MOH playing while we are talking to a client that has phoned from outside
Im using a queue for calls to enter … Also a welcome announcement plays before it enters the queue. My FreePBX version : FreePBX 13.0.101
Also does the same for calls going directly to a extension
Hi,
Do you can confirm that call doesn’t hang up when this happens?
Hi there Thanks for reply
yes the call doesn’t hang up then line is engaged all the time untill i reboot system
Even one side that is hearing MoH, hang up the call, channel will be busy?
Are you using telephony card or SIP trunk from ITSP? and what type of card if your answer is yes?
Thanks for your reply No both sides are hearing the MOH can talk to each other but while talking u can hear MOH. When hang up call the channel is busy .Im using a Digium 8 Port Analog Card
May this is related to you card. you can check logs on Asterisk CLI when this happens?
Not sure what u mean do u think it a faulty card ?
here is the log for when client calls in from outside ,cant see any problem realy.
Note some times the line is busy but i get this error below also
[2016-04-14 09:15:27] WARNING[23041][C-00000007]: app_macro.c:310 _macro_exec: No such context ‘macro-from-dahdi-8’ for macro ‘from-dahdi-8’. Was called by s@from-analog
[2016-04-14 09:15:28] NOTICE[3287]: res_pjsip/pjsip_distributor.c:361 log_unidentified_request: Request from ‘“asterisk” sip:[email protected]’ failed for ‘10.0.0.131:5060’ (callid: [email protected]:5060) - No matching endpoint found
[2016-04-14 09:15:36] NOTICE[12206]: res_pjsip/pjsip_distributor.c:361 log_unidentified_request: Request from ‘“asterisk” sip:[email protected]’ failed for ‘10.0.0.104:5060’ (callid: [email protected]:5060) - No matching endpoint found
[2016-04-14 09:15:36] WARNING[23041][C-00000007]: chan_sip.c:22377 func_header_read: This function can only be used on SIP channels.
[2016-04-14 09:15:36] WARNING[23041][C-00000007]: chan_sip.c:22377 func_header_read: This function can only be used on SIP channels.
[2016-04-14 09:15:36] WARNING[23041][C-00000007]: chan_sip.c:22377 func_header_read: This function can only be used on SIP channels.
[2016-04-14 09:15:36] WARNING[23041][C-00000007]: chan_sip.c:22377 func_header_read: This function can only be used on SIP channels.
botswana*CLI>
Sorry forgot to say that i have changed the incoming route to only do a announcement and then ring to a extension but now it goes direct to voicemail announcement and then the caller can hear the voicemail prompt while u are talking to other person once the voicemail announcement is done then u can talk further with person without the voice announcement
This guy was in IRC yesterday, and getting him to actually tell us what his problem was was like getting blood from a stone. He seems to be doing everything in his power to make it impossible for anyone to help him.
You’ll see that the log he’s linked IS NOT to a person picking up a call, it’s a call going to voicemail.
What we managed to figure out: His calls are coming in via PRI. We have no idea how his extensions are connected, or what sort of phones he’s using. We asked him for logs several times, and he didn’t provide them.
Oh, the Music on Hold only happened when the call went through a queue. I tried to ask him ‘what debugging have you tried’ and he didn’t understand the question. He did say that when he changed an inbound route to go to an extension directly WITHOUT going through a queue, it worked fine.
I assume there’s just a terrible language barrier here.
@corswors it might be easier if you explain the problem in your native language, and we’ll try to translate it ourselves.
Sorry but all u said is not true… It may look like it goes to voicemail but once again im telling u when external call goes to extension i can hear the person picked up call but at the same time the voicemail prompts plays back… I tried various method and of inbound call routes and at the moment is the direct extension. Im using a Snom 300 phone and the phone only rings once
xroba ou’ll see that the log he’s linked IS NOT to a person picking up a call, it’s a call going to voicemail.CALL THE NUMBER YOUR SELF IF U DONT BELIEVE ME 002673971442
-- Executing [s@macro-get-vmcontext:300] NoOp("DAHDI/8-1", "") in new stack
-- Executing [s-NOANSWER@macro-vm:2] VoiceMail("DAHDI/8-1", "8003@default,u") in new stack
-- <DAHDI/8-1> Playing 'vm-theperson.ulaw' (language 'en')
-- <DAHDI/8-1> Playing 'digits/8.ulaw' (language 'en')
-- <DAHDI/8-1> Playing 'digits/0.ulaw' (language 'en')
-- <DAHDI/8-1> Playing 'digits/0.ulaw' (language 'en')
-- <DAHDI/8-1> Playing 'digits/3.ulaw' (language 'en')
-- <DAHDI/8-1> Playing 'vm-isunavail.ulaw' (language 'en')
-- <DAHDI/8-1> Playing 'vm-intro.ulaw' (language 'en')
-- <DAHDI/8-1> Playing 'beep.ulaw' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/8003/tmp/i3pyeV format: wav, 0x7f956c0488f8
I don’t know what to say. That’s a call going to voicemail. You can see it in the log. You say it’s not. What can we do? You can say it’s not going to voicemail, but the log says it is going to voicemail.
This has nothing to do with what you originally said, which is calls going to Queues have music on hold played at the same time.
Edit: By any chance is this a virtual machine? Could there be two identical virtual machines running at the same time?
I did not say it did not go to voicemail there is no indication of saying NO or DID NOT play voicemail I said the voicemail recording plays back while im talking to the caller on the other side of the call LIKE talking on a phone u know what i’m talking about ? there is a 2 way conversation while the voicemail message plays back. Yes its got nothing to do with orginal fault of queue but its still similar im trying different things like u asked and now u keep on not understanding
If I wasn’t lying in bed right now, I’d ban you, but I need to do that from my desktop PC. May I strongly suggest you tone your messages down, and actually respond to the questions asked?
To answer your question, I am Australian, and English is my native language. I can also use the word “you” instead of the letter “u”.
So. Feel free to try again. This time without the sass.
There is another Asterisk PABX but its is running on a different IP address and has its own analog lines and registered phones registered on that server’s ip address .