[SR-Users] kamailio's Makefile system : Enabling CFLAGS/LDFLAGS for udp_server.c

Hi, I'm doing some experimentation on kamailio's udp_server.c. Now i need libpcre for that. So i first thought if i enable the regex module, the CFLAGS/LDFLAGS would be available from udp_server.c. But it turns out it doesn't build(doesn't *link* to be more precise). Kamailio's build system is pr

[SR-Users] Detecting problems of kamailio under very high load

We have multiple kamailio servers with 4 cpu cores and 16G RAM. We use kamailio+rtpproxy as a outbound sip proxy. Usually there are many thousands of concurrent sip sessions of occurring there. Periodically sometimes it just stops serving request and spits out 5** replies. At that point we usually

[SR-Users] How to forward sip message "statefully" so that the dialogue is preserved

We want to implement the following topology : INVITE--->kamailio1-forward--->kamailio2--->route[NATMANAGE]--->Next Sip endpoint Client < route[NATMANAGE]

Re: [SR-Users] Making rtp streams appear unidirectional

On Mon, Jan 12, 2015 at 5:52 PM, Daniel-Constantin Mierla wrote: > > On 12/01/15 11:14, aft wrote: > > > > On Fri, Jan 9, 2015 at 12:37 AM, Daniel-Constantin Mierla < > mico...@gmail.com> wrote: > >> Hello, >> >> On 01/01/15 08:29, aft wrote: &

Re: [SR-Users] Making rtp streams appear unidirectional

On Fri, Jan 9, 2015 at 12:37 AM, Daniel-Constantin Mierla wrote: > Hello, > > On 01/01/15 08:29, aft wrote: > > Hi, > > > > Is it possible to make the rtp stream appear unidirectional? > > > > By that i mean, > > > > The rtp stream from client

[SR-Users] Making rtp streams appear unidirectional

Hi, Is it possible to make the rtp stream appear unidirectional? By that i mean, The rtp stream from client to proxy will go through one rtpproxy and proxy to client stream will go through another rtpproxy instance? If not, is it possible to mimic something like that by running rtpproxy in brid

[SR-Users] Making media packets associated with calls appear unidirectional using rtpproxy and kamailio

Hi, I'm trying use two instances of rtpproxy listening to two different public IPs, to handle incoming and outgoing legs. That means i want to implement this network topology: SipClient->media from client to proxy--->rtpproxy1 Sipclient<--media from proxy to client<---rtpproxy2

Re: [SR-Users] NATMANAGE quirks

hem and you should get a > better idea of what happens there. > > When $du is set? I mean if it's set elsewhere, i can save it in a variable and then use it in NATMANAGE route > Cheers, > Daniel > > > On 21/10/14 14:29, aft wrote: > > Hi, > > I'

[SR-Users] NATMANAGE quirks

Hi, I'm trying to re packet rtp streams based on Destination IP. I've implemented it like below : route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(F

Re: [SR-Users] Creating array out of hashtable

Thanks for the reply. It was very helpful. On Mon, Sep 8, 2014 at 3:38 PM, Daniel-Constantin Mierla wrote: > Hello, > > On 08/09/14 10:50, aft wrote: >> >> [1]Is it possible to iterate a htable without using "keys" like an array? > > > there is no po

[SR-Users] Creating array out of hashtable

[1]Is it possible to iterate a htable without using "keys" like an array? [2] Is it possible to search for "keys" using regxp? -- -Cheers -Arif ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org h

Re: [SR-Users] Adding IPs from which 503 messages originates to a database

On Thu, Sep 4, 2014 at 1:57 PM, Daniel-Constantin Mierla wrote: > > On 04/09/14 09:19, aft wrote: >> >> On Wed, Sep 3, 2014 at 11:36 PM, Daniel-Constantin Mierla >> wrote: >>> >>> Hello, >>> >>> The $si is the source ip of the sip pack

Re: [SR-Users] Adding IPs from which 503 messages originates to a database

o use mqueue to push sql queries from sip worker to rtimer as > you need to write something to database. > Should i push $si using mq_add() and fetch it using mq_fetch()? > Cheers, > Daniel > > > On Wednesday, September 3, 2014, aft wrote: >> >> I'm tryi

[SR-Users] Adding IPs from which 503 messages originates to a database

I'm trying to configure HOMER sip capture server to do some accounting of the cases when "503 Service Unavailable" message arrives from capture agents. I've implemented the following to record the source IPs from which 503 message originates. onreply_route { ..

Re: [SR-Users] Adding a media attribute to SDP

log("found rtpproxy") is not executed. But from a packet capture, i can see it does have a "a=nortpproxy:yes" field. Why its not matching? On Mon, Sep 1, 2014 at 4:23 PM, aft wrote: > On Mon, Sep 1, 2014 at 4:18 PM, Daniel-Constantin Mierla > wrote: >> Hello, >

Re: [SR-Users] Adding a media attribute to SDP

ld i search for? I do not see an "append_body()" method :( > > Cheers, > Daniel > > > On 01/09/14 12:12, aft wrote: >> >> Hi, >> >> I want to add this media attribute to SDP. >> >>a=ptime: >> >> [1] Its not cover

[SR-Users] Adding a media attribute to SDP

Hi, I want to add this media attribute to SDP. a=ptime: [1] Its not covered in SDPops module [2] its not added when i use rtpproxy_manage("z") to repacketize the rtp stream. So how should i do it? -- -Cheers -Arif ___ SIP Express Router (SER)

Re: [SR-Users] What does "::" operator do in statements related to hashtable?

27;s not an operator, actually. :-) It's just a convention. You can call >> your keys anything you like (that's grammatically valid). >> >> On 08/28/2014 09:37 AM, aft wrote: >> >>> Hi, >>> >>> From kamailio documentation, the usage of has

[SR-Users] What does "::" operator do in statements related to hashtable?

Hi, >From kamailio documentation, the usage of hashtable is given as : modparam("htable", "htable", "a=>size=4;") ... $sht(a=>test) = 1; $sht(a=>$ci::srcip) = $si; I get it that in the first statement, a is the hashtable, a new key-value pair is added to one of its empty bucket (test,1). What i

[SR-Users] Understanding hashtable usage

Hi, I've got a piece of config code which should solve my problem, but i'm not getting the code. It involves hashtable module. Here it goes : modparam("htable", "htable", "a=>size=8;autoexpire=400") modparam("htable", "htable", "b=>size=8;autoexpire=31") Ok we have two hashtable , a and b, both

Re: [SR-Users] Problem with rtpproxy and rtpproxy_manage()

y. Actually packets come from gateway fine at the rtpproxy box. But these packets are not relayed to the caller. See my new mail for details. > > Cheers, > Daniel > > > On 08/05/14 13:25, aft wrote: > > On Wed, May 7, 2014 at 3:17 PM, aft wrote: > > On Wed, May 7, 2

[SR-Users] One way audio

Hi, The network topology is : SipSoftphone(caller)->kamailio/rtpproxy>Softswitch>sipGateway-->mobile phone(callee) The rtpproxy is managed in kamailio script by : if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) {

Re: [SR-Users] Problem with rtpproxy and rtpproxy_manage()

On Wed, May 7, 2014 at 3:17 PM, aft wrote: > On Wed, May 7, 2014 at 2:50 PM, Daniel-Constantin Mierla > wrote: >> Hello, >> >> you should provide a ngrep output of such call (incoming invite to the >> forwarded ack for 200ok), we can check the sdp. [1] T

Re: [SR-Users] Problem with rtpproxy and rtpproxy_manage()

and, I didn't have good experiences with rtpproxy application > from git head, can you try with 1.2.1? Well the same thing happens with 1.2.1 also. Reasons for using the git-head is to see whether the problem went away. > > Cheers, > Daniel > > > On 07/05/14 08:34, aft wr

[SR-Users] Problem with rtpproxy and rtpproxy_manage()

Hi, I'm using kamailio from latest git-HEAD. The rtpproxy i'm using also from latest git. Our network topology is following : sip-softphone->kamailio/rtpproxy>softswitch>gateway Because of saving bandwidth we need to use the "re-packetization" feature of rtpproxy. When we d

Re: [SR-Users] Module version mismatch ERROR

On Mon, Jul 1, 2013 at 10:48 PM, Victor Seva wrote: > You have to build kamailio and all the modules that you are using > again. Don't know why this is not been removed. I've done that. To be honest have done multiple times. Something is getting cached i guess. I don't know about kamailio's usage

Re: [SR-Users] Module version mismatch ERROR

module_interface_ver 169: 0021b608 8 OBJECT GLOBAL DEFAULT 22 module_flags 193: 0021b610 8 OBJECT GLOBAL DEFAULT 22 module_version On Mon, Jul 1, 2013 at 8:18 PM, Victor Seva wrote: > 2013/7/1 aft : >> Hi, >> >> After updating my git tr

[SR-Users] Module version mismatch ERROR

Hi, After updating my git tree I'm on this commit : * master1123ed4 core: Also consider PROTO_WS(S) in forward(). origin/master 1123ed4 core: Also consider PROTO_WS(S) in forward(). After building kamailio is showing this an

Re: [SR-Users] Installing Kamailio on EC2

What have you tried? On Wed, Jun 26, 2013 at 2:55 PM, Sunil Chandrasekharan wrote: > Hi All, > > I am trying to install Kamailio on Amazon EC2. I read out in web about some > people already achieved configuring Kamailio on EC2. > > Kindly support on the pre-requisties to be installed before i go

Re: [SR-Users] [SOLVED] Not getting correct result after sending a OPTION query

istrar. > > The subscriber table only shows what users can subscribe to your system and > with what credentials. Thanks for the answer. That worked like a charm. > > Marius > > > On Sat, Jun 22, 2013 at 3:41 PM, aft wrote: >> >> ( I posted a same message through gm

[SR-Users] Not getting correct result after sending a OPTION query

( I posted a same message through gmane interface. gmane is not working right now at my place, So i can't respond to the replies i got there). I have now these users : arif@khost:~$ kamctl db show subscriber database engine 'MYSQL' loaded Control engine 'FIFO' loaded ++--+

[SR-Users] Not getting correct result after sending a OPTION query

Hi, I'm trying to understand intricacies of SIP protocol. I'm installed a stock kamailio from git repo. kamcmd> core.version kamailio 4.0.2 (x86_64/linux) f87866 Now i'm trying to send OPTION request by "sipsak". I've added two users : arif@khost:~$ kamctl db show user ... ++-

Re: [SR-Users] Simulating kamailio config flows

On Mon, Apr 22, 2013 at 9:08 PM, Andreas Granig wrote: > Hi, > > I'd like to put a topic up for discussion to test kamailio config logics. > > The standard way of doing so is to start kamailio as usual and write sipp or > sipsak scenarios to perform automated tests. This can get quite complex > pr

Re: [SR-Users] No RTP between rtpproxy and smartphone application

check sdp's at all pit stops On Apr 1, 2013 10:24 PM, " " wrote: > Hello, > I have the following topology. > Kamailio as SIP Proxy and Asterisk as B2BUA. In Kamailio I use rtpproxy > for NAT users. > The problem that I have is the following: > -When the CSipSimple is registered the first

Re: [SR-Users] mediaproxy-ng Tutorial

fascinating stuff. On Apr 5, 2013 6:06 PM, "Richard Fuchs" wrote: > On 04/05/13 03:53, Daniel-Constantin Mierla wrote: > > > She fallback to user space can happen even during a call? Or is just > > about when the call is initialized, the application detects is some > > problem when setting up for

Re: [SR-Users] mediaproxy-ng Tutorial

On Wed, Apr 3, 2013 at 5:50 AM, Richard Fuchs wrote: > On 04/02/13 17:39, aft wrote: > >> So the bottom line is i have to include the code in both places. >> >> Another thing is i'm assuming you know much about the development of this >> media >> re

Re: [SR-Users] mediaproxy-ng Tutorial

On Tue, Apr 2, 2013 at 9:13 PM, Richard Fuchs wrote: > On 04/02/13 10:54, aft wrote: > >> I was actually asking How it works? I mean when there is kernel based >> forwarding is enabled, what does the daemon do compared to when the kernel >> based forwarding is not enabled?

Re: [SR-Users] mediaproxy-ng Tutorial

On Tue, Apr 2, 2013 at 8:42 PM, Richard Fuchs wrote: > On 04/02/13 10:02, aft wrote: > >> Daemon installation failed with the following : >> >> call.c:15:27: fatal error: xmlrpc_client.h: No such file or directory > > Check out the list of dependencies in the debian/

Re: [SR-Users] mediaproxy-ng Tutorial

On Tue, Apr 2, 2013 at 7:44 PM, Richard Fuchs wrote: > On 04/02/13 09:15, aft wrote: > >> So i was asking how to install mediaproxy-ng itself? > > If you're on a Debian system, you can simply issue dpkg-buildpackage and > then install the packages it produces. Oth

Re: [SR-Users] mediaproxy-ng Tutorial

On Tue, Apr 2, 2013 at 7:05 PM, Richard Fuchs wrote: > Hi, > > On 04/01/13 10:03, Aft nix wrote: > >> I stumbled upon this git://github.com/sipwise/mediaproxy-ng.git which >> looked very neat to me. Its said that it can be used with kamailio. It >> seems like its ba

Re: [SR-Users] mediaproxy-ng Tutorial

ot;sipwise" patched kamailio source has a module mediaproxy-ng to use mediaproxy-ng with kamailio. > dani > > > On Mon, Apr 1, 2013 at 5:03 PM, Aft nix wrote: >> >> Hi, >> >> I stumbled upon this git://github.com/sipwise/mediaproxy-ng.git which >> look

[SR-Users] mediaproxy-ng Tutorial

Hi, I stumbled upon this git://github.com/sipwise/mediaproxy-ng.git which looked very neat to me. Its said that it can be used with kamailio. It seems like its backed sipwise inc. But no documentation is given there. Anyone know of a tutorial/documentation for how to use it? -- -aft

[SR-Users] Dialog is not cleaned up after failed calls and BYEs.

G_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 83eafc compiled on 18:34:58 Jan 17 2013 with gcc 4.4.6 Any suggestion?(It seems dialog module is not working as it should be. I've also tried calling unset_profile() forcefully in failure route with no luck) -- -af

Re: [SR-Users] Use different flags in rtpproxy_manage() based on from which route its called

On Fri, Nov 9, 2012 at 2:08 PM, Daniel-Constantin Mierla wrote: > Hello, > > > On 11/8/12 11:40 AM, Aft nix wrote: >> > in route[NATMANAGE] you can see that there are some functions that help to > detect if a request or a reply is processed, respectively > is_request(

[SR-Users] Use different flags in rtpproxy_manage() based on from which route its called

anage() is it possible to emulate the above functionality? Thanks in advance. -- -aft ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] implementing call-quota with dialog module

ewhat flawed. The idea was to count calls per "domain". If the the idea is reflected in my approach then i will head for minor tweaking. But if my implementation is just plain wrong, then i will have to do it manually, like using sqlops and do the dialog accounting myself. cheers. >

[SR-Users] implementing call-quota with dialog module

UG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$fd"); if (get_pro

Re: [SR-Users] Error starting kamailio with dialog module usage

On Sun, Sep 30, 2012 at 10:15 PM, SamyGo wrote: > Is it typo or by chance you are missing a $ in line 507 before var(100) > You are spot on :) Thanks. > On Sep 30, 2012 2:50 PM, "Aft nix" wrote: >> >> Hi, >> >> After implementing a "C

[SR-Users] Error starting kamailio with dialog module usage

when i issue : $kamailio -c but it fails to start with following info: Sep 29 19:37:49 108 /usr/local/sbin/kamailio[1865]: ERROR: [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:507 Sep 29 19:37:49 108 /usr/local/sbin/kamailio[1865]: ERROR: [route.c:1216]:

[SR-Users] Limiting concurrent calls

ofile altogether. Any code snippet to limit number of concurrent calls will be very helpful. Thanks in advance. -- -aft ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.or

[SR-Users] Adding "make menuconfig"

ne who know kamailio build system well plus some intro to ncurses programming. So i guess devs should look into it. cheers aft ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router

Re: [SR-Users] sip over tls is not working

ets over TLS works which requires establishing a TLS connection >>> and >>> exchanging an HTTP request and response. It doesn't sound like this >>> connection is even getting passed the TLS handshake part? >>> >>> Peter >>> >> >> Hi

Re: [SR-Users] sip over tls is not working

t; > Peter > Hi, That was my first guess. I will run some tests with plain tcp socket and post update. cheers. > > On Wed, 2012-07-11 at 17:14 +0200, Klaus Darilion wrote: > > Maybe there were some changes fore websocket support which cause > problems. Do plain TCP connectio

Re: [SR-Users] sip over tls is not working

xhttp_reply("200", "OK", "text/html","OK - $hu - > [$si:$sp]"); > } > > > regards > Klaus > > > > On 10.07.2012 12:44, Aft nix wrote: >> >> On Mon, Jul 9, 2012 at 10:24 PM, Daniel-Constantin Mierla >> wr

Re: [SR-Users] sip over tls is not working

On Wed, Jul 11, 2012 at 6:56 PM, Klaus Darilion wrote: > I just tested TLS with Kamailio 3.3.0 and Eyebeam and it works. Make sure to > specify "ca_list" if intermediate certificates are used. > I was working with master branch, not 3.3 branch. > > regards > Klaus

Re: [SR-Users] DB create statements

R) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- -aft ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] sip over tls is not working

M, Klaus Darilion wrote: >> >> Use wireshark to analyze the TLS handshake >> >> regards >> klaus >> >> On 09.07.2012 13:27, Aft nix wrote: >>> >>> Hi, >>> >>> I have enabled tls parameters as follows: >>> >>>

Re: [SR-Users] sip over tls is not working

On Mon, Jul 9, 2012 at 7:04 PM, Klaus Darilion wrote: > Use wireshark to analyze the TLS handshake > Thanks for the suggestion. I'll analyze it and post my findings. > regards > klaus > > > On 09.07.2012 13:27, Aft nix wrote: >> >> Hi, >> >>

[SR-Users] sip over tls is not working

receiver, connection passed to the least busy one (3289651) [tcp_main.c:3967]: selected tcp worker 0 0(8) for activity on [tls::], 0xb5701580 [tcp_main.c:3576]: BUG: handle_ser_child: fd -1 for 0 (pid 2491) I'm using kamailio from git. its updated to the latest. Thanks in advance. --

Re: [SR-Users] New developer - Vicente Hernando

ooking forward to your contributions! > > Cheers, > Daniel Happy codding, Vincente cheers, aft ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] $200 bounty

Daniel-Constantin Mierla writes: > Hello, > > in my opinion this discussion got a bit too much off-topic in several > messages, inducing inappropriate feeling about how this community > behaves. Please everyone have in mind: > > - every message here is sent to all subscribed people, it is not > b

Re: [SR-Users] $200 bounty

n request. =) > To bad back port breaks the build. :) cheers. > With best regards, > > Fred > http://qxork.com > > _______ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.or

Re: [SR-Users] $200 bounty

ssy artsy girl. > David > -- -aft ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] $200 bounty

SER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- -aft ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] $200 bounty

r than "product" based. Its quiet ok for you to hire someone who is active in kamailio list. But "kamailio-users" is not the place to talk business. We all have to support us and our families. But do it in business list please. I dont think people will show different response if

Re: [SR-Users] Looking for RTP Proxy in TCP

On Fri, Jun 8, 2012 at 6:41 PM, Daniel-Constantin Mierla wrote: > > On 6/8/12 1:26 PM, Aft nix wrote: >> >> On Fri, Jun 8, 2012 at 2:18 PM, Andrew Pogrebennyk >> wrote: >>> >>> The papers talk about transport protocol for signaling, not media/RTP. >

Re: [SR-Users] Looking for RTP Proxy in TCP

ese practices force VOIP providers for "obsfucation" techniques for escaping the DPI. Interesting thing is i know of some situation when they blocked VPN because people were using it to make VOIP calls. Now you can use VPN for other purposes! Some telco even went to lengths to block UDP &qu

Re: [SR-Users] Limiting calls from users in prepaid scenerio

On Thu, Jun 7, 2012 at 5:24 PM, Daniel-Constantin Mierla wrote: > Hello, > > > On 6/5/12 11:17 AM, Aft nix wrote: >> >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Hi all, >> >> We are thinking about launching a different "

[SR-Users] Limiting calls from users in prepaid scenerio

cfg. I don't know if any module in kamailio exports mysql_queury() to cfg, which i need for my purpose. And may be there is some built in module which provides a higher level layer which will enable myself achieving my goal. If in the end i need "exported" mysql functions

Re: [SR-Users] [NOT Solved]Creating RURI ($ru) from Contact ($ct)

Hi, My problem is now fixed. The last trick was adding a is_direction("upstream") in route[WITHINDLG] for choosing which BYE message i need to modify. Thanks everyone for your help. Cheers. -- -aft ___ SIP Express Router (SER) an

Re: [SR-Users] [NOT Solved]Creating RURI ($ru) from Contact ($ct)

On Wed, May 30, 2012 at 4:37 PM, Andrew Pogrebennyk wrote: > Hi, > > On 05/30/2012 12:22 PM, Aft nix wrote: >> So i'm interested if RFC 3261 provides any mechanism by which we can >> differentiate a BYE whether its from "caller" or "callee".

Re: [SR-Users] [NOT Solved]Creating RURI ($ru) from Contact ($ct)

On Tue, May 29, 2012 at 3:03 PM, Aft nix wrote: > On Tue, May 29, 2012 at 2:57 PM, Daniel-Constantin Mierla > wrote: >> Hello, >> >> in kamailio, if you want to apply immediately the changes done to the sip >> request, then use msg_apply_changes() from textopsx mod

Re: [SR-Users] [NOT Solved]Creating RURI ($ru) from Contact ($ct)

pvs for source ip and port are $si and $sp -- see the PV cookbook > in the wiki site at kamailio.org > > Cheers, > Daniel > Hi Daniel, Thank you for quick reply. I will test both and post updates. Cheers. > > On 5/29/12 10:39 AM, Aft nix wrote: >> >> Hi, >>

Re: [SR-Users] [NOT Solved]Creating RURI ($ru) from Contact ($ct)

atenation). 2. Or if possible i reference the changed contact from the temporary storage, and put this into my hashtable. Please direct me which way should i go. Cheers. On Mon, May 28, 2012 at 8:25 PM, Aft nix wrote: > Hi all, > I'm posting the script which worked. In future if any

[SR-Users] Fwd: [Solved]Creating RURI ($ru) from Contact ($ct)

            }  #!endif Thanks everyone for the help. I'm marking this thread as solved. -- -aft -- -aft ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Creating RURI ($ru) from Contact ($ct)

On Mon, May 28, 2012 at 4:52 PM, Daniel-Constantin Mierla wrote: > Hello, > > On 5/27/12 12:46 AM, Aft nix wrote: >> >> [...] >> >> Hi Aleksandrov >> Thanks for the reply. >> >> Yes my checks should be more specific. >> >> On the side

Re: [SR-Users] Creating RURI ($ru) from Contact ($ct)

On Sun, May 27, 2012 at 3:57 AM, Vitaliy Aleksandrov wrote: > On 05/26/2012 05:39 PM, Aft nix wrote: > > Hi, > > I'm trying to create a ruri from contact header like following : > > In request route I've saved contact header in a hashtable using h

[SR-Users] Creating RURI ($ru) from Contact ($ct)

Hi, I'm trying to create a ruri from contact header like following : In request route I've saved contact header in a hashtable using htable module. request_route{ - - #!ifdef WITH_HASH if (is_method("INVITE") && !has_totag()){ xdb

Re: [SR-Users] kamailioi is not relaying BYE message to UAC

if its possible to detect the fault in this BYE > and construct a new one and then relay it to the UAC. > > I mean can i do this : > > contact-header = INVITE's contact-header > if (contact-header != BYE's ruri) > { > construct BYE message with contact header >

[SR-Users] Detect and Correct wrong BYEs from proxy.

(is_method('BYE')) { t_relay() } } -- -aft ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] kamailioi is not relaying BYE message to UAC

nd then relay it to the UAC. I mean can i do this : contact-header = INVITE's contact-header if (contact-header != BYE's ruri) { construct BYE message with contact header t_relay() } Cheers aft > > On 5/21/12 9:21 PM, Arif Hossain wrote: >>