vicidial.org

When the agent logs in the softphone rings and he is able to hear "you're the only one in this conference". The problem occurs when dialing, the call does dial but in the end it say Call Congest

sip show peers gives this;

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xbs201/xbs201              192.168.1.103    D   N      37168    Unmonitored           
switch2voip-inbound        66.33.146.52                 5060    Unmonitored           
Switch2Voip/AccNo          169.132.196.11               5060    Unmonitored           


these are the cli outputs for sip debug;

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Jun 19 05:02:39] NOTICE[12655]: chan_sip.c:8178 sip_reregister:    -- Re-registration for  AccNo@sip.switch2voip.us

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10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2493f1123cf73a23529fc98876235f76@192.168.1.101' in 32000 ms (Method: REGISTER)
[Jun 19 05:02:55] NOTICE[12655]: chan_sip.c:13779 handle_response_register: Outbound Registration: Expiry for sip.switch2voip.us is 120 sec (Scheduling reregistration in 105 s)


These are the specs of the system

Vicidial VERSION: 2.2.1-237 BUILD: 100510-2015
asterisk-1.4.39.1
dahdi-linux-complete-2.9.1.1+2.9.1
libpri-1.4.15
No extra hardware

add qualify=yes to your carriers, and post here the carriers.

Registration String :

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register =>USERNAME:PASSWORD@server/siptrunking

Account Entry :

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[Switch2Voip]
username={USERNAME}
type=peer
secret={PASSWORD}
progressinband=never
port=5060
nat=auto
insecure=very
ignoresdpversion=yes
host=server
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=g729&g711&g723
fromuser=+1XXXXXXXXXX (Where XXXXXXXXXX is the phone number you wish to display on the Caller ID.)

Dialpan:

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exten => _91.,1,Set(callerid(num)=+1XXXXXXXXXX)
exten => _91.,2,Set(callerid(ani)=Phone number)
exten => _91.,3,AGI(agi://127.0.0.1:4577/call_log)
exten => _91.,4,Dial(sip/${EXTEN:1}@Switch2Voip,55,o)
exten => _91,5,Hangup

And just out of curiosity i want to know, if there is any DNS error will that prohibit the registration?
This is because i tried many different VoIP providers and i got the same error in the CLI.

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 == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
[Jun 19 23:39:07] ERROR[6908]: utils.c:967 ast_carefulwrite: write() returned error: Connection reset by peer
[Jun 19 23:39:07] ERROR[6908]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1


And at one point it does register :

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[Jun 19 23:46:54] NOTICE[7749]: chan_sip.c:13822 handle_response_peerpoke: Peer 'Switch2Voip' is now Reachable. (258ms / 2000ms)
Really destroying SIP dialog '384fc2e1089f89021b6ab7742b08c400@192.168.1.101' Method: OPTIONS
Really destroying SIP dialog '0cff96ec6202fa9d115c9a7f186afbae@192.168.1.101' Method: OPTIONS

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from 169.132.196.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6cf62443;received=ipaddress;rport=5060
From: <sip:AccNo@sip.switch2voip.us>;tag=as12b287ee
To: <sip:AccNo@sip.switch2voip.us>
Call-ID: 1dac18d043957f4f19310ad67cf691ee@192.168.1.101
Expires: 120
CSeq: 103 REGISTER
Contact: <sip:siptrunking@192.168.1.101:5060>
Server: Net2Phone ippbx
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '1dac18d043957f4f19310ad67cf691ee@192.168.1.101' in 32000 ms (Method: REGISTER)
[Jun 19 23:46:54] NOTICE[7749]: chan_sip.c:13779 handle_response_register: Outbound Registration: Expiry for sip.switch2voip.us is 120 sec (Scheduling reregistration in 105 s)

<--- SIP read from 192.168.1.103:55857 --->

Try disabling the iptables (before you disable iptables make sure you have physical access to that the server, or you will not be able to connect again)
to disable iptables run iptables -F

Try it:

[Switch2Voip]
username={USERNAME}
type=peer
secret={PASSWORD}
progressinband=never
nat=auto
insecure=very
ignoresdpversion=yes
host=server (I guess here is the IP address of name of your Switch2Voip account)
dtmfmode=rfc2833
qualify=yes
context=trunkinbound
canreinvite=no
allow=g729&g711&g723
fromuser=+1XXXXXXXXXX (Where XXXXXXXXXX is the phone number you wish to display on the Caller ID.)

None of the above posted solutions worked.

after you added qualify=yes, can you post here the output of the command on the asterisk cli " sip show peers"

Switch to voip register string needs to look like this.

register=>xxxxxxxx:xxxx@sip.switch2voip.us/siptrunking

I solved the problem by entering the carrier settings separately on sip.conf
The carrier registers and I'm able to make calls.
Not exactly sure what the problem was but this seems like a temporary fix.

If your vicidial server did not register to the carrier, the ONLY thing that matters was the registration string.

Given the odds, you placed a slightly different string in the file than you had in your carrier settings.

Try deleting the registration string from the carrier settings. The delete everything else from the sip.conf file (be sure it's all in the Vicidial carrier, of course, just as it was removed from the sip.conf file) and activate the carrier. That should not affect registration or inbound/outbound calls.

Then try moving the registration string into the carrier settings and delete it from the sip.conf file as well.

If you want to be very careful about it, move only one setting at a time. Check after each change. You'll find it eventually.