it is possible to manual dial in viciphone?

[dhijrwn]

Im trying to enter phone numbers in viciphone it will ring but it will going to unregister the webphone status.

[mflorell]

Are you using the "Phone Only" page? (linked from Reports -> Admin Utilities)

If so, and you are still have problems, post what shows up on the Asterisk CLI when you try to dial.

[dhijrwn]

I'm not using the Phone Only page Matt,

https://imgur.com/a/0ubdPL6

this is the log from cli, when im trying to use the webphone using manual dial in /agc/vicidial.php

[Jul 2 22:47:32] == Using SIP RTP CoS mark 5
[Jul 2 22:47:32] > 0x7f8bb801c8f0 -- Strict RTP learning after remote address set to: 120.29.97.xxx:2417
[Jul 2 22:47:32] -- Executing [17149331234@trunkinbound:1] AGI("SIP/6001-00000055", "agi-DID_route.agi") in new stack
[Jul 2 22:47:32] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jul 2 22:47:33] -- <SIP/6001-00000055>AGI Script agi-DID_route.agi completed, returning 0
[Jul 2 22:47:33] -- Executing [9998811112@default:1] Wait("SIP/6001-00000055", "2") in new stack
[Jul 2 22:47:35] -- Executing [9998811112@default:2] Answer("SIP/6001-00000055", "") in new stack
[Jul 2 22:47:35] -- Executing [9998811112@default:3] Playback("SIP/6001-00000055", "ss-noservice") in new stack
[Jul 2 22:47:35] -- <SIP/6001-00000055> Playing 'ss-noservice.gsm' (language 'en')
[Jul 2 22:47:35] > 0x7f8bb801c8f0 -- Strict RTP learning after ICE completion
[Jul 2 22:47:35] > 0x7f8bb801c8f0 -- Strict RTP switching to RTP target address 120.29.97.xxx:2417 as source
[Jul 2 22:47:40] -- Executing [9998811112@default:4] Playback("SIP/6001-00000055", "vm-goodbye") in new stack
[Jul 2 22:47:40] -- <SIP/6001-00000055> Playing 'vm-goodbye.gsm' (language 'en')
[Jul 2 22:47:40] > 0x7f8bb801c8f0 -- Strict RTP learning complete - Locking on source address 120.29.97.xxx:2417
[Jul 2 22:47:41] -- Executing [9998811112@default:5] Hangup("SIP/6001-00000055", "") in new stack
[Jul 2 22:47:41] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/6001-00000055'
[Jul 2 22:47:41] WARNING[22556][C-0000006d]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Jul 2 22:47:41] -- Executing [h@default:1] AGI("SIP/6001-00000055", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Jul 2 22:47:41] -- <SIP/6001-00000055>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0

[carpenox]

edit your IP's out when u send CLI logs but yes you can manual dial normally, which viciphone are you using? try my version out: https://github.com/carpenox/CyburPhone

[dhijrwn]

edit your IP's out when u send CLI logs but yes you can manual dial normally, which viciphone are you using? try my version out: https://github.com/carpenox/CyburPhone

Hi, this one https://github.com/ccabrerar/ViciPhone, viciphone 2.1 master branch.

[striker]

its dialing over trunkinbound context ,which is meant for inbound in vicdial
Executing [17149331234@trunkinbound:1]

in webrtc templated add context=default
and try to make calls.

[dhijrwn]

its dialing over trunkinbound context ,which is meant for inbound in vicdial
Executing [17149331234@trunkinbound:1]

in webrtc templated add context=default
and try to make calls.

Thank you it is ringing now, but when it is answered I dont hear anything.

[carpenox]

good catch striker, lol i totally missed that

[Sufyan Khan]

its dialing over trunkinbound context ,which is meant for inbound in vicdial
Executing [17149331234@trunkinbound:1]

in webrtc templated add context=default
and try to make calls.

Thank you it is ringing now, but when it is answered I dont hear anything.

Did you find any solution for sound? Same happened to me I can dial from webphone and on other side call is answered but there is no sound. while if I dial from manual dial option from agent panel it works fine. Also I am unable to hear agents call if I dial their sesssion ID from webphone, with softphone I can dial session ID and I can hear agents calls without any issue.

[williamconley]

its dialing over trunkinbound context ,which is meant for inbound in vicdial
Executing [17149331234@trunkinbound:1]

in webrtc templated add context=default
and try to make calls.

Thank you it is ringing now, but when it is answered I dont hear anything.

Did you find any solution for sound? Same happened to me I can dial from webphone and on other side call is answered but there is no sound. while if I dial from manual dial option from agent panel it works fine. Also I am unable to hear agents call if I dial their sesssion ID from webphone, with softphone I can dial session ID and I can hear agents calls without any issue.

Please share your version information. ViciDIAL full version with build, Vicibox Installer full version(or link to instructions if not vBox). WebPhone version.

Errors on your asterisk CLI during an occurrence? Irregularities?

Does the web phone work otherwise? Can you dial directly from a non-web phone successfully?

[carpenox]

The web phone doesn't work like that unfortunately, you need to dial from the manual dial link

[williamconley]

The web phone doesn't work like that unfortunately, you need to dial from the manual dial link

are you suggesting that the web phone in the vicidial interface is only capable of "making noise" in the meetme room, rather than being just like a soft phone? thus the manual dial interface is required to cause Vicidial to generate a call from the meetme room to a Lead?

That sounds about right.

[carpenox]

correct .