WEBRTC - Outbound Calls does not Work!

[d001]

I have installed the webphone on my server. Inbound calls work okay, but outbound calls do not.

These are my CLI logs:

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 Using SIP RTP CoS mark 5
       > 0x7feea8004320 -- Strict RTP learning after remote address set to: xx.xx.xx.xx:50974
 -- Executing [355xxxxxxxxx@trunkinbound:1] AGI("SIP/5001-00000037", "agi-DID_route.agi") in new stack
     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
     -- <SIP/5001-00000037>AGI Script agi-DID_route.agi completed, returning 0
     -- Executing [355xxxxxxxxx@trunkinbound:2] Hangup("SIP/5001-00000037", "") in new stack
     == Spawn extension (trunkinbound, 355xxxxxxxxx, 2) exited non-zero on 'SIP/5001-00000037'
 WARNING[20922][C-00000061]: func_hangupcause.c:138 hangupcause_read: Unable to find information for channel
     -- Executing [h@trunkinbound:1] AGI("SIP/5001-00000037", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
     -- <SIP/5001-00000037>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0

The server is installed on a local device and configured with GoIP. I have three active trunks connected to a primary trunk, which selects the numbers randomly. It works okay with X-Lite for both inbound and outbound calls, but not with the webphone.

The agent extension successfully logs into the conference and hears "You are the only one in the conference." However, when I try to make an outbound call from the phone pad, it displays "Rejected (603 - Declined)," and the CLI shows the above log.

I have installed https://github.com/carpenox/CyburPhone

[d001]

Resolved!

I just added allow=g729 to the carrier configuration.

Then, I restarted Asterisk with the command:

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sudo systemctl restart asterisk

Now, I can make outbound and inbound calls from the WEBRTC phone, but I am encountering another problem. It always displays:

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No one is in your session: 8600051
Go Back

Call Agent Again

When I check the SIP peers in the Asterisk CLI:

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vicibox11*CLI> sip show peers
Name/username             Host                Dyn Forcerport Comedia ACL Port    Status      Description
5001/5001                 192.168.1.xxx       D   Yes        Yes     52530       OK (9 ms)
trunkA/355xxxxxxxxx       192.168.1.xxx       D   Yes        Yes     5060        OK (2 ms)
trunkB/355xxxxxxxxx       (Unspecified)       D   Yes        Yes     0           UNKNOWN
trunkC/355xxxxxxxxx       (Unspecified)       D   Yes        Yes     0           UNKNOWN

My phone and carrier are registered.

According to this answer by dreedy on the Vicidial forum here 'http://www.vicidial.org/VICIDIALforum/viewtopic.php?f=4&t=41623&p=150883&hilit=No+one+is+in+your+session%3A+8600051#p152907',
I should add context=default at the end of the template in Admin > Templates, but it didn’t work for me.

Also, i tried and this one but it didn’t work 'http://www.vicidial.org/VICIDIALforum/viewtopic.php?f=4&t=40077&p=142884&hilit=No+one+is+in+your+session%3A+8600051#p142861'

Does anyone have any idea what else I can try?

[williamconley]

Are you saying you have a soft phone registered to your vicidial phone account as a sip extension? And you want to use this same extension as a WebRTC phone?

Having a "peer" does not mean that there is a live call (which is a requirement for "someone in your session"). When you log in with Viciphone, you should be presented with a link to "call the agent" immediately after login. Pushing that button will cause the WebRTC phone to ring and then auto-answer which creates a call between your WebRTC phone and an asterisk MeetMe room.

[d001]

Are you saying you have a soft phone registered to your vicidial phone account as a sip extension? And you want to use this same extension as a WebRTC phone?

No, I have done the following as usual:

Generated SSL certificates.
Cloned the Vicidial VICIphone WebRTC repository.
Navigated to Admin >> Phone >> and set "Webphone" to "Yes" with: Template ID: VICIphone - VICIphone WebRTC.

As I understand it (though I am not a specialist in system administration, Asterisk, or networking; I am just a software developer),
once I set "Webphone" to "Yes," the phone (SIP) will always be registered/connected with the user.
This is confirmed when I use the sip show peers command and the status shows as "OK."

Having a "peer" does not mean that there is a live call (which is a requirement for "someone in your session"). When you log in with Viciphone, you should be presented with a link to "call the agent" immediately after login. Pushing that button will cause the WebRTC phone to ring and then auto-answer which creates a call between your WebRTC phone and an asterisk MeetMe room.

Yes, I understand.
Actually, when the agent is logged in it causes the WebRTC phone to ring and creates a call between your WebRTC phone and an Asterisk MeetMe room.
However, when I make a call, it displays "No one is in your session: 8600051."

Thank you very much for your support.

Best regards,

[williamconley]

Progress.

FYI: BEFORE logging in as an agent with a ViciPhone that has not been used today:

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ViciPhone/ViciPhone       (Unspecified)                            D  Yes        Yes            0        UNKNOWN

AFTER logging in (with agent still logged in):

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ViciPhone/ViciPhone       100.100.100.100                           D  Yes        Yes            59665    OK (48 ms)

AFTER logging out again (even after 5 minutes: it seems to lose registration around 10-12 minutes after logout):

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ViciPhone/ViciPhone       100.100.100.100                           D  Yes        Yes            59665    OK (48 ms)

Thus the assertion that changing to Yes for Webphone is the catalyst for registering is not quite correct. That first phone call, however, does appear to register. And that registration may last for quite a while. Also note that while WebRTC phone installs may differ: The WebRTC phone is not generally Invoked until the agent logs in.

when the agent is logged in it causes the WebRTC phone to ring and creates a call between your WebRTC phone and an Asterisk MeetMe room.

Does the agent hear (and I Quote!) "You are currently the only person in this conference"?

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asterisk -rx "core show calls"
asterisk -rx "sip show channels"

[d001]

FYI: BEFORE logging in as an agent with a ViciPhone that has not been used today:

When the agent is not logged in, the status is unreachable as shown below:

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Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
1001/1001                 192.1xx.x.xxx                            D  Yes        Yes            51272    UNREACHABLE


AFTER logging in (with agent still logged in):

When the agent is logged in:

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Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
1001/1001                 192.1xx.x.xxx                            D  Yes        Yes            51325    OK (24 ms)

However, I logged in all my agents and this happened...

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vicibox11*CLI>
[Aug  5 14:30:22] NOTICE[13414]: chan_sip.c:25047 handle_response_peerpoke: Peer 'TrunkA' is now Lagged. (2573ms / 2000ms)
[Aug  5 14:30:24] NOTICE[29674]: chan_sip.c:25047 handle_response_peerpoke: Peer '1998' is now Lagged. (2344ms / 2000ms)
[Aug  5 14:30:37] NOTICE[13414]: chan_sip.c:30637 sip_poke_noanswer: Peer 'TrunkA' is now UNREACHABLE!  Last qualify: 2573
[Aug  5 14:30:38] NOTICE[13414]: chan_sip.c:30637 sip_poke_noanswer: Peer '1998' is now UNREACHABLE!  Last qualify: 2344
vicibox11*CLI>
[Aug  5 14:31:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug  5 14:31:04] NOTICE[13414]: chan_sip.c:30637 sip_poke_noanswer: Peer '1001' is now UNREACHABLE!  Last qualify: 13
[Aug  5 14:31:07]   == Manager 'sendcron' logged on from 127.0.0.1
vicibox11*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
1001/1001                 192.168.x.xxx                            D  Yes        Yes            51349    UNREACHABLE
1998/1998                 192.168.x.xxx                            D  Yes        Yes            52326    LAGGED (3429 ms)
TrunkA/355xxxxxxxxx             8x.9x.xx.xx                                 Yes        Yes            5060     LAGGED (3920 ms)
gs102/gs102               (Unspecified)                            D  Yes        Yes            0        UNKNOWN
4 sip peers [Monitored: 2 online, 2 offline Unmonitored: 0 online, 0 offline]
[Aug  5 14:31:59] NOTICE[13414]: chan_sip.c:30637 sip_poke_noanswer: Peer '1001' is now UNREACHABLE!  Last qualify: 2658
[Aug  5 14:31:59] NOTICE[13414]: chan_sip.c:30637 sip_poke_noanswer: Peer 'TrunkA' is now UNREACHABLE!  Last qualify: 2811
[Aug  5 14:32:01] NOTICE[13414]: chan_sip.c:30637 sip_poke_noanswer: Peer '1998' is now UNREACHABLE!  Last qualify: 3813

AFTER logging out again (even after 5 minutes: it seems to lose registration around 10-12 minutes after logout):

I do not understand this very well. If the agent logs out, it disconnect the user in real time unless I'm misunderstanding something...

Does the agent hear (and I Quote!) "You are currently the only person in this conference"?

Yes, the agent hears the message: "You are currently the only person in this conference."

asterisk -rx "core show calls"
asterisk -rx "sip show channels"

logs:

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vicibox11:~ # asterisk -rx "core show calls"
1 active call
6 calls processed
vicibox11:~ #
vicibox11:~ #
vicibox11:~ # asterisk -rx "sip show channels"
Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry     Peer
192.1xx.x.xxx    1001             67293c7.......x  (ulaw)           No       Tx: ACK                    1001
8x.9x.xx.xx      (None)           0391fd8.......x  (nothing)        No       Rx: OPTIONS                <guest>
2 active SIP dialogs
vicibox11:~ #

Thanks for your time!

[williamconley]

AFTER logging out again (even after 5 minutes: it seems to lose registration around 10-12 minutes after logout):

I do not understand this very well. If the agent logs out, it disconnect the user in real time unless I'm misunderstanding something...

The OK/REACHABLE status does NOT mean there's a call active in the system.

sip show peers does NOT show active calls. It merely shows "accounts". Like empty parking spaces. NO cars in 'em. LOL: Just a spot where a car might go should one appear.

These two commands, however, show Active Calls:

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asterisk -rx "core show calls"
asterisk -rx "sip show channels"

Signalling between accounts and the server aren't necessary to delve into until there's a problem. Like ... NO call between the asterisk server and the agent, that would be a problem. But the phrase "you are the only person in this conference" is a clear indicator that there IS a live call between asterisk and agent in a meetme room. I say this because that audio message is generated by the "Meetme room" in which your agent has just landed by answering a phone call via WebRTC.

All that being said: DO NOT make calls from a phone pad. Calls in Vicidial are made from the Vicidial web interface. In the case of a manually dialed call, the instructions for an agent to make a call include clicking on "manual dial" and entering the phone number in the field provied on the Manual Dial web interface and clicking "dial". None of that involves using a phone pad.

If you are dialing a phone on a phone pad, it's not part of the Vicidial interface making a phone call. You'll not get any client data on the agent's web interface and Vicidial won't have any stats on the call.

Do your agents using VOIP phones use their phone pads to make calls this way?

[d001]

All that being said: DO NOT make calls from a phone pad. Calls in Vicidial are made from the Vicidial web interface. In the case of a manually dialed call, the instructions for an agent to make a call include clicking on "manual dial" and entering the phone number in the field provied on the Manual Dial web interface and clicking "dial". None of that involves using a phone pad.

If you are dialing a phone on a phone pad, it's not part of the Vicidial interface making a phone call. You'll not get any client data on the agent's web interface and Vicidial won't have any stats on the call.

Do your agents using VOIP phones use their phone pads to make calls this way?

I didn't know this. Currently, my agents are making calls using the manual dialing option, connecting SIP numbers with the external softphone (X-Lite). However, sometimes they forget to answer the calls that come through X-Lite. We are considering adding WebRTC, which would always be connected.

The OK/REACHABLE status does NOT mean there's a call active in the system.

sip show peers does NOT show active calls. It merely shows "accounts". Like empty parking spaces. NO cars in 'em. LOL: Just a spot where a car might go should one appear.

These two commands, however, show Active Calls:

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asterisk -rx "core show calls"
asterisk -rx "sip show channels"

Signalling between accounts and the server aren't necessary to delve into until there's a problem. Like ... NO call between the asterisk server and the agent, that would be a problem. But the phrase "you are the only person in this conference" is a clear indicator that there IS a live call between asterisk and agent in a meetme room. I say this because that audio message is generated by the "Meetme room" in which your agent has just landed by answering a phone call via WebRTC

Very clear!


Thanks a lot William.

[williamconley]

If you work with us on the forum and actually answer questions (which is quite refreshing, BTW, thanks) we can often figure out where you are and perhaps fill you in on things you simply haven't had experience with yet that May help you understand Asterisk/Vicidial a bit better.

Now read the Vicidial Manager's Manual. Take it with you to lunch. It's got A LOT of Vicidial features you will eventually use (and of course a lot you won't, LOL). Can't hurt, right?

[d001]

Thank you for the advice! I appreciate the encouragement and will definitely take a closer look at the Vicidial Manager's Manual. I'm always eager to learn more and improve my understanding.

The Vicidial Manager's Manual has been extremely helpful in configuring some servers, especially since I didn’t have a mentor to guide me. The forum has served as a valuable resource, guiding me through each problem. Although finding the solution sometimes took time, I eventually found what I needed.

Also, whenever I've found a solution, I've posted it here and tried to share it in an easy-to-understand way, knowing that beginners like me might be looking for the solution too.

Thanks again!

[carpenox]

welcome to the community

[d001]

Thank you! my pleasure