The more interesting question is why the reinvite is being rejected:
Status-Line: SIP/2.0 400 Bad Request
Message Header
Via: SIP/2.0/UDP
[PUBLIC-MITEL]:5060;received=[PUBLIC-MITEL];branch=z9hG4bK-d8754z-2648756d6bb46b65-1---d8754z-;rport=5060
From: ;tag=0_1970073968-102
On 08/13/2015 03:40 PM, Fred Posner wrote:
On 08/13/2015 03:30 PM, Alex Balashov wrote:
I see a dialog established followed by a reinvite. Which frame # do you
take to be the "on-hold invite"?
I was taking frame 8 as a new invite. I'll see what I'm doing wrong.
No, it's a reinvite. The dis
On 08/13/2015 03:30 PM, Alex Balashov wrote:
> I see a dialog established followed by a reinvite. Which frame # do you
> take to be the "on-hold invite"?
>
I was taking frame 8 as a new invite. I'll see what I'm doing wrong.
--fred
___
SIP Express Ro
I see a dialog established followed by a reinvite. Which frame # do you
take to be the "on-hold invite"?
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: htt
On 08/13/2015 02:42 PM, Alex Balashov wrote:
> On 08/13/2015 02:34 PM, Fred Posner wrote:
>
>> Sadly, no. Straight up INVITE.
>
> Wait, what? That doesn't make any sense. Can you provide a full
> signalling capture?
>
Here's an example exported from the pcap. This example has topology
hiding en
On 08/13/2015 02:34 PM, Fred Posner wrote:
Sadly, no. Straight up INVITE.
Wait, what? That doesn't make any sense. Can you provide a full
signalling capture?
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-80
On 08/13/2015 02:30 PM, Alex Balashov wrote:
> I assume these are reinvites? If so, how would the strategy for
> handling them deviate from handling of all other in-dialog requests?
>
> -- Alex Balashov | Principal | Evariste Systems LLC
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direc
I assume these are reinvites? If so, how would the strategy for handling them
deviate from handling of all other in-dialog requests?
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678
Hello all,
Always fun integrating with proprietary pbx's... is there a way that is
"generally accepted" as handling hold invites from devices such as Mitel
or Toshiba?
We're not proxying the media, so these devices are trying to send an
Invite to the endpoint when calls are placed on hold.
Fred
Hi Guys,
Trying to use cnxcc but cnxcc.so is not in the modules folder. What is the
easiest way to install?
Thanks
Keith
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.or
Hello Sammy,
Thanks for the detailed explanation.
Btw, awesome blog! There are a lot of useful tutorials!
Keep up the good work ☺
Regards,
Grant
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
SamyGo
Sent: Thursday, August 13, 2015 4:47 PM
To: Kamailio (SER) - Users
Good to know that your issue is resolved. I see you're applying only one
rtpproxy command, possibly your's only need incoming calls from carrier.
Also in BYE you need to use unforce_rtpproxy function to release ports.
Same solution can have many solutions; I'll share how mine works, for the
sake o
Am 13.08.2015 um 12:58 schrieb Daniel-Constantin Mierla:
> Hello,
>
> maybe the coredump got corrupted. Was it opened with other applications?
No, it's modification date hasn't changed (and there are two core dumps).
>
> Can you reproduce the issue?
I'll be working on it.
> Any error log mes
It’s working!
Thank you very much!!
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Waite, Hugh
Sent: Thursday, August 13, 2015 4:02 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] need help with RTPProxy in bridged mode
Hi,
If the media is coming f
Hi,
If the media is coming from a different IP address than the signalling, you may
need to use the ‘r’ flag to force the address in the SDP to be trusted.
You should also use the same flags in the same order, in the on-reply route.
(This is all in
http://kamailio.org/docs/modules/4.3.x/modules/
I enabled DBUG logging when starting rtpproxy:
The logs output:
INFO:handle_delete:369832-3648463033-238704: forcefully deleting session 1 on
ports 57012/35102
INFO:remove_session:369832-3648463033-238704: RTP stats: 0 in from callee, 0 in
from caller, 0 relayed, 0 dropped, 464 ignored
INFO:rem
Hello,
Yeah, I also noticed I forgot the / . Now the SDP c parameter is set correctly,
but the audio from private to public isn’t relayed by rtpproxy.
I ran a tcp dump on both interfaces (private and public), and it showed me RTP
is being received from Freeswitch and also from our carrier, but
Hi,
Try starting rtpprpxy with a / in between the two IP addresses.
For example -l 1.1.1.1/2.2.2.2
Besides that it depends where you are placing your rtpproxy function.
BR,
Sammy
On Aug 13, 2015 8:36 AM, "Grant Bagdasarian" wrote:
> Hello,
>
>
>
> I’m using RTPproxy for the first time in bridged
Hello,
I'm using RTPproxy for the first time in bridged mode and I can't get
kamailio/rtpproxy to rewrite the c parameter to the correct public ip address
of kamailio.
The setup is as following:
Carrier --[fiber]-- Kamailio -[lan]- Freeswitch
Kamailio is listening on t
Hi,
Daniel thanks for you assistance thus far.
So I uncomment the two lines... Kamailio now starts.
If I add in the 'enable_sctp = 1' parameter into the kamailio.cfg file, and
attempt to start Kamailio again, I get the same errors again:
# systemctl status kamailio.service -l
kamailio.serv
Hello,
>From which version has this been generated?
Also, can you provide the output of "kamctl ul show" and the relevant
modparam sections of your config (usrloc, dmq, dmq_usrloc).
Cheers,
Charles
On 13 August 2015 at 10:23, Kelvin Chua wrote:
> I don't know if this is related, this happens
Hello;
i dont wanna open new thread as spamming. 1 of them build error for new
version. other one is about revert_uri() function on upgrade. maybe, i wrote
wrong subject.
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/upgrade-V4-2-5-to-V3-1-Build-error-tp140428p
Is this related to the upgrade issue?
Looks like a reply to a different thread.
Cheers,
Daniel
On 13/08/15 10:59, ycaner wrote:
> i found a changes for rever_uri() function . i am using carrierroute module
> to route changes. so after a query destination number , a prefix is added to
> Request u
Hello,
maybe the coredump got corrupted. Was it opened with other applications?
Can you reproduce the issue? Any error log messages before the crash?
Cheers,
Daniel
On 13/08/15 11:57, Sven Neuhaus wrote:
> Hi,
>
> we're experiencing crashes in the siptrace.so module with Kamailio
> 4.3.1. I loo
Hi,
we're experiencing crashes in the siptrace.so module with Kamailio
4.3.1. I looked at the core dump with gdb but it complained about
missing debugging symbols so all I could see was that it called
do_action() before branching into an unknown function in the siptrace
module.
I then installed t
I don't know if this is related, this happens on 4.3.1 as well but usrloc
crashes once it gets a DMQ
#0 0x7f2a82a5727c in get_urecord_by_ruid (_d=0x0, _aorhash=8118438,
_ruid=0x7ffe6028d5c0, _r=0x7ffe6028d568, _c=0x7ffe6028d578)
at udomain.c:1153
#1 0x7f2a8282029a in usrloc_get_all_u
i found a changes for rever_uri() function . i am using carrierroute module
to route changes. so after a query destination number , a prefix is added to
Request uri and sends request. after get error first request goes to
failure route and works revert_uri() . revert_uri changes to first request
u
Hello;
i am trying to upgrade kamailio to new version. i have a look
the db schema and version table , can't see any problem. jansson
and json packages are installed to try to new modules. (Thanks to
who add these. ) When it is build , gives errors about lib
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