2 11:49:17]   == Spawn extension (default, 58600052, 1) exited non-zero on 'Local/58600052@default-9588,2'
[Apr  2 11:49:17]     -- Executing [h@default:1] DeadAGI("Local/58600052@default-9588,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr  2 11:49:17]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr  2 11:49:17]   == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600052@default-9588,1'
[Apr  2 11:49:17]     -- Executing [h@default:1] DeadAGI("Local/58600052@default-9588,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr  2 11:49:17]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr  2 11:49:19]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  2 11:49:21]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  2 11:49:24]     -- Registered SIP 'cc100' at 122.164.115.209 port 10032
[Apr  2 11:49:24]     -- Saved useragent "eyeBeam AudioOnly release 3014w stamp 26704" for peer cc100
[Apr  2 11:49:24]   == Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/cc110-0000002c'
[Apr  2 11:49:24]     -- Executing [h@default:1] DeadAGI("SIP/cc110-0000002c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr  2 11:49:24]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Apr  2 11:49:24] NOTICE[30196]: chan_sip.c:13408 handle_response_peerpoke: Peer 'cc100' is now Reachable. (341ms / 2000ms)
[Apr  2 11:49:24] WARNING[30196]: chan_sip.c:13662 handle_response: Remote host can't match request NOTIFY to call '40056e036c7f472a@c3lzMi5xcHJpbXVzLmNvbQ..'. Giving up.
[Apr  2 11:49:25]     -- Unregistered SIP 'cc100'
[Apr  2 11:49:26]     -- Registered SIP 'cc100' at 122.164.115.209 port 10032
[Apr  2 11:49:26]     -- Saved useragent "eyeBeam AudioOnly release 3014w stamp 26704" for peer cc100
[Apr  2 11:49:27] NOTICE[30196]: chan_sip.c:13408 handle_response_peerpoke: Peer 'cc100' is now Reachable. (356ms / 2000ms)


I dont get calls at all if i use a different SIP it works fine
The Voip/SIP people gave me the dial plan
n ext 761. to use used
i changed 761 it in campaign too

[williamconley]

exten => _76X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _76X.,2,Dial(SIP/${EXTEN:2}@kdots,,tTo)
exten => _76.,3,Hangup

76X. and 76. are not the same extension. Therefore you have two different dial plans that are not related.

Ordinarily this would ignore the "exten => _76.,3," entry as it has no ",1," priority level, but it appears your system is magically ONLY executing the ",3," entry. Never bumpted into that before, but then again, this is open Source.

Also, you appear to have "auto-congesting" issues, which means your server is unable to reach the remote network with a sip packet.

Perhaps you should use the Vicidial Manager's Manual to construct a Standard sip account entry and/or verify that your server can, indeed, reach the remote server via ping and/or sip traffic.

[striker]

hi

[Net4]
type=friend
username=Username
fromuser=Username
secret=Pass
host=Domain
fromdomain=Domain
dtmfmode=rfc2833
allow=g729
allow=ulaw
disallow=all
canreinvite=no

exten => _76X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _76X.,2,Dial(SIP/${EXTEN:2}@kdots,,tTo)
exten => _76.,3,Hangup

as you have given the context as Net4 and dialling @kdots.
make the below changes in your carrier setting of vicidial

Globals String : NET4TRUNK=SIP/Net4

exten => _761X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _761X.,2,Dial(${NET4TRUNK}/${EXTEN},,tTor)
exten => _761X.,3,Hangup[/quote]

Note: make sure your sip trunk is registered properly
in asterisk cli type sip show peers and sip show registry , to check the status of the sip trunk.
also you have mentioned g729 codec, did you installed the g729 codec in your vicidal server.