JR Richardson writes:
Well, you sure are a smart fella. I reverted the patch and reloading
the address table works as expected now. For clarification I was
testing with kamailio-3.0.2_src_2010-06-30_a6141a.tar.gz snapshot.
Let me know when you have the updated fix and I'll do sime testing.
Juha Heinanen writes:
reverting the patch thus most likely reintroduced this other bug.
perhaps the bug that you saw has something to do with the last sentence
of the commit.
jr,
did you try with db_mode=1? it works ok here.
-- juha
___
SIP
Use Asterisk or FreeSwitch for this fronted by SER/Kamailio.
FreeSwitch FIFO application or Asterisk Queues + AGI.
Neill;o)
Aeonvista Ltd
Opening Up New Ideas
On 1 July 2010 05:06, Trung Anh dangtrung...@gmail.com wrote:
Hi everyone,
I have a problem about Agents status.
Now, i want
Daniel-Constantin Mierla writes:
I guess you use this command via rpc interface, not via mi.
yes, i use 'ctl mi address_reload', noticed that it didn't work and then
went and fixed it.
Looking at
the code, the mi init child is not done for rpc processes, resulting in
some mi commands not
Hello,
keep cc-ing the mailing list for any message, so next people that come
and scan mailing list archive for the same issue discover the solution
and whether was good or not.
Cheers,
Daniel
On 6/30/10 11:40 PM, Nicolas Rüger wrote:
Hello Daniel,
thanks for the fast answer. I followed
Hello,
On 5/27/10 3:32 PM, Anton Roman wrote:
Hi,
regarding the missing Contact header in the REFER message, it can be
solved by including some lines in /modules_k/dialog/dlg_ transfer.c.
In this file, the Contact of the initial INVITE generated with
dlg_bridge command can be easily changed
Daniel-Constantin Mierla writes:
Juha, I just committed on master and tested over MI where works fine.
Can you do some tests over rpc? If all ok then can be backported.
it would be nice if i could, but after trying to cherry-pick my msilo
commit to sr_3.0, i got errors about debian-lenny
Hi, I have a similar issue, my proxy sip doesn't relay the ACK coming from a
Cisco GW (caller), to a PSTN-GW (callee), then the OK and ACK are retransmitted
for a while till the pstn-gw send a BYE.
Attach a text graph, where:
10.10.10.128 is a Cisco GW.
10.15.15.122 is my Proxy SIP (and oldie
On 7/1/10 1:18 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
Juha, I just committed on master and tested over MI where works fine.
Can you do some tests over rpc? If all ok then can be backported.
daniel,
i deleted whole sr git repo and downloaded a new one. after
-Original Message-
From: Juha Heinanen [mailto:j...@tutpro.com]
Sent: Thursday, July 01, 2010 1:43 AM
To: JR Richardson; SR-Users
Subject: Re: [SR-Users] permissions module address_reload not working
Juha Heinanen writes:
reverting the patch thus most likely reintroduced this
On Tue, Jun 29, 2010 at 12:24:03PM +0200, Henning Westerholt wrote:
On Friday 18 June 2010, Iñaki Baz Castillo wrote:
2010/6/18 Iñaki Baz Castillo i...@aliax.net:
2010/6/17 Klaus Darilion klaus.mailingli...@pernau.at:
looks like the header is buggy, it should be Digest:
challenge
Sehr geehrte Damen und Herren,
leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich
mich bis einschliesslich 12.07.2010 im Urlaub befinde.
In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter
unter der Rufnummer 04231 / 776 .
Mit freundlichen
Daniel-Constantin Mierla writes:
this is another issue ... the mi commands were designed to be run from
special processes, created by mi transport modules (mi_fifo,
mi_datagram, etc). There are special init functions called for these
processes. However, with 3.0 xmlrpc module can call
Am 01.07.2010 15:14, schrieb Ján ONDREJ (SAL):
But I con't find any response in ACK packet send to our provider:
The ACK is just to finalize the rejected INVITE. Kamailio should send
another INVITE request with proper credentials.
regards
Klaus
On Thursday 01 July 2010, Ján ONDREJ (SAL) wrote:
ok, i'll change it in the code.
Thank you.
Hello Jan,
Even if it's better, it still don't work for me.
Here is what is logged in log:
0(1515) DEBUG: uac [auth_hdr.c:409]: hdr is Authorization: Digest
username=07500,
Am 01.07.2010 14:38, schrieb Claudio Furrer:
Hi, I have a similar issue,
It is not possible to debug this issue without full SIP trace!
ngrep -Wbyline -t -d any port 5060
regards
Klaus
Sorry, here the trace (gziped).
Regards,
Caio
call_ack_fail1.txt.gz
Description: Binary data
On 7/1/10 4:39 PM, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
this is another issue ... the mi commands were designed to be run from
special processes, created by mi transport modules (mi_fifo,
mi_datagram, etc). There are special init functions called for these
processes.
Sehr geehrte Damen und Herren,
leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich
mich bis einschliesslich 19.07.2010 im Urlaub befinde.
In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter
unter der Rufnummer 04231 / 776 .
Mit freundlichen
Daniel-Constantin Mierla writes:
No time to do real tests, if you can run some and report back, it would
be great. Once some testing is done, can be ported to 3.0.
daniel,
i tested address_reload, trusted_reload, and sht_reload mi commands over
xmlrpc and they all worked fine. also 'ctl mi
Hello,
The guy operating the Asterisk at the far end have been inspecting the traces
I've sent him. Hen notes that the ACK relayed doesn't entirely comply to the
standard. The values of branch in the VIA header of the relayed ACK
response incorrectly have the value 0 (also rport in the VIA
Hi,
voice:/# ps auxf |grep rtpproxy |grep -v grep
rtpproxy 1291 0.0 0.0 26800 876 ?Ssl Jun18 0:10
/usr/sbin/rtpproxy -u rtpproxy rtpproxy -l my.public.ip.here -s
udp:localhost 7722
voice:/#
kamailio.cfg:
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_ACCDB
#!define
On 1 July 2010 23:03, Dmitri Korotkov dmitri.korot...@festart.ee wrote:
Hi,
default kamailio config file(its routing part) already has rtpproxy support
in case if WITH_NAT is defined.
And there is no problems when NATed subscribers calls one to other...
I have problem only with PSTN and
Hello,
I have setup an application level gateway that bridges my two networks.
I have a situation where an INVITE will cross the gateway ( thus
rewriting the SDP ) go through a proxy and come back across the gateway.
At this point, it would be wasteful to use the RTP Proxy as the audio
will
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