421
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422
Other Yate server issues / Re: The clones
« on: April 10, 2014, 03:39:04 AM »
A yate sip listener binds on an IP address.
Which interface is used is up to the OS, you can't make this decision in yate.
What is the reason you are using multiple yate instances?
Which interface is used is up to the OS, you can't make this decision in yate.
What is the reason you are using multiple yate instances?
423
Other Yate server issues / Re: The clones
« on: April 10, 2014, 02:48:28 AM »
I'm afraid I don't understand.
I don't understand the separation between address and interface.
Is this a single machine with multiple interfaces?
Y0, Y1 ... are multiple yate instances in the same machine?
I don't understand the separation between address and interface.
Is this a single machine with multiple interfaces?
Y0, Y1 ... are multiple yate instances in the same machine?
424
Other Yate server issues / Re: The clones
« on: April 10, 2014, 01:02:35 AM »
Indicate the account for the outgoing call when routing.
Here is a regexroute example:
^1\(.*\)$=line/\0;line=user1
This will make the call using the user1 transport (listener)
Here is a regexroute example:
^1\(.*\)$=line/\0;line=user1
This will make the call using the user1 transport (listener)
425
Other Yate server issues / Re: The clones
« on: April 09, 2014, 08:18:49 AM »
As written in previous post user1 will register using udp1 listener, user1 will register using the udp2 listener.
Where is the uncertainty?
Where is the uncertainty?
426
Other Yate server issues / Re: The clones
« on: April 09, 2014, 07:47:28 AM »
Setup listeners in ysipchan.conf:
[listener udp1]
type=udp
addr=10.0.0.1
[listener udp2]
type=udp
addr=10.0.0.2
Setup the local ip/port in in account parameters:
Here is an accfile.conf example:
[user1]
ip_transport_localip=10.0.0.1
ip_transport_localport=5060
[user2]
ip_transport_localip=10.0.0.2
ip_transport_localport=5060
user1 will register using the udp1 listener, user2 will register using the udp2 listener.
Set the line parameter when routing a call on specific account.
See http://docs.yate.ro/wiki/Ysipchan for more info.
[listener udp1]
type=udp
addr=10.0.0.1
[listener udp2]
type=udp
addr=10.0.0.2
Setup the local ip/port in in account parameters:
Here is an accfile.conf example:
[user1]
ip_transport_localip=10.0.0.1
ip_transport_localport=5060
[user2]
ip_transport_localip=10.0.0.2
ip_transport_localport=5060
user1 will register using the udp1 listener, user2 will register using the udp2 listener.
Set the line parameter when routing a call on specific account.
See http://docs.yate.ro/wiki/Ysipchan for more info.
427
Other Yate server issues / Re: The clones
« on: April 09, 2014, 05:52:42 AM »
Hi,
You should give us more info.
Some usage case scenario would help us to help you.
You should give us more info.
Some usage case scenario would help us to help you.
428
Other Yate server issues / Re: Routing call based on SIP message
« on: March 31, 2014, 01:20:47 AM »
Hi,
See http://docs.yate.ro/wiki/Call_Forker on how to fork a call.
Here is an example of playing announcement:
^123$=fork;fork.stop=noanswer^;callto.1=sip/sip:123@127.0.0.1;callto.2=|;callto.3=wave/play//myfile.au
The above rule will call on sip to 123@127.0.0.1. If the call fails fork will stop if reason is not 'noanaswer' (480). If the call fails with 480 it will call to wavechan module to play the file myfile.au on incoming call leg.
See http://docs.yate.ro/wiki/Call_Forker on how to fork a call.
Here is an example of playing announcement:
^123$=fork;fork.stop=noanswer^;callto.1=sip/sip:123@127.0.0.1;callto.2=|;callto.3=wave/play//myfile.au
The above rule will call on sip to 123@127.0.0.1. If the call fails fork will stop if reason is not 'noanaswer' (480). If the call fails with 480 it will call to wavechan module to play the file myfile.au on incoming call leg.
429
Other Yate server issues / Re: CallerID as my UserName of the Trunk
« on: March 31, 2014, 12:46:27 AM »
Hi,
Seems the calling party number is TRK00519-001.
I can't say more without sip debug.
Start yate with more debug and timestamps. Command line options:
-vvvvv -Dt
Make sure sip messages are displayed:
ysipchan.conf:
[general]
printmsg=yes
Seems the calling party number is TRK00519-001.
I can't say more without sip debug.
Start yate with more debug and timestamps. Command line options:
-vvvvv -Dt
Make sure sip messages are displayed:
ysipchan.conf:
[general]
printmsg=yes
430
Other Yate server issues / Re: CallerID as my UserName of the Trunk
« on: March 28, 2014, 01:40:31 AM »
Hi,
Can you give us more info?
Like the protocol you are using, config, log ...
Can you give us more info?
Like the protocol you are using, config, log ...
431
Other Yate server issues / Re: How can RTP transport directly from client to client?
« on: March 24, 2014, 02:42:25 AM »
You might not edit the right file on Windows.
See http://docs.yate.ro/wiki/Debugging_Yate_Client_on_Windows
See Introduction section.
See http://docs.yate.ro/wiki/Debugging_Yate_Client_on_Windows
See Introduction section.
432
Linux / Re: the client cannot make sound in centos
« on: March 24, 2014, 02:08:39 AM »
Try calling locally the tonegen module: type tone/ring in call destination area and press 'Call'. You should hear a ring tone.
If you have a server between you and the party you are talking to try calling it directly.
Try using oss (most probably won't help, but try it anyway)
For the next log(s) please start yate with -Dt to see the time a message is logged.
And forget about the QT sounds not available warning, it has nothing to do with call media.
If you have a server between you and the party you are talking to try calling it directly.
Try using oss (most probably won't help, but try it anyway)
For the next log(s) please start yate with -Dt to see the time a message is logged.
And forget about the QT sounds not available warning, it has nothing to do with call media.
433
Linux / Re: the client cannot make sound in centos
« on: March 20, 2014, 02:10:28 AM »
The alsa module wasn't built or is not loaded.
The default sound module is alsachan.
Install alsa devel package, run configure again and re-build yate.
You can use the osschan module. Change yate-qt4.conf:
[client]
device=oss//dev/dsp
The default sound module is alsachan.
Install alsa devel package, run configure again and re-build yate.
You can use the osschan module. Change yate-qt4.conf:
[client]
device=oss//dev/dsp
434
Other Yate server issues / Re: How can RTP transport directly from client to client?
« on: March 10, 2014, 06:41:30 AM »
I can't help you without yate log.
RTP forward is working. If you say is not working I need to see the log to see what happens.
RTP forward is working. If you say is not working I need to see the log to see what happens.
435
Other Yate server issues / Re: How can RTP transport directly from client to client?
« on: March 10, 2014, 06:08:19 AM »
If there is no rtp_forward=possible in call.route it means the caller didn't offered media.
Can you dump the call.route message?
And the received INVITE?
Better, can you dump a log with message sniffer and sip messages?
Can you dump the call.route message?
And the received INVITE?
Better, can you dump a log with message sniffer and sip messages?