hello :
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
that describe the realtime integration between Kamailio and Asterisk.
I have same question:
"Asterisk listens on IP 192.168.178.25 port 5080"
which asterisk's config
@Olle
"If you have a server on a public IP running behind Kamailio you might not
need RTPproxy relaying for calls to and from that server. Asterisk will
handle NAT by itself and doesn't need help if you turn on NAT support in
Asterisk. In that case, RTPproxy just adds delay to your calls."
Can yo
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Hi Daniel,
Thanks for the reply. Can you point me to some configuration snippets,
wikis etc. I'm on a bit of a steep learning curve at the moment as I
am new to Kamailio. Many thanks for this software!
Cheers,
John
On 07/05/13 08:23, Daniel-Constant
Hi Khoa. You are correct. This means that the server calls shutdown very
shortly after accepting the connection. The server has reach max TCP connection
limit, or server is busy handling other connections. The following link "Why
will a TCP Server send a FIN immediately after accepting a connect
Leo Brown writes:
> Is this normal behaviour of SIP, and either way is there any way to
> ask Kamailio to mangle the 200 OK to split the Record-Route header
> into the separate entries that is possibly more "standard"…?
route uris can be split into one or more r-r headers. sip UAs need to
be abl
Hi Henning,
My inbound carrier has done some digging and discovered that the reason is they
choose the *first* address from the *last* Record-Route header.
Most gateways apparently add multiple Record-Route headers, whereas when we get
a OK to our INVITE from the carrier, the various Record-Rou
On May 7, 2013, at 3:36 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 5/6/13 6:26 PM, Andrzej wrote:
>> Is it possible to run on phones Cisco 79xx series BLFs? So to show the
>> status of the shared lines. Phones registered with Kamailio.
> look at pua_dialoginfo and presence_dialoginfo
2013/5/7 Daniel-Constantin Mierla :
> Hello,
>
> among the topics discussed just before the last major release series (4.0.x)
> was one about restructuring the source code tree. It started mainly as a
> proposal to move source code belonging to core in a dedicated folder, but
> there could be more
Hello,
among the topics discussed just before the last major release series
(4.0.x) was one about restructuring the source code tree. It started
mainly as a proposal to move source code belonging to core in a
dedicated folder, but there could be more variants. It's time start
discussing if we
Hello,
it's probably the time to have a IRC development meeting to set the
targets for the next major release as well as discuss what is needed
these days around the project.
I proposed next week on Thursday, May 16, at 14:00GMT (16:00 Berlin
time). I created a page to collect the topics, on
Hi all,
I use ims_icscf in Kamailio to send LIR message to HSS. But, In LIR message
that I received in HSS haven't OriginatingRequest AVP.
How I can insert it into LIR message?
Thanks & Best regards,
Khue Nguyen.
___
SIP Express Router (SER) and Kamail
It's great!
Thank you,
Julia
_
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel-Constantin Mierla
Sent: Tuesday, May 07, 2013 10:19 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] max number of pipes
Hi Daniel,
Thansk for the reply.
Please find the attached ngrep data for port 5060
I have used QuteCom client
Best Regards,
Roy.
On Tue, May 7, 2013 at 12:41 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
>
> On 5/7/13 9:05 AM, Juha Heinanen wrote:
>
>> Raj Roy Ghandhi writes:
>>
>> When
Hello,
On 5/6/13 6:26 PM, Andrzej wrote:
Is it possible to run on phones Cisco 79xx series BLFs? So to show the
status of the shared lines. Phones registered with Kamailio.
look at pua_dialoginfo and presence_dialoginfo modules. If they support
the broadsoft specs, then sca module is another on
7 maj 2013 kl. 05:10 skrev Khoa Pham :
> Hi,
>
> As someone asked in
> http://lists.sip-router.org/pipermail/sr-users/2010-December/066792.html
> about TCP overload, Daniel answerd that
>
> "if you have some lengthily operations that have to be applied to your sip
> traffic and you get cong
I don't have asterisk/freeswitch now. But i'm considering it now.
How is rtpproxy performance compared to asterisk/freeswitch for handling media
relay.
I only need the media relay capability. Don't need conference, ivr,
transcoding, etc...
- Pesan Asli -
Dari: Olle E. Johansson
If y
Hi Klaus,
Thanks for your detailed explanation. Really appreciate it.
- Pesan Asli -
Dari: Klaus Darilion
Kepada: John Chen ; Kamailio (SER) - Users Mailing List
Cc:
Dikirim: Senin, 6 Mei 2013 14:49
Judul: Re: [SR-Users] Do we need rtpproxy for all kind of NAT?
On 04.05.2013 16:37
Hello,
On 5/7/13 12:36 AM, johnc wrote:
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Hi,
I have built a kamailio based phone system along the lines of the
tutorial below. I have updated it for version 4.0 and it is running
successfully on Debian 7.0. I have set my system up with a Gandi SSL
c
Hello,
On 5/6/13 7:30 PM, Julia wrote:
Hello,
Is there a limit on the max number of pipes for *pipelimit *module?
no, there is no limit. You can define as many as you want in database
table, provided that you have enough memory to store the associated
structures in memory (which is hard to
Hello,
On 5/7/13 9:05 AM, Juha Heinanen wrote:
Raj Roy Ghandhi writes:
When I register with Jitsi or QuteCom I get
5(10515) ERROR: presence [publish.c:388]: No E-Tag and no body found
6(10517) INFO: presence [notify.c:1601]: NOTIFY sip:1...@sip.abc.com via
sip:1001@124.43.201.156:5060 on
Raj Roy Ghandhi writes:
> When I register with Jitsi or QuteCom I get
>
> 5(10515) ERROR: presence [publish.c:388]: No E-Tag and no body found
> 6(10517) INFO: presence [notify.c:1601]: NOTIFY sip:1...@sip.abc.com via
> sip:1001@124.43.201.156:5060 on behalf of sip:1...@sip.abc.com for event
>
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