Messages

406

Yate bugs / Re: iLBC negotiation

« on: June 05, 2014, 02:03:50 AM »

Can you post the SDP offer and response?
Note that in yate mode 20 and 30 are different formats.
If yate advertise mode 20 and receives mode 30 the received format will be ignored (not enabled).

407

Linux / Re: how to modify the yate client's ui

« on: June 04, 2014, 12:46:05 AM »

QComboBox selection changed (the activated() signal) is automatically signaled by QtWindow (see QtWindow::doInit() in qt4client.cpp).
The handler for it can be implemented in any ClientLogic descendent. See ClientLogic::select(), the handler for UI selection change notifications (Ref. yatecbase.h, engine/ClientLogic.cpp)

Tell me if you need more info.

408

SIP to H.323 proxy / Re: Yate sending an incorrect address to PBX

« on: May 30, 2014, 06:04:38 AM »

You did received the H323 call. So TCP works.


Check your setup.
The call is accepted by H323.
It is rejected with reason 'offline'.
The log is showing you are trying to route the incoming call on 'lync' sip line which is offline.
A sip line is an account you are registering to a server (yate is the client).
The log shows nothing about any sip line registering to another server.

409

SIP to H.323 proxy / Re: Yate sending an incorrect address to PBX

« on: May 29, 2014, 07:55:57 AM »

I'm afraid you lost me.
Is the sip line connected or not?
Can you post again the config files, including the accfile.conf (I suppose you are using it to login on sip)?
Can you post a wireshark capture to see the sip account login and a failed call?
Post yate log also.

410

SIP to H.323 proxy / Re: Yate sending an incorrect address to PBX

« on: May 29, 2014, 04:40:45 AM »

Can you be more clear?
What do you mean by rule not working? Does it match? Is the line connected?

I don't understand the other phrase.
Which h323 module tries to connect to which PBX?

411

Other Yate server issues / Re: Jabber server. Why Yate treats the message as destined external domain?

« on: May 19, 2014, 05:00:46 AM »

The jingle module will add a server item for domain 'mydomain.com' to the jabber server.
This is the reason the jabber server puts the externaldomain parameter in call.route.
There is no check for modules registering server items with the same domain(s) as the jabber server.

412

Yate users hangout place / Re: yate basic configuration

« on: May 16, 2014, 03:16:33 AM »

Config seems ok.
Sip logs are in yate logs (they are displayed at debug level 9).
Reload can be done in telnet console, reload command (see http://docs.yate.ro/wiki/Rmanager).
Yate logs are in var/log/yate (or in other location if you start yate with -l).

413

Yate users hangout place / Re: yate basic configuration

« on: May 16, 2014, 01:22:52 AM »

What do you mean by see online on conf file?
If you want to route calls only if the account is online you should use javascript.
When you login an account the module will send an user.notify message with its status.
In javascript you can store account status and route if online.

Some configuration options are applied on reload (no restart needed).
Just ask what you need to know.
For example regexroute rules are always replaced on reload.

To enable rtp forward see http://docs.yate.ro/wiki/RTP_Forwarding

If you see the account logon success message and calls are failing post a yate log to see what happens.

414

SIP to H.323 proxy / Re: Yate sending an incorrect address to PBX

« on: May 15, 2014, 05:59:29 AM »

The log shows a call to a sip line (lync) who is not registered.
There is no such line.

Also the correct regexroute target to route is:

rule=sip/01032;line=lync

415

SIP to H.323 proxy / Re: Yate sending an incorrect address to PBX

« on: May 15, 2014, 04:00:40 AM »

So you have an H323 issue.
I can't see the problem: the failed H323 call is not even routed.

I'm not sure I can help you with h323, but:

You should post the h323chan.conf file along with a wireshark capture and yate log.
Set debug=10 in h323chan.conf, 'general' section. This will put a lot of debug from H323 library.

416

SIP to H.323 proxy / Re: Yate sending an incorrect address to PBX

« on: May 15, 2014, 12:56:00 AM »

Yes, we need the log (configuration would help also).

Start yate with -Dt to see the debug message time.
Also start it with max debug level: -vvvvv
Message sniffer would help also (see http://docs.yate.ro/wiki/Debugging_and,_or_Investigation_of_messages)

417

Windows / Re: Yate client call transfer

« on: May 05, 2014, 06:22:22 AM »

You can make a new call and link audio:

Push the 'show call transfer' button in call entry (the one with 2 phones on it, if you move the mouse over it a tooltip will be shown).
This will show an edit box where you can type the target for the new call.
Push the button on the right of it to make the call.
Once the new call is answered its audio data will be linked with the incoming call.

After that:
- You can hangup the new call and continue the initial call
- If the initial call is hungup the new call will be terminated also

418

Other Yate server issues / Re: cdrbuild warning in log

« on: April 17, 2014, 12:47:49 AM »

Someone forgot them there.
They are harmless.
It will be fixed.

419

Other Yate server issues / Re: warning messages in log but Yate working fine

« on: April 15, 2014, 01:19:13 AM »

This is a warning put by rtp module if rtp is late.
This may happen if remote party is late in sending rtp data.
It may signal media is missing (sometimes media is not sent at all).

You may disable the warning in yrtpchan.conf:

[timeouts]
warnfirst=no

Or you may set a longer interval to warn (set it to 0 to disable):
[timeouts]
timeout=10000

See the yrtpchan.conf.sample file for comments.

420

Other Yate server issues / Re: Register on demand

« on: April 14, 2014, 01:05:20 AM »

If you don't want to register a line but you need credentials when calling you can:
1. Set the line in accfile.conf (username and password) but don't set the registrar
2. When calling set the sip target and the line

accfile.conf:
[user]
protocol=sip
username=my_user
password=my_password

regexroute.conf:
[default]
match_rule=sip/sip:number@ip.address;line=user

If challenged, sip will use 'user' line credentials to authenticate.
This way you don't need to register an account to server to use its credentials.