Viciphone 2.0

[williamconley]

Code: Select all

[Jun 12 09:28:24] CANCEL sip:8q8hlhsg@192.0.2.53;transport=wss SIP/2.0

your first line is CANCEL. What happened before it? I would assume the previous line (or lines) resulted in the cancel. But we can't see them.

[carpenox]

This seems to be the underlying issue:

] ERROR[6241]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[Jun 12 16:37:49] ERROR[6241]: tcptls.c:488 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
[Jun 12 16:37:49] == WebSocket connection from 'xx.192.xx.210:64550' forcefully closed due to fatal write error


Where is Viciphone trying to write information to?

[ccabrera]

Guys,

You might want to try the current code from Github. I've found out a variable was being wrongly referred, and ending up giving an error when trying to establish the initial call.

Just update to most current release and try again. I also added an 'auto login' feature so that Vicidial calls the webphone as soon as it gets registered.

Regards,

[williamconley]

Guys,

You might want to try the current code from Github. I've found out a variable was being wrongly referred, and ending up giving an error when trying to establish the initial call.

Just update to most current release and try again. I also added an 'auto login' feature so that Vicidial calls the webphone as soon as it gets registered.

Regards,

I presume you refer to https://enlaza.mx/tools/viciphonev2/viciphone.php rather than https://github.com/vicimikec/ViciPhone

Correction: https://github.com/ccabrerar/ViciPhone

Now all three links in one place.

[carpenox]

actually i think its: https://github.com/ccabrerar/ViciPhone

[carpenox]

It works now thx

[ccabrera]

actually i think its: https://github.com/ccabrerar/ViciPhone

Yep, that's it. The one I have from my company URL (enlaza.mx) is a live demo, but I also use it for testing, so I don't recommend using it for production.


If you happen to find any other bugs or pull request, you may send it so from the Github project page.

[Smartercom]

Hi to all, i want to test this viciphone 2.0 and i'm exciting to contribute , now i have installed webphone and all is working, new webphone seems more fast but i have this error in asterisk cli , i have simulated a call to my number 3933150xxxxx and i have rejected call but i have this error random even if all is working...is really serious problem? Is related to viciphone or i have done some errors in configuration?..

port 8089 is opened and asterisk is listen by launching this command
netstat -tulpan| grep 8089

i have this result

tcp 0 0 0.0.0.0:8089 0.0.0.0:* LISTEN 2245/asterisk

if i launch this command asterisk -rx "http show status"

i have this result

HTTPS Server Enabled and Bound to 0.0.0.0:8089

if i go on https://mydomain.it:8089/ws

i see

Upgrade Required
Asterisk/13.29.2-vici

this is complete log for dial attempt to my number 33150xxxxx..can someone help me?

[Jun 16 03:17:06] ERROR[2379]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[Jun 16 03:17:06] ERROR[2379]: tcptls.c:466 tcptls_stream_close: [Jun 16 03:17:03] Connected to Asterisk 13.29.2-vici currently running on Testmarco (pid = 2266)
[Jun 16 03:17:06] ERROR[2379]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[Jun 16 03:17:06] ERROR[2379]: tcptls.c:466 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
[Jun 16 03:17:06] == WebSocket connection from '185.178.95.241:55197' forcefully closed due to fatal write error
[Jun 16 03:17:07] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 16 03:17:07] == Manager 'sendcron' logged off from 127.0.0.1
[Jun 16 03:17:08] ERROR[2379]: chan_sip.c:4271 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jun 16 03:17:13] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 16 03:17:13] -- Called 8600051@default
[Jun 16 03:17:13] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000001;2", "8600051,F") in new stack
[Jun 16 03:17:13] -- Local/8600051@default-00000001;1 answered
[Jun 16 03:17:13] -- Executing [3933150xxxxx@default:1] AGI("Local/8600051@default-00000001;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jun 16 03:17:13] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=200))
[Jun 16 03:17:13] -- <Local/8600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jun 16 03:17:13] -- Executing [3933150xxxxx@default:2] Dial("Local/8600051@default-00000001;1", "SIP/33150xxxxx@Prova1,,tTo") in new stack
[Jun 16 03:17:13] == Using SIP RTP CoS mark 5
[Jun 16 03:17:13] -- Called SIP/33150xxxxx@Prova1
[Jun 16 03:17:14] == Manager 'sendcron' logged off from 127.0.0.1
[Jun 16 03:17:20] > 0x7fcc58013ad0 -- Strict RTP learning after remote address set to: 46.30.248.61:4740
[Jun 16 03:17:20] -- SIP/Prova1-00000006 is making progress passing it to Local/8600051@default-00000001;1
[Jun 16 03:17:20] > 0x7fcc58013ad0 -- Strict RTP switching to RTP target address 46.30.248.61:4740 as source
[Jun 16 03:17:20] -- SIP/Prova1-00000006 is ringing
[Jun 16 03:17:20] -- SIP/Prova1-00000006 is making progress passing it to Local/8600051@default-00000001;1
[Jun 16 03:17:22] -- Got SIP response 486 "Busy Here" back from 149.202.190.63:5060
[Jun 16 03:17:22] -- SIP/Prova1-00000006 is busy
[Jun 16 03:17:22] == Everyone is busy/congested at this time (1:1/0/0)
[Jun 16 03:17:22] -- Executing [3933150xxxxx@default:3] Hangup("Local/8600051@default-00000001;1", "") in new stack
[Jun 16 03:17:22] == Spawn extension (default, 3933150xxxxx, 3) exited non-zero on 'Local/8600051@default-00000001;1'
[Jun 16 03:17:22] -- Executing [h@default:1] AGI("Local/8600051@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY---------------SIP 486 Busy Here)") in new stack
[Jun 16 03:17:22] -- <Local/8600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -------SIP 486 Busy Here) completed, returning 0
[Jun 16 03:17:22] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000001;2'
[Jun 16 03:17:22] WARNING[3836][C-00000004]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Jun 16 03:17:22] -- Executing [h@default:1] AGI("Local/8600051@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17--------------------)") in new stack
[Jun 16 03:17:22] -- <Local/8600051@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[Jun 16 03:17:22] ERROR[2379]: chan_sip.c:4271 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jun 16 03:17:36] ERROR[2379]: chan_sip.c:4271 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
[Jun 16 03:17:50] ERROR[2379]: chan_sip.c:4271 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data

[ccabrera]

Whats your config in sip-vicidial.conf for your phone prova1?

I believe the problem is in your template, rather than on httpd.conf

[carpenox]

seems like youre having same issue i was having before i switched to new code, did u delete the directory before reinstalling?

[Smartercom]

I have found the problem, errors appears when phone is not reacheble, so it is as system gives error message when do not qualify phones..infact i have removed qualify in template conf and now error is gone away but i have lose information about registration of my peers.

This is my template rtc

type=friend
host=dynamic
encryption=yes
icesupport=yes
directmedia=no
qualify=no
transport=wss
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/certbot/live/mydomain.it/cert.pem
dtlsprivatekey=/etc/certbot/live/mydomain.it/privkey.pem
dtlssetup=actpass
rtcp_mux=yes


are you agree with this modifications?or there is another clean way to resolve issue?Anyway last code of viciphone 2.0 works well ..more fast and clean.
Waiting for suggestions.Thanks a lot!!

[Smartercom]

seems like youre having same issue i was having before i switched to new code, did u delete the directory before reinstalling?

Yes i have deleted old folder and used folder from github


now i have only this 2 lines of errors:

Code: Select all

[Jun 17 10:00:38] ERROR[2385]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[Jun 17 10:00:38] ERROR[2385]: tcptls.c:466 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor



all working well, can you help me?

[carpenox]

is your SSL cert valid from letsencrypt? are you seeing a lock when your go to the webpage your hosting? such as vicidial interface

[Smartercom]

Yes lock Is present and i use certbot for ssl.

[ccabrera]

Then perhaps the httpd.conf is not using the same certificates as the template within sip-vicidial.conf

[carpenox]

i had that error but once i did "rm -rf ViciPhone" and git clone from ccabrera github after the bug fix its been working fine. Did you restart asterisk and apache services or the server itself?

[Smartercom]

Then perhaps the httpd.conf is not using the same certificates as the template within sip-vicidial.conf

in my httpd.conf there is no reference to certificates as in template, where i have to control if certificates are the same?

[ccabrera]

I suggest you read https://viciphone.com/?page_id=351

[Smartercom]

i had that error but once i did "rm -rf ViciPhone" and git clone from ccabrera github after the bug fix its been working fine. Did you restart asterisk and apache services or the server itself?

downloaded from here https://codeload.github.com/ccabrerar/V ... zip/master and installed in folder after deleted all....is link right?

[Smartercom]

i had that error but once i did "rm -rf ViciPhone" and git clone from ccabrera github after the bug fix its been working fine. Did you restart asterisk and apache services or the server itself?

downloaded from here https://codeload.github.com/ccabrerar/V ... zip/master and installed in folder after deleted all....is link right?

i have rebooted all and now no errors displayed....i'mm test it and stress more and will update here..Thanks a lot!!!

[carpenox]

np, glad i could help. and thx again ccabrera for your new coding

[Smartercom]

Hi to all, i'm testing viciphone 2.0 again with success.Today i have found a little problem that i would resolve with your help...when admin configured with webphone want to use a listen or barge function the channel used in asterisk remain opened....it seems like asterisk ignore it even if i use hangup request all or hangup request channelname...with zoiper i can close channel by hungup button....is possible with webphone? i have to use zoiper to avoid this problem but i would use only webphone for all purpose.

[carpenox]

ok so within the campaign summary screen where u click listen or barge, you set it to an extension correct? Just go back into more options where you set that extension and clear it out and hit submit again. it will hang it up. then u can go back in and set the extension again to get ready for the next one.

[Smartercom]

hi to all, i have discovered issue with viciphone 2.0 from dev version https://enlaza.mx/tools/viciphonev2/viciphone.php.
Problem appear with tone for ivr ...all buttons i submit is not working only button working is 3...can you help me?

[williamconley]

https://enlaza.mx/tools/viciphonev2/viciphone.php

https://github.com/vicimikec/ViciPhone

https://github.com/ccabrerar/ViciPhone

Maybe You'll Get Lucky with another source.

[Smartercom]

https://enlaza.mx/tools/viciphonev2/viciphone.php

https://github.com/vicimikec/ViciPhone

https://github.com/ccabrerar/ViciPhone

Maybe You'll Get Lucky with another source.

thanks for reply but

https://enlaza.mx/tools/viciphonev2/viciphone.php not working tones

https://github.com/vicimikec/ViciPhone viciphone 1.0 give me errors with descriptor and tls

https://github.com/ccabrerar/ViciPhone is the same i have used with not working tones

anyone has other solutions?

[williamconley]

There is a distinct possibility you should consider posting details about your problem. Were there any errors during your installation? Or during the pressing of a button? In asterisk or in apache ...?

Also: Have you confirmed that "buttons for the IVR" work when using a different phone? It's possible the problem is systemic and not related to Viciphone, after all.

[ccabrera]

There seems to be a issue with DTMFs, which I never noticed because I always hide the dialpad so the agents can't try to use it as a regular phone.

While I fix that, you can still use Vicidial's own DTMF SEND box, and that works just fine.

[ccabrera]

Did some changes and tested it. Before changes I was able to replicate the error, now DTMFs get sent as INFO just fine.

The fix was pushed to Github. Download the new code and try again (please remember to clear cache).

[carpenox]

I will unload it this morning and report any bugs. thx

[carpenox]

OK I have switched this out with the updated one. Everything works fine including DTMF so far....

[ccabrera]

I would suggest that any further issues are opened through Github so I can reply and patch them on the same place.

[carpenox]

will do, thanks

[carpenox]

can viciphone support multiple channels? like if you try to monitor thru realtime with viciphone, you can just click into another agent without first clearing out the webphone and entering again or switching back and forth between webphones

[andicrb76]

Anyone can solve this problem ?
I have the same problem , and i'm using PBXWebPhone and viciphone, same result

[carpenox]

which problem?

[carpenox]

i made some changes to your fork christian, if anyone wanted to see them: https://github.com/carpenox/CyburPhone

[dspaan]

Is there any news about the new viciphone?

[ccabrera]

@carpenox, seems like you downloaded my code and uploaded it again as a completely new project, so changes are much harder to compare. Please next time do a fork from my project and do the changes as a pull request so comparison is simpler. Besides, I downloaded your Cyburphone code into a branch of mine and couldn't find much differences aside from aesthetics and name, so if you could let me what to look at, would make my work easier.

Right now I'm working on a 2.1 branch. If tests go by as expected, I'll release it soon. You may download the current test code from https://github.com/ccabrerar/ViciPhone/tree/v2.1 . So far it includes:

- Fix for a re-registration bug which causes Viciphone to display as "Registered" instead of "Incall" after 10 minutes of logging in. This causes agents to think call has been dropped, while it is only aesthetic (call is still on, so many of you would notice if you are not looking at the webphone display).
- Updated sipjs to version 0.15.11
- I would like to include a spectralizer (like this one: https://webrtc.github.io/samples/src/co ... ia/volume/) so there's visual feedback while on a call.

After this 2.1 branch gets merged into master, I'll set it as a new release and begin working on v3 to use sipjs 0.19, which is completely incompatible with sipjs 0.15 used by Viciphone 2.x.

[dspaan]

Oh that fix for re-registration would be nice, i was always wondering why it sometimes says registered instead of incall.