CallWeaver - Tickets

Ticket #526 (Open) reported by liugang - sip trunk over TCP

Thu, 06 Jan 2011 01:46:51 GMT

Dear sir: my CW is 1.2.1,download from http://devs.callweaver.org/release/callweaver-1.2.1.tgz,and compile with—enable-sip-tcp-support,success. but how can setup a sip trunk over tcp with other sip pbx. asterisk 1.6 support sip tcp trunk with tcpenabled=true in sip.conf.

I need a help,thans all

Ticket #428 (Open) changed by Konstantinos Arvanitis - MuxMon produce shifted audio

Wed, 05 Jan 2011 12:28:36 GMT

There are memory leaks with the proposed patch. The attached patch fixes those leaks, but I don’t have the setup or the time to test if it fixes the actual shift.

Please test and report.

Ticket #525 (Open) reported by Carlos Galveias - ICD billsec=0

Mon, 20 Dec 2010 18:12:22 GMT

Hi, Im doing an application to use AMI to make autodial and send to an icd queue. Everything works fine except that the billsec is allways 0.

I use a class to send the originate to callweaver but i does:

Action: Originate channel: Local/$number application=ICDCustomer queue=$campaign

On the cdr i have this:

acctid: 2342749 calldate: 2010-12-20 17:40:59+00 clid: “teste name” <...

Ticket #428 (Open) changed by Alexey - MuxMon produce shifted audio

Sun, 19 Dec 2010 20:12:08 GMT

I have some problem on 1.2.1. After applying the patch (regen_frame.patch), Callweaver starts, but on calls dies with pid 75644 (callweaver), uid 444: exited on signal 11 in dmesg.

Ticket #524 (Open) changed by Carlos Galveias - srtp issue (and others)

Wed, 13 Oct 2010 21:06:13 GMT

I managed to install the RC version , i had to reinstall libpri. Anyway i can confirm the same behaviour.

SRTP enabled phone dials out to a VoIP provider or ZAP channel and while call is in progress (Establishing, ringing, etc) i just hear white noise. After the call gets answered the audio is ok.

Regards

Carlos

Ticket #523 (WorksForMe) changed by Zoran Pericic - Disconnect after 30sec with re-INVITE and t38udptsupport=yes

Wed, 13 Oct 2010 19:29:33 GMT

Problem was at t38_status which was setted to T38_NEGOTIATED before any negotiation.

Don’t know much about SIP protocol, but I think this should be setted after successful negotiation.

Could someone test this with T38 UDPTL? It work for normal calls.

Ticket #524 (Open) reported by Carlos Galveias - srtp issue (and others)

Wed, 13 Oct 2010 16:41:40 GMT

Hi all,

Ive been trying TCP/TLS and SRTP and since ive been using a older trunk version (ive made some changes manually (like vale, etc, but its still a opbx_ and not a cw_). The reason is that at some point it lost stabillity and that version i have is in production and makes 1.000.000 calls (750 calls with audio in a queue for 2,15m , 5 new calls per sec).

Anyway, everything is worki…

Ticket #523 (WorksForMe) changed by Zoran Pericic - Disconnect after 30sec with re-INVITE and t38udptsupport=yes

Sat, 09 Oct 2010 16:24:09 GMT

Skip last comment, I haven’t reload sip peer info.

Working – both clients canreinvite=yes, t38udptlsupport=no – both clients canreinvite=no, t38udptlsupport=yes – one clients canreinvite=no other canreinvite=yes, t38udptlsupport=yes

Not working – both clients canreinvite=yes, t38udptlsupport=yes

Ticket #523 (WorksForMe) changed by Zoran Pericic - Disconnect after 30sec with re-INVITE and t38udptsupport=yes

Sat, 09 Oct 2010 15:08:53 GMT

Another related bug, I disabled t38udptlsupport and set one side canreinvite=no but CW again tries to establish RTP bridge.

This time I use ekiga and mediatrix, which don’t support re-INVITE or it just seems.

If needed could make tcpdump.

Ticket #523 (WorksForMe) changed by Zoran Pericic - Disconnect after 30sec with re-INVITE and t38udptsupport=yes

Sat, 09 Oct 2010 14:08:28 GMT

Part of extensions.conf