331
Other Yate server issues / Re: again timeot
« on: February 23, 2015, 01:47:21 AM »
The timeout and maxcall parameters should be handled by the outgoing channels (datacard or sip).
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332
Installation from packets/SVN/.exe/AppStore / Re: Ubuntu 14.10 packages
« on: February 19, 2015, 01:34:34 AM »
There might be an issue related to ringer.
Try to disable the ringer: Settings -> Options -> Telephony page -> Ringer option
Try to disable the ringer: Settings -> Options -> Telephony page -> Ringer option
333
Other Yate server issues / Re: YATE setup according to OSTN standard hangs up
« on: January 20, 2015, 03:07:05 AM »
Your issue may be the same as the ones listed below:
See: http://forum.yate.ro/index.php?topic=323
See also the thread at:
http://yate.null.ro/archive/?action=show_msg&actionargs%5B%5D=90&actionargs%5B%5D=90
You should post a yate log to make sure.
See: http://forum.yate.ro/index.php?topic=323
See also the thread at:
http://yate.null.ro/archive/?action=show_msg&actionargs%5B%5D=90&actionargs%5B%5D=90
You should post a yate log to make sure.
334
Windows / Re: Build Yate solution on Windows
« on: January 16, 2015, 02:32:16 AM »
Hi,
The AMR codec is covered by patents.
The yate module only implements library usage (a data translator).
For external dependencies you need:
- Devel files (include headers): the compiler need them
- Symbol libraries (the 'lib' files): the linker need them
- Runtime modules (the 'dll' files): needed at runtime
Some info here:
http://docs.yate.ro/wiki/Compiling_and_Installing_Yate_on_Windows#Compiling_Yate_on_Windows
Here are some other third party libraries (not listed there) you'll need:
- zlib: for zlibcompress module
- openssl: for openssl module
- speex: for speexcodec module
You may ignore some modules if you don't need them.
E.g. if you can't build the zlibcompress module you won't have support for compression (it is used by jabber modules to compress stream data if compression is available).
The AMR codec is covered by patents.
The yate module only implements library usage (a data translator).
For external dependencies you need:
- Devel files (include headers): the compiler need them
- Symbol libraries (the 'lib' files): the linker need them
- Runtime modules (the 'dll' files): needed at runtime
Some info here:
http://docs.yate.ro/wiki/Compiling_and_Installing_Yate_on_Windows#Compiling_Yate_on_Windows
Here are some other third party libraries (not listed there) you'll need:
- zlib: for zlibcompress module
- openssl: for openssl module
- speex: for speexcodec module
You may ignore some modules if you don't need them.
E.g. if you can't build the zlibcompress module you won't have support for compression (it is used by jabber modules to compress stream data if compression is available).
335
Yate users hangout place / Re: Can't get EventLog files on Windows platform to work
« on: January 14, 2015, 02:06:42 AM »
Seems the eventlogs module is not built and packet for Windows.
You should add a post to Feature requests section.
You should add a post to Feature requests section.
336
Other Yate server issues / Re: CallerID setup
« on: January 07, 2015, 01:22:15 AM »
You can put in the domain parameter the value you want to be set in the domain part of the SIP 'From' header URI.
The string your are setting there is what you want to be.
If you are not setting it yate will build the 'From' header from transport local ip address.
The string your are setting there is what you want to be.
If you are not setting it yate will build the 'From' header from transport local ip address.
337
Other Yate server issues / Re: CallerID setup
« on: January 06, 2015, 02:16:05 AM »
Edit regexroute.conf.
Change the following:
[extra]
call.execute=30
[call.execute]
${domain}^$=;domain=YOUR_DOMAIN_NAME
NOTE: This will affect all outgoing calls. I suppose for your scenario it will do the job.
Change the following:
[extra]
call.execute=30
[call.execute]
${domain}^$=;domain=YOUR_DOMAIN_NAME
NOTE: This will affect all outgoing calls. I suppose for your scenario it will do the job.
338
Yate bugs / Re: "wave/record" do not record call from yate-qt4 with "inactive call"
« on: January 05, 2015, 02:34:58 AM »
How do you record the call in yate-qt4?
There is no such option in the user interface.
There is no such option in the user interface.
339
Yate bugs / Re: "wave/record" do not record call from yate-qt4 with "inactive call"
« on: December 23, 2014, 09:30:39 AM »
Hi,
Can you describe what are you doing?
And what exactly is the issue?
Can you describe what are you doing?
And what exactly is the issue?
340
Other Yate server issues / Re: ptime in SDP - Need 20ms, getting 30 ms
« on: December 23, 2014, 02:08:51 AM »
Disable ilbc30:
ysipchan.conf:
[codecs]
ilbc30=no
Keep in mind this will disable it for all calls.
ysipchan.conf:
[codecs]
ilbc30=no
Keep in mind this will disable it for all calls.
341
Yate users hangout place / Re: Make Call, Play 2 Audio files and Disconnect ( Hello World script )
« on: December 16, 2014, 02:35:28 AM »
call.execute is used to call from a channel.
You don't have one in your script.
Some modules (like dumb, tonegen) have the ability to build a channel and re-send the call.execute message.
You must add a 'target' (containing the called party number) parameter to your message.
If you do it the dumb module will send a call.route message and then a call.execute on behalf of the newly created dumb channel.
If you add a 'direct' parameter the dumb module will send a call.execute (to indicated 'direct' will be set in 'callto' parameter) without routing the call.
On return your call.execute will contain the 'id' of the dumb channel and 'peerid' its peer channel id.
You may then use these to track the call.
Don't forget to track call termination.
You may use call.cdr with operation=finalize: check the 'chan' parameter
You may also track chan.disconnected for either channel (chan.disconnected is sent by the disconnected channel, i.e. for a call only 1 channel will send it).
When you attach your sources (play) instruct wavefile to notify on termination and do your stuff when a source terminates.
See http://docs.yate.ro/wiki/Chan.attach and http://docs.yate.ro/wiki/Chan.notify
When you are done just enqueue a call.drop message with 'id' set to one of the tracked channel's id.
You don't have one in your script.
Some modules (like dumb, tonegen) have the ability to build a channel and re-send the call.execute message.
You must add a 'target' (containing the called party number) parameter to your message.
If you do it the dumb module will send a call.route message and then a call.execute on behalf of the newly created dumb channel.
If you add a 'direct' parameter the dumb module will send a call.execute (to indicated 'direct' will be set in 'callto' parameter) without routing the call.
On return your call.execute will contain the 'id' of the dumb channel and 'peerid' its peer channel id.
You may then use these to track the call.
Don't forget to track call termination.
You may use call.cdr with operation=finalize: check the 'chan' parameter
You may also track chan.disconnected for either channel (chan.disconnected is sent by the disconnected channel, i.e. for a call only 1 channel will send it).
When you attach your sources (play) instruct wavefile to notify on termination and do your stuff when a source terminates.
See http://docs.yate.ro/wiki/Chan.attach and http://docs.yate.ro/wiki/Chan.notify
When you are done just enqueue a call.drop message with 'id' set to one of the tracked channel's id.
342
Installation from packets/SVN/.exe/AppStore / Re: <ybts:CONF> Missing radio band or C0 configuration
« on: December 10, 2014, 01:44:30 AM »
512..885 is an interval.
As the comment says the parameter is an integer value.
You must set a value within given interval!
As the comment says the parameter is an integer value.
You must set a value within given interval!
343
Yate users hangout place / Re: simple sip phone dial
« on: December 10, 2014, 01:42:09 AM »
Sorry, bad regexp (| char must be escaped), set stoperror:
$ev->params["stoperror"] = "noanswer\|busy^";
Also replace 'stoperror' (obsolete) with new 'fork.stop':
$ev->params["fork.stop"] = "noanswer\|busy^";
$ev->params["stoperror"] = "noanswer\|busy^";
Also replace 'stoperror' (obsolete) with new 'fork.stop':
$ev->params["fork.stop"] = "noanswer\|busy^";
344
Yate users hangout place / Re: Not working provider ringback tone
« on: December 09, 2014, 04:58:28 AM »
183 comes after 180 indeed.
set multi_ringing=yes in ysipchan.conf [general] section.
Make sure you don't add it with comment char before it (no ; char).
If still not working please attach the log again.
Try to start yate with timestamps (-Dt should be fine).
set multi_ringing=yes in ysipchan.conf [general] section.
Make sure you don't add it with comment char before it (no ; char).
If still not working please attach the log again.
Try to start yate with timestamps (-Dt should be fine).
345
Yate users hangout place / Re: Not working provider ringback tone
« on: December 09, 2014, 03:29:43 AM »
Hi,
A yate log with sip signalling and message sniffer would help.
A yate log with sip signalling and message sniffer would help.