"Currently you are the only person in the call"

[Faizankhan1995]

I am new to ViciDial so sorry if there is something missing in question. I have installed viciDial using viciBox.iso (ViciBox_v8_1.x86_64-8.1.2.iso) on local machine. I was able to login as admin. I have added campaigns, users , phones (using EyeBeam which is registered successfully) and carriers. When the agent logs in and answers the call the machine says "Currently you are the only person in the call" i have tried manual dial but is displays an error saying
Call Rejected: CONGESTION
Cause: 34 - No circuit/channel available.
SIP: 503 - Service Unavailable)
I have white listed my public IP in VoIP provider portal.
I am not sure what the problem is and how to debug it. Can someone guide me about how to identify the possible problems and how to fix it. Any help is appreciated.

I am using default configurations of ViciBox so all the versions would be default.

Regards,
Faizan

[bourneshell]

Hi Faizan,

Can you provide the following:
1. Carrier config
2. Dial Plan
3. Is the server behind a router / firewall?

[williamconley]

I am new to ViciDial so sorry if there is something missing in question. I have installed viciDial using viciBox.iso (ViciBox_v8_1.x86_64-8.1.2.iso) on local machine. I was able to login as admin. I have added campaigns, users , phones (using EyeBeam which is registered successfully) and carriers. When the agent logs in and answers the call the machine says "Currently you are the only person in the call" i have tried manual dial but is displays an error saying
Call Rejected: CONGESTION
Cause: 34 - No circuit/channel available.
SIP: 503 - Service Unavailable)
I have white listed my public IP in VoIP provider portal.
I am not sure what the problem is and how to debug it. Can someone guide me about how to identify the possible problems and how to fix it. Any help is appreciated.

I am using default configurations of ViciBox so all the versions would be default.

Regards,
Faizan

1) Excellent job posting your Installer Version (ViciBox_v8_1.x86_64-8.1.2.iso)

2) Please remember to Always post your Vicidial Version With Build (available on the bottom left corner of almost every admin web page). Note that any version of the installer CAN install any version of Vicidial, so the installer doesn't tell us which version of Vicidial you're running.

3) If you have not already done so, now is a good time to download the Vicidial Manager's Manual from EFLO.net and start at page one. Do not skip anything (even things you have done already). Page one until "it all works" without any skips. When you get to the stage where you are installing the Carrier, there are hundreds of posts on this site with examples and troubleshooting for configuring your carrier. If you have any problem, post the Vicidial Version, Manager's Manual Version, plus the page and line where you hit your snag. Post what you expected to happen and what really happened. In the case of a failed call, also post Asterisk CLI output from a single attempt. ONE call, beginning to end, with no other traffic (and not 3000 lines of unrelated call, either! just one call attempt beginning to end). You should also post any relevent configuration settings (in this case: the carrier settings! X out the user/pass/ip of the carrier, of course).

Asterisk CLI output is available by typing:

Code: Select all

asterisk -R

at the linux command line.

This site has gotten thousands of new users up and running, and The Vicidial Group created the FREE version of the Vicidial Manager's Manual to avoid posting 150+ pages of content to cover all possible problems and outcomes for each new user who makes the attempt.

Happy Hunting!

[Faizankhan1995]

VERSION: 2.14-711a
BUILD: 190607-1525
© 2019 ViciDial Group

1- Dail Plan Entry :

Code: Select all

[quote]exten => _84452.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _84452.,n,Dial(SIP/globex/${EXTEN:5},,tToR)
exten => _84452.,n,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})[/quote]

2- Carrier Config :

Code: Select all

[globex]
disallow=all
allow=ulaw
dtmf=rfc2833
type=friend
qualify=no
canreinvite=no
nat=comedia
host=192.69.217.137

3- No the server is not behind any firewall.

4- Below are the Asterisk CLi Logs :

Code: Select all

vicibox-local:~ # asterisk -R
[Jun 27 17:34:33] Connected to Asterisk 13.24.1-vici currently running on vicibox-local (pid = 1478)
[Jun 27 17:35:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 27 17:35:33]     -- Registered SIP '8010' at 192.168.10.24:8520
[Jun 27 17:35:33]        > Saved useragent "eyeBeam release 3007n stamp 17816" for peer 8010
[Jun 27 17:35:33] NOTICE[1546]: chan_sip.c:24659 handle_response_peerpoke: Peer '8010' is now Reachable. (5ms / 2000ms)
[Jun 27 17:36:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 27 17:38:50]   == Using SIP RTP CoS mark 5
[Jun 27 17:38:50]     -- Called 8010
[Jun 27 17:38:51]        > 0x7f150400bae0 -- Strict RTP learning after remote address set to: 192.168.10.24:6074
[Jun 27 17:38:51]     -- SIP/8010-00000000 answered
[Jun 27 17:38:51]     -- Executing [8600051@default:1] MeetMe("SIP/8010-00000000", "8600051,F") in new stack
[Jun 27 17:38:51]     -- Created MeetMe conference 1023 for conference '8600051'
[Jun 27 17:38:51]     -- <SIP/8010-00000000> Playing 'conf-onlyperson.gsm' (language 'en')
[Jun 27 17:38:52]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 27 17:38:52]        > 0x7f150400bae0 -- Strict RTP switching to RTP target address 192.168.10.24:6074 as source
[Jun 27 17:38:56]        > 0x7f150400bae0 -- Strict RTP learning complete - Locking on source address 192.168.10.24:6074
[Jun 27 17:39:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 27 17:39:05]     -- Hungup 'DAHDI/pseudo-370170875'
[Jun 27 17:39:05]   == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/8010-00000000'
[Jun 27 17:39:05]     -- Executing [h@default:1] AGI("SIP/8010-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)") in new stack
[Jun 27 17:39:05]     -- <SIP/8010-00000000>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK) completed, returning 0
[Jun 27 17:39:14]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 27 17:39:14]     -- Called 8445219702164322@default
[Jun 27 17:39:14]     -- Executing [8445219702164322@default:1] AGI("Local/8445219702164322@default-00000000;2", "agi://127.0.0.1:4577/call_log") in new stack
[Jun 27 17:39:14]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=FE))
[Jun 27 17:39:14]     -- <Local/8445219702164322@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jun 27 17:39:14]     -- Executing [8445219702164322@default:2] Dial("Local/8445219702164322@default-00000000;2", "SIP/globex/19702164322,,tToR") in new stack
[Jun 27 17:39:14]   == Using SIP RTP CoS mark 5
[Jun 27 17:39:14]     -- Called SIP/globex/19702164322
[Jun 27 17:39:14]     -- Local/8445219702164322@default-00000000;1 is ringing
[Jun 27 17:39:14]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 27 17:39:14]     -- Called 8445219158580200@default
[Jun 27 17:39:14]     -- Executing [8445219158580200@default:1] AGI("Local/8445219158580200@default-00000001;2", "agi://127.0.0.1:4577/call_log") in new stack
[Jun 27 17:39:14]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=FE))
[Jun 27 17:39:14]     -- <Local/8445219158580200@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jun 27 17:39:14]     -- Executing [8445219158580200@default:2] Dial("Local/8445219158580200@default-00000001;2", "SIP/globex/19158580200,,tToR") in new stack
[Jun 27 17:39:14]   == Using SIP RTP CoS mark 5
[Jun 27 17:39:14]     -- Called SIP/globex/19158580200
[Jun 27 17:39:14]     -- Local/8445219158580200@default-00000001;1 is ringing
[Jun 27 17:39:14]     -- Got SIP response 503 "Service Unavailable" back from 192.69.217.137:5060
[Jun 27 17:39:14]     -- SIP/globex-00000001 is circuit-busy
[Jun 27 17:39:14]   == Everyone is busy/congested at this time (1:0/1/0)
[Jun 27 17:39:14]     -- Executing [8445219702164322@default:3] DeadAGI("Local/8445219702164322@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION----------") in new stack
[Jun 27 17:39:14] WARNING[3516][C-00000001]: res_agi.c:4619 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases!
[Jun 27 17:39:14]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=FE))
[Jun 27 17:39:14]     -- <Local/8445219702164322@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION---------- completed, returning 0
[Jun 27 17:39:14]     -- Auto fallthrough, channel 'Local/8445219702164322@default-00000000;2' status is 'CONGESTION'
[Jun 27 17:39:14]     -- Local/8445219702164322@default-00000000;1 is circuit-busy
[Jun 27 17:39:14]     -- Executing [h@default:1] AGI("Local/8445219702164322@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION---------------SIP 503 Service Unavailable)") in new stack
[Jun 27 17:39:14]     -- Got SIP response 503 "Service Unavailable" back from 192.69.217.137:5060
[Jun 27 17:39:14]     -- SIP/globex-00000002 is circuit-busy
[Jun 27 17:39:14]   == Everyone is busy/congested at this time (1:0/1/0)
[Jun 27 17:39:14]     -- Executing [8445219158580200@default:3] DeadAGI("Local/8445219158580200@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION----------") in new stack
[Jun 27 17:39:14] WARNING[3521][C-00000002]: res_agi.c:4619 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases!
[Jun 27 17:39:14]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=FE))
[Jun 27 17:39:14]     -- <Local/8445219158580200@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION---------- completed, returning 0
[Jun 27 17:39:14]     -- Auto fallthrough, channel 'Local/8445219158580200@default-00000001;2' status is 'CONGESTION'
[Jun 27 17:39:14]     -- Local/8445219158580200@default-00000001;1 is circuit-busy
[Jun 27 17:39:14]     -- Executing [h@default:1] AGI("Local/8445219158580200@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION---------------SIP 503 Service Unavailable)") in new stack

Thanks to community for support.
Regards,
Faizan

[williamconley]

[Jun 27 17:39:14] -- Got SIP response 503 "Service Unavailable" back from 192.69.217.137:5060
[Jun 27 17:39:14] -- SIP/globex-00000002 is circuit-busy
[Jun 27 17:39:14] == Everyone is busy/congested at this time (1:0/1/0)

Service Unavailable = you can't dial this number through our service. Now you have to find out what the problem is: Is it ... you? or the number you dialed? you'll have to ask the carrier since the error doesn't contain a reason.

[Faizankhan1995]

I am working on it and now I am able to make calls but there is no voice. Any idea about it ?

Regards,
Faizan

[williamconley]

Firewall. you know: that thing you say "you're not behind"? Yep. That's the problem. lol

Well, without the smartass remarks, there are a couple possibilities:
1) /etc/asterisk/sip.conf must have externip=xx.xx.xx.xx that matches your server's public IP.
2) If you are on a private network (such as 192.168.x.x or 10.x.x.x), you'll need to change the sip account's "nat" entry to something other than "no". If your system is old, "yes" will do it. But if it's fairly new, you may need nat=force_rport,comedia since nat=new is deprecated
3) In all cases you'll need to be sure that the IP of the sip server to whom you are communicating is allowed both through your ROUTER (which is another word for "firewall", for all practical purposes) and allowed to access to your servers firewall (if iptables is on).
4) Additionally, the audio server's IP address must also be allowed IF your carrier uses a different server for audio than it does for signalling. Signalling is usually port 5060, but audio actually happens on a random port from 10000 to 25000. So your router or firewall may drop packets from the audio server since it has no idea why that random IP is sending packets to a random port. Your server will request that random port be opened, and directed back to your server ... but an unknown IP sending packets to that random port may cause them to be discarded instead of being routed to your server as a security measure.
5) Worst case scenario: If your server is behind a firewall (NAT) and the Carrier's server is also behind a firewall (Now we are in "double-NAT" territory) challenges arise that are a bit more in-depth. Some routers are coded to catch and enable this. Others are coded to screw it up! In which case turning off whatever "feature" screws this up may be necessary. Luckily there are NO rules for this so you (the lucky end user) may get to figure it out on your own.

Most importantly: This is NOT a Vicidial feature/function/concept. This is purely asterisk / "sip server" communications. The asterisk community is huge. Vicidial is fairly small. So if you don't get this quickly with this help, reach out with google and see if you can find someone else with your carrier and your "situation" (by situation i refer to "NAT" vs "No NAT"). If NAT is still a foreign concept to you: Network Address Translation is the act of converting packets from a "Public" to a "Private" network, which is what happens when there is a router, which is on the Public IP, transmitting packets to servers/computers on the Private/Internal network (192.168.x.x or 10.x.x.x in most cases). SIP does not technically handle double NAT at all, single NAT is all it can handle, so assistance from the router is necessary either using a built-in "SIP algorithm" or with some simple rules about "send all packages for these ports to this IP", but that manual method can make it impossible to have multiple SIP servers (or at least difficult to manage the ports).