What functions exist (if any), to add custom parameters to the From: and To:
headers?
___
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record routing log message about missing socket. Otherwise,
you should ignore it if the sip routing is ok.
Cheers,
Daniel
On 01/09/14 18:55, Alex Villacís Lasso wrote:
[...]
Maybe I should explain my setup better.
The test setup I want to run is, in a way, twice natted. The asterisk instance
runs
El 01/09/14 10:50, Alex Villacís Lasso escribió:
El 01/09/14 05:15, Daniel-Constantin Mierla escribió:
On 29/08/14 23:58, Andres wrote:
On 8/29/14, 1:42 PM, Alex Villacís Lasso wrote:
Please consider the following SIP packet exchange, as seen by a tcpdump running
on 201.234.196.170. Here
El 01/09/14 05:15, Daniel-Constantin Mierla escribió:
On 29/08/14 23:58, Andres wrote:
On 8/29/14, 1:42 PM, Alex Villacís Lasso wrote:
Please consider the following SIP packet exchange, as seen by a tcpdump running
on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170
El 29/08/14 14:44, Paul Belanger escribió:
On Fri, Aug 29, 2014 at 11:55 AM, Alex Villacís Lasso
wrote:
El 28/08/14 19:09, Paul Belanger escribió:
On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso
wrote:
As a continuation of my project, I am trying to set up Kamailio as a
Websocket
Please consider the following SIP packet exchange, as seen by a tcpdump running
on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170:
IP 198.58.101.75.5060 > 201.234.196.170.5060
INVITE sip:*43@201.234.196.170:5060 SIP/2.0
Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7
El 28/08/14 19:09, Paul Belanger escribió:
On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso
wrote:
As a continuation of my project, I am trying to set up Kamailio as a
Websocket bridge to Asterisk. The asterisk instance is running as localhost,
with its own websocket support disabled, but
As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for
websockets. Additionally, I have removed
El 26/08/14 12:02, Alex Villacís Lasso escribió:
El 25/08/14 18:28, Alex Balashov escribió:
On 08/25/2014 07:25 PM, Alex Villacís Lasso wrote:
However, I do not find an equivalent to bridge mode in the rtpengine
command-line parameters.
Bridging mode of this type is not supported by
El 25/08/14 18:28, Alex Balashov escribió:
On 08/25/2014 07:25 PM, Alex Villacís Lasso wrote:
However, I do not find an equivalent to bridge mode in the rtpengine
command-line parameters.
Bridging mode of this type is not supported by rtpengine.
If this is true, then mediaproxy-ng
I have a rtpproxy configuration that spawns several rtpproxy instances, using
bridge mode. An example is shown below:
/usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s udp:127.0.0.1
7723 192.168.2.18/127.0.0.1 -m 1 -M 2
Here, rtpproxy bridges between 192.168.2.18 and 127.0
El 13/08/14 03:12, Daniel-Constantin Mierla escribió:
Would be nicer not to forward all headers inline, that results in a message
easy to read (especially on mobile devices) and therefore faster to answer.
You use subst over contact and set contact allias, which both append values there, deleti
014 14:02:09 -0500
From: Alex Villacís Lasso
User-Agent: Mozilla/5.0 (X11; Linux x86_64; rv:24.0) Gecko/20100101
Thunderbird/24.7.0
MIME-Version: 1.0
To: sr-users@lists.sip-router.org
References: <53e54348.9090...@palosanto.com> <53e8712a.2050...@gmail.com>
In-Reply-To:
stdef "/ASTERISKPORT/5080/"
asterisk.bindip = "ASTERISKIP" ...
asterisk.bindport = ASTERISKPORT ...
subst_hf("Contact", "/ASTERISKIP:ASTERISKPORT/$td/", "a");
Cheers,
Daniel
On 08/08/14 23:38, Alex Villacís Lasso wrote:
Consider the fol
Consider the following snippet:
if (is_present_hf("Contact")) {
xlog("L_ALERT", "= reply to SUBSCRIBE has Contact: $ct\n");
xlog("L_ALERT", "= want to replace with $td\n");
xlog("L_ALERT", "= regexp to use is
/$sel(cfg_get.asterisk.bindip):$sel(cfg_get.asteris
El 01/08/14 01:24, Daniel-Constantin Mierla escribió:
The uac from/to replacement relies that parties keep the same from/to headers
content.
The mechanism to replace A with B is to combine both and get the key X which is
added in the record-route as parameter. Then practically from A and X res
El 31/07/14 11:11, Alex Villacís Lasso escribió:
El 30/07/14 17:33, Alex Villacís Lasso escribió:
My kamailio.cfg configuration file is attached.
I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP
El 30/07/14 17:33, Alex Villacís Lasso escribió:
My kamailio.cfg configuration file is attached.
I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP), presence seems to work correctly. However, when using
My kamailio.cfg configuration file is attached.
I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP), presence seems to work correctly. However, when using SIP.js via a websocket, Kamailio is unable to send the N
I am currently handling a system that runs kamailio and asterisk in the same machine. The kamailio instances are being used to emulate multiple SIP domains, by means of From/To mangling of incoming packets, which are then routed to Asterisk. The attached
kamailio.cfg does this work.
There is an
I am trying to track down a memory leak that was triggered by a patch I wrote for my local copy of kamailio 4.1.4 . For this, I am following the documentation at http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory . This page claims that once
memlog is set in the configuration file, a
El 02/07/14 15:02, Alex Villacís Lasso escribió:
El 02/07/14 10:51, Alex Villacís Lasso escribió:
El 30/06/14 04:24, Daniel-Constantin Mierla escribió:
Hello,
iirc, this is a check when trying to convert from base64 to old uri, but I
didn't have time to dig properly inside the source
El 02/07/14 10:51, Alex Villacís Lasso escribió:
El 30/06/14 04:24, Daniel-Constantin Mierla escribió:
Hello,
iirc, this is a check when trying to convert from base64 to old uri, but I
didn't have time to dig properly inside the sources.
I enhanced the log message to print the values o
El 01/07/14 14:44, Alex Villacís Lasso escribió:
El 26/06/14 18:39, Alex Villacís Lasso escribió:
I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink
El 30/06/14 04:24, Daniel-Constantin Mierla escribió:
Hello,
iirc, this is a check when trying to convert from base64 to old uri, but I
didn't have time to dig properly inside the sources.
I enhanced the log message to print the values of the uris, via commit:
-
http://git.sip-router.org/cgi
El 26/06/14 18:39, Alex Villacís Lasso escribió:
I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink wants to implement it), and MSRP. Ordinary voice
El 28/06/14 00:31, Juha Heinanen escribió:
there is at least one show stopper to make blink rls to work with any
standards compliant implementation. this is tracker item that i opened
year ago.
Do you have an URL for this tracker item?
-- juha
i tried to test address book stuff with blink, bu
Jun 27 15:28:42 elx /usr/sbin/kamailio[25189]: ERROR: uac [replace.c:590]:
restore_uri(): new URI shorter than old URI
Jun 27 15:28:44 elx /usr/sbin/kamailio[25187]: ERROR: uac [replace.c:590]:
restore_uri(): new URI shorter than old URI
Jun 27 15:28:48 elx /usr/sbin/kamailio[25188]: ERROR: uac
I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink wants to implement it), and MSRP. Ordinary voice calls work
correctly.
With presence, I have managed
So far, I have a kamailio.cfg that routes INVITEs between Asterisk and the external networks. Now I want to start adding MSRP support. However, with my current configuration, the test client (Blink) sends an INVITE with a SDP payload that specifies msrps
media. This gets routed to Asterisk, and it
Just a quick question. What is supposed to be wrong with the below message?
Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR:
[parser/parse_fline.c:243]: parse_first_line(): parse_first_line: bad message
(offset: 22)
Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: [parser/msg_parser
El 11/06/14 08:47, Daniel-Constantin Mierla escribió:
Hello,
can you put the patch on the tracker to review it -- in this way should not be
forgotten.
- http://sip-router.org/tracker/
I looked a bit, but has some new functions and touches quite a lot the existing
parts, so I need more time t
El 03/06/14 14:09, Alex Villacís Lasso escribió:
El 02/06/14 12:31, Alex Villacís Lasso escribió:
El 31/05/14 01:30, Daniel-Constantin Mierla escribió:
On 30/05/14 22:02, Alex Villacís Lasso wrote:
[...] The xml document retrieval is failing again. Now it is because retrieval
is
El 02/06/14 12:31, Alex Villacís Lasso escribió:
El 31/05/14 01:30, Daniel-Constantin Mierla escribió:
On 30/05/14 22:02, Alex Villacís Lasso wrote:
[...] The xml document retrieval is failing again. Now it is because retrieval
is truncating the MEDIUMTEXT field at 1024 characters:
Adjust
El 31/05/14 01:30, Daniel-Constantin Mierla escribió:
On 30/05/14 22:02, Alex Villacís Lasso wrote:
[...] The xml document retrieval is failing again. Now it is because retrieval
is truncating the MEDIUMTEXT field at 1024 characters:
Adjust the value of parameter:
http://kamailio.org/docs
El 30/05/14 12:42, Alex Villacís Lasso escribió:
El 30/05/14 11:02, Alex Villacís Lasso escribió:
El 29/05/14 12:15, Alex Villacís Lasso escribió:
El 29/05/14 07:51, Daniel-Constantin Mierla escribió:
Hello,
the error message is printed in case of an invalid xml document.
Can you edit
El 30/05/14 11:02, Alex Villacís Lasso escribió:
El 29/05/14 12:15, Alex Villacís Lasso escribió:
El 29/05/14 07:51, Daniel-Constantin Mierla escribió:
Hello,
the error message is printed in case of an invalid xml document.
Can you edit modules/xcap_server/xcap_misc.c and add inside
El 29/05/14 12:15, Alex Villacís Lasso escribió:
El 29/05/14 07:51, Daniel-Constantin Mierla escribió:
Hello,
the error message is printed in case of an invalid xml document.
Can you edit modules/xcap_server/xcap_misc.c and add inside function:
int xcaps_check_doc_validity(str *doc)
the
El 29/05/14 07:51, Daniel-Constantin Mierla escribió:
Hello,
the error message is printed in case of an invalid xml document.
Can you edit modules/xcap_server/xcap_misc.c and add inside function:
int xcaps_check_doc_validity(str *doc)
the log:
LM_ERR("xmld document is: [[%.*s]]\n", doc->len
I am trying to set up XCAP support for SIP SIMPLE, and using jitsi as the sample client. The kamailio database was previously configured using ODBC and kamailio-4.1.3 and was apparently working correctly. I was following the guide at
http://nil.uniza.sk/instant-messaging/simple/configuring-xcap-su
El 26/05/14 11:05, Alex Villacís Lasso escribió:
My current working theory is that the Record-Route headers are incorrect in the
original trace. The trace, as captured in the firewall, looks like this:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
38.126.208.41:5060;rport=5060;branch
My current working theory is that the Record-Route headers are incorrect in the
original trace. The trace, as captured in the firewall, looks like this:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
38.126.208.41:5060;rport=5060;branch=b18bbcc394bb5dae43946b12f5d5fe0e
Record-Route:
Record-Route:
The f
El 21/05/14 11:41, Alex Villacís Lasso escribió:
El 21/05/14 10:28, Alex Villacís Lasso escribió:
El 21/05/14 00:52, Juha Heinanen escribió:
Alex Villacís Lasso writes:
I am trying to explain the situation to our carrier, but I want to
rule out possible misconfigurations on our side. Are
El 21/05/14 10:28, Alex Villacís Lasso escribió:
El 21/05/14 00:52, Juha Heinanen escribió:
Alex Villacís Lasso writes:
I am trying to explain the situation to our carrier, but I want to
rule out possible misconfigurations on our side. Are there common
misconfigurations that produce the
El 21/05/14 00:52, Juha Heinanen escribió:
Alex Villacís Lasso writes:
I am trying to explain the situation to our carrier, but I want to
rule out possible misconfigurations on our side. Are there common
misconfigurations that produce the symptoms described here? Are there
any issues evident
I am trying to diagnose a SIP issue between our carrier and our network. The carrier has a CARRIER_IP and a different CARRIER_MEDIA_IP, and it submits an INVITE packet to MY_PUBLIC_IP using MY_DID. The firewall at our public IP (where the attached traffic
sample was taken) redirects it to a partic
I have this setup for kamailio + asterisk, in which kamailio is supposed to listen on all ethernet interfaces on UDP port 5060, and will forward traffic from/to asterisk running on the same machine, and listening on localhost, udp port 5080. The scenario
for the problematic call is somewhat like t
How do I configure kamailio to listen to multiple ports, on all known
interfaces?
I have tried "listen=udp:*:5060", "listen=udp:*:5062" on separate lines, but I
get the following errors:
Not starting : invalid configuration file!
0(7806) : [cfg.y:3411]: yyerror_at(): parse error in config fi
El 02/05/14 10:40, Alex Villacís Lasso escribió:
El 24/04/14 19:09, Alex Villacís Lasso escribió:
I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com
El 24/04/14 19:09, Alex Villacís Lasso escribió:
I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0
I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but
has been heavily modified. Currently asteri
I am having this issue when using kamailio-4.1.2 (after patching the segfault bug). The kamailio is configurated as a frontend for asterisk. When using UDP for SIP, this works correctly. However, I am now setting up websockets (no TLS yet), and I am
getting a disconnect after an interval. I am get
El 09/04/14 16:17, Daniel-Constantin Mierla escribió:
The crash was not related to websocket at all, but to usage of uac_reg_lookup()
when you don't set reg_db_url parameter for uac module.
So, my question is why are you using uac_reg_lookup()? Because it is useless if you dont set reg_db_url a
le is initialized only when reg_db_url is
set, otherwise there is no source of user profile to do registration for, and
no point to do uac_reg_lookup().
I will add safety checks for this case.
Cheers,
Daniel
On 09/04/14 18:30, Alex Villacís Lasso wrote:
El 09/04/14 03:11, Daniel-Constantin Mierla esc
param("uac", "reg_db_url", DBASTURL)
# uacreg table is actually a view in DBASTURL
#modparam("uac", "reg_contact_addr", "127.0.0.1")
Cheers,
Daniel
On 09/04/14 00:23, Alex Villacís Lasso wrote:
El 04/04/14 16:26, Alex Villacís Lasso escribió:
I am c
El 04/04/14 16:26, Alex Villacís Lasso escribió:
I am currently trying to replace a pure-Asterisk implementation of SIP
messaging through Websockets, with a Kamailio-4.1.2-based implementation.
However, when I try to send a message with jsSIP, Kamailio crashes:
Program terminated with signal
I am currently trying to replace a pure-Asterisk implementation of SIP
messaging through Websockets, with a Kamailio-4.1.2-based implementation.
However, when I try to send a message with jsSIP, Kamailio crashes:
Program terminated with signal 11, Segmentation fault.
#0 0x7f0e5cf31be3 in r
El 24/03/14 09:46, Alex Villacís Lasso escribió:
El 24/03/14 04:17, Daniel-Constantin Mierla escribió:
Hello,
I found some cases when variables were not freed, but I cannot test as I am not
using unixodbc.
Can you cherry pick the patch:
-
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip
El 24/03/14 04:17, Daniel-Constantin Mierla escribió:
Hello,
I found some cases when variables were not freed, but I cannot test as I am not
using unixodbc.
Can you cherry pick the patch:
-
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=edc78dfb148c22f0d256485193bbdb0185
I have a CentOS 6 installation with the following packages installed from the
RPM build service from Kamailio:
kamailio-unixodbc-4.1.2-1.1.x86_64
kamailio-4.1.2-1.1.x86_64
kamailio-presence-4.1.2-1.1.x86_64
kamailio-utils-4.1.2-1.1.x86_64
I am also using db_unixodbc for all database accesses (w
On the previous issue (disjoint networks), I eventually settled for binding Asterisk on localhost:5080, and using rtpproxy to route media to the networks as required. The only snag is that I had to write a script to rewrite the Kamailio configuration in
order to take current IPs into account.
No
El 26/02/14 11:39, Alex Villacís Lasso escribió:
El 26/02/14 05:25, Klaus Darilion escribió:
Puh, too many questions in one email.
I am sorry about that. I wanted to provide as much useful information as possible on my email, including what I had observed, and what I have already tried. After
) asterisk-only scenario, media is routed between the test networks through asterisk when it does all of the SIP
negotiation itself.
So, what setup have you choosen? Then we can think about problems.
regards
Klaus
Am 25.02.2014 23:31, schrieb Alex Villacís Lasso:
As part of a project, I h
As part of a project, I have installed a CentOS 6 test system (a virtual machine) with Asterisk 11.7.0 and Kamailio 4.1.1 downloaded from http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_CentOS-6/x86_64/. I am trying to setup a
combination of Kamailio and Asterisk that wi
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