vicidial.org

Hello everyone,
my configuration is

ViciBox Redux v.5.0.2-130821| VERSION: 2.8-432a BUILD: 140404-1007 | asterisk-1.8.3 | Single Server |4 port allo gsm card | No Extra Software After Installation | Intel(R) Core(TM) i5-3450 CPU @ 3.10GHz

I want to transfer incoming call to my mobile(09971407XXX).
for this i make a did 874591511, select DID Route as EXTEN, and put Extension value 09971407XXX.

when i tried to make call to my server, the call got disconnect after tow ring leaving cli> message

[Jun 3 14:17:05] -- Executing [s@trunkinbound:1] Answer("GSM/2", "") in new stack
[Jun 3 14:17:05] event=OK
[Jun 3 14:17:06] -- Executing [s@trunkinbound:2] Goto("GSM/2", "trunkinbound,874591511,1") in new stack
[Jun 3 14:17:06] -- Goto (trunkinbound,874591511,1)
[Jun 3 14:17:06] -- Executing [874591511@trunkinbound:1] AGI("GSM/2", "agi-DID_route.agi") in new stack
[Jun 3 14:17:06] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jun 3 14:17:06] -- <GSM/2>AGI Script agi-DID_route.agi completed, returning 0
[Jun 3 14:17:06] -- Executing [09971407XXX@default:1] Answer("GSM/2", "") in new stack
[Jun 3 14:17:06] -- Executing [09971407XXX@default:2] Dial("GSM/2", "GSM/2/09971407XXX,,To") in new stack
[Jun 3 14:17:06] DEBUG[13610]: chan_gsm.c:3878 gsm_request: [ALLO_GSM] gsm_data '2/09971407XXX'
[Jun 3 14:17:06] portstr:'2' ext:'09971407XXX' gsm_data '2/09971407XXX'
[Jun 3 14:17:06] WARNING[13610]: chan_gsm.c:3839 check_and_fetch_port: There is already a call on port :2
[Jun 3 14:17:06] ERROR[13610]: chan_gsm.c:3920 gsm_request: [ALLO_GSM] port may be invalid !!!!!!!!!!!
[Jun 3 14:17:06] WARNING[13610]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'GSM' (cause 0 - Unknown)
[Jun 3 14:17:06] == Everyone is busy/congested at this time (1:0/0/1)
[Jun 3 14:17:06] -- Executing [09971407XXX@default:3] Hangup("GSM/2", "") in new stack
[Jun 3 14:17:06] == Spawn extension (default, 09971407XXX, 3) exited non-zero on 'GSM/2'
[Jun 3 14:17:06] -- Executing [h@default:1] AGI("GSM/2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
[Jun 3 14:17:06] -- <GSM/2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


please let me know if i miss any information.
Thank you

You recieved an incoming call on 'GSM/2' then tried to dial out on 'GSM/2'. This obviously wont work. How many channels is your GSM device?

first of all thank you for your reply geoff3dmg
i have two channels, GSM/1 and GSM/2, but i dont know how to transfer the call to my mobile using gsm1 when their is incoming call on gsm2.
please help me to solve this problem.

update:
here i do some changes, change gsm1 context to trunkinbound for Inbound calling and gsm2 to outbound(context).
now when i make incoming call to gsm1,the call will transfer to my mobile (9971407XXX) with the help gsm2(outbound) and my mobile start ringing but when i receive the call,
their is no voice from both side but both the phone (caller and receiver) remain in call till the phone hangup by one.
Am i doing right,if yes then why i can't hear any sound.

please help...

Before you can get Vicidial to send calls through (or receive calls from) any trunk, the trunk must work in Asterisk. Please find the necessary manuals/sales rep/etc to make the trunk work in Asterisk (a MUCH larger platform with many support portals) and then come back here for the "work in Vicidial" portion. Not saying we won't help at all, but I am saying that you have a piece of hardware that has drivers specifically built for ... Asterisk. There is undoubtedly support from the manufacturer and the sales channel to make that happen. Once that's done, Vicidial becomes easy. Otherwise, you may need to find someone in the Vicidial forums who has direct experience in this hardware and that may not be so easy to do.

Begin by posting the actual model number of the gsm card and perhaps even a link to the purchase site and/or manual if you've found it online already.

That being said, you should try to get the call from the GSM into a SIP phone for a logged in agent. After that's been done, the rest is "configuration". Jumping straight to a non-standard calling path often adds confusion to the issue and overloads your IT learning curve. Just have the call arrive and go to a SIP phone. Also practice by having a SIP call arrive first so you can see how that method works, then perhaps you can apply that knowledge to the GSM channel with a bit more experience under your belt.

One of my favorite troubleshooting methods in Vicidial: Start with (any) known working configuration and change ONE thing at a time (while testing with each change) until you get to your desired configuration. It may take a long time, but it's often free if approached from that direction