Migration of WCS4/Asterisk/MeetMe project to WCS5
- Thread starter marcw
- Start date
Okay, next problem. It is Asterisk related but I hope you can give me a hint anyway because the Asterisk configuration on my other server was coded by Alex. I am using the same configuration on the new server (adjusted where needed) but when now the callee pick up the phone and MeetMe put him into the conference the callee is confronted with the message that he have to enter the conference number. Obviously the reason for this is that the dstchannel is always empty.
Here are the extensions:
Do you have any idea what the reason could be?
Best
Marc
Code:
- Executing [s@fun1:2] NoOp("SIP/mpapp2-00000000", "INIT_CHANNEL: SIP/mpapp2-00000000") in new stack
-- Executing [s@fun1:3] Set("SIP/mpapp2-00000000", "SHARED(INIT_CHANNEL,)=SIP/mpapp2-00000000") in new stack
-- Executing [s@fun1:4] Set("SIP/mpapp2-00000000", "SHARED(CONF,)=00491777000001") in new stack
Code:
[fun1]
exten => s,1,NoOp(${CONF})
exten => s,n,NoOp(INIT_CHANNEL: ${CDR(channel)})
exten => s,n,Set(SHARED(INIT_CHANNEL,${CDR(dstchannel)})=${CDR(channel)})
exten => s,n,Set(SHARED(CONF,${CDR(dstchannel)})=${CONF})
exten => s,n,MeetMe(${CONF},Akmqd)
exten => s,n,Hangup
[fun2]
exten => s,1,NoOp(${SHARED(CONF)})
exten => s,n,Wait(0.5)
exten => s,n,NoOp(${SHARED(INIT_CHANNEL)})
exten => s,n,MeetMe(${SHARED(CONF)},kxqd)
exten => s,n,AGI(kill_channel.sh,${SHARED(INIT_CHANNEL)})
exten => s,n,NoOp("NOT_IN_THE_MEETME")
exten => s,n,HangupBest
Marc
Solved! It sounds stupid but it seems that the macros from the extensions.conf in Asterisk 11 only worked due to a bug. The dest channel normally is not overgivable if setting the G option.
Anyway, if someone should find this thread anywhen in the future here is my solution. Please compare it with the code I entered above:
Also an adjustment in another part was necessary:
Best regards
Marc
Anyway, if someone should find this thread anywhen in the future here is my solution. Please compare it with the code I entered above:
Code:
[separator]
exten => s,1,Goto(fun1,s,1)
exten => s,2,Goto(fun2,s,1)
[fun1]
exten => s,1,NoOp(Data 1: CONF=${CONF} INIT_CHANNEL=${INIT_CHANNEL})
exten => s,n,MeetMe(${CONF},Akmqd)
exten => s,n,Hangup
[fun2]
exten => s,1,NoOp(Data 2: CONF=${CONF} INIT_CHANNEL=${INIT_CHANNEL})
exten => s,n,Wait(0.5)
exten => s,n,MeetMe(${CONF},kxqd)
exten => s,n,AGI(kill_channel.sh,${INIT_CHANNEL})
exten => s,n,NoOp("NOT_IN_THE_MEETME")
exten => s,n,HangupThe underline char made also a difference.
Best regards
Marc
Hi Max!
Here I am back. Like old shoes you cannot throw away.
I have set up now my calling proccess as you mentioned in this thread a longer time ago. So I
1. start a phone call with the client2/examples/demo/sip/phone/phone.html
2. find the conference-id in the asterisk conferences folder
3. create a .call file containing i.E.
-----
Channel: Local/s@inject-sound
Application: Playback
Data: 15_14
Archive: Yes
Set: CONF=cde54620-d931-11e8-a543-4303c23be106
----
4. move the created file into the asterisk outgoing folder
The funny thing is that all seems to work fine and also the asterisk console tells me that the sln file is played but nothing is hearable. I can hear the callee's voice in the phone.html but not the sln file I move to the outgoing folder.
from Asterisk console
Output from ARC http://locophono.com:8081/rest-api/call/find_all
(just for debugging purposes):
What am I doing wrong? (WCS 5.1.3546)
Best regards
Marc
Here I am back. Like old shoes you cannot throw away.
I have set up now my calling proccess as you mentioned in this thread a longer time ago. So I
1. start a phone call with the client2/examples/demo/sip/phone/phone.html
2. find the conference-id in the asterisk conferences folder
3. create a .call file containing i.E.
-----
Channel: Local/s@inject-sound
Application: Playback
Data: 15_14
Archive: Yes
Set: CONF=cde54620-d931-11e8-a543-4303c23be106
----
4. move the created file into the asterisk outgoing folder
The funny thing is that all seems to work fine and also the asterisk console tells me that the sln file is played but nothing is hearable. I can hear the callee's voice in the phone.html but not the sln file I move to the outgoing folder.
Of course I tried to change out the stripCodec value with PCMU, PCMA and G722 but without any success.
from Asterisk console
Output from ARC http://locophono.com:8081/rest-api/call/find_all
(just for debugging purposes):
What am I doing wrong? (WCS 5.1.3546)
Best regards
Marc
Last edited:
Hello Marc.
It seems like Asterisk uses PCMA (alaw) codec. So check if file that you try to playback on Asterisk uses this codec too.
stripCodec forces WCS do not use codecs listed. For example if Asterisk wants to use PCMA you can exclude other codecs like this:
Code:
stripCodecs: "opus,OPUS,PCMU,G722,G729"This implementation must be the hell's prove for me. I never felt so stupid like in the past weeks (no: months! ) Whatever I do, whatever I try, I fail. So I ask you politely to log into my server to find out why I cannot simply play an audio file by moving the .call file into the outgoing folder. The Asterisk log tails me that the file is playing correctly but nothing is to hear. Also localAudioCodec and remoteAudioCodec are both PCMA now. I send my login data and the testing setup to support@flashphoner.com . Feel free to charge me something for your effort.
Best regards
Marc
) Whatever I do, whatever I try, I fail. So I ask you politely to log into my server to find out why I cannot simply play an audio file by moving the .call file into the outgoing folder. The Asterisk log tails me that the file is playing correctly but nothing is to hear. Also localAudioCodec and remoteAudioCodec are both PCMA now. I send my login data and the testing setup to support@flashphoner.com . Feel free to charge me something for your effort.Best regards
Marc
Good day, Marc.
We have reproduced the problem using softphone and found solution. You should change one string in your PHP script from
to
and Asterisk sound injection will work, playback is hearable both in softphone and browser. We have changed it for you.
In this case, Asterisk 1234567890.call file looks such as
where 1234567890 is callee number (and conference number too).
We have reproduced the problem using softphone and found solution. You should change one string in your PHP script from
Code:
exec('echo "Set: CONF='.$output.'" >> /tempcids/'.$cidf);
Code:
exec('echo "Set: CONF='.$cid.'" >> /tempcids/'.$cidf);In this case, Asterisk 1234567890.call file looks such as
Code:
Channel: Local/s@inject-sound
Application: Playback
Data: /path/to/sound/file
Archive: Yes
Set: CONF=1234567890OH MY GOD!! THAT IS WORKING!!! I'm real happy now! Thank you very much for your investigations, Max! Can I pay a fee for your work?
I cannot belive that I just mixed up two variables in my own php script. Oh boy...
Best regards
Marc
P.S. And also the Firefox works now after founding again the tweak you posted earlier
And Edge as well. Seems that I can start now building my project!
I cannot belive that I just mixed up two variables in my own php script. Oh boy...
Best regards
Marc
P.S. And also the Firefox works now after founding again the tweak you posted earlier
Code:
if(Browser.isFirefox()) {
var audioContext = new AudioContext();
var emptyAudioStream = audioContext.createMediaStreamDestination().stream;
constraints.customStream = emptyAudioStream;
}
Last edited:
Hello
Glad it works for you. It would be enough to purchase a monthly subscription from here https://flashphoner.com/simple-license