Re: [SR-Users] XCAP configuration

Dear Sir, In case of presence the condition which you mentioned works (i.e if($hu=~"^/xcap-root") { }) but when i want to capture the authentication request from sipml5 client(SIP over websocket), the conditions does't match(e.g if($hu=~"^/webrtc") or if($hu=~"^/sip:" etc). When i tri

Re: [SR-Users] Kamailio does not send reply to source port

On 19.08.2013 13:04, Alex Balashov wrote: On 08/19/2013 07:00 AM, Mino Haluz wrote: (palmface). I am always forgetting about this, it should be called where initial request is processed or also for sequential requests? Initial. Sequential requests and their replies are sent according to a

Re: [SR-Users] kamailio configuration for pass-thru proxy and username rewrite

If you want to hide your upstream providers from your customers, you should use a B2BUA (sems, Asterisk). Probably the most simple setup would be: customers -- Kamailio -- Asterisk -- PSTN Provider Kamailio is used for Registrations, NAT traversal, Authentication ... Calls to the

Re: [SR-Users] TLS versions and ciphers questions.

Kamailio uses OpenSSL. Thus, it mainly supports what your OpenSSL version supports. This means, Kamailio can use all TLS versions and ciphers that your libssl supports. You can configure Kamailio to use certain ciphers, see: http://kamailio.org/docs/modules/4.0.x/modules/tls.html#cipher_list Y

[SR-Users] Writing a module

Hello, Is this page still up-to-date: http://www.kamailio.org/dokuwiki/doku.php/development:write-module ? I'm researching what it takes to create our own custom Kamailio module. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing li

Re: [SR-Users] Writing a module

On 08/20/2013 06:41 AM, Grant Bagdasarian wrote: Hello, Is this page still up-to-date: http://www.kamailio.org/dokuwiki/doku.php/development:write-module ? I’m researching what it takes to create our own custom Kamailio module. 1. Skilled C programming. 2. What do you want to accomplish? A

Re: [SR-Users] XCAP configuration

Hi Daniel, I inspect the value of $hu .When I logging into kamailio server from SIPML client (SIP over WebSocket) it prints only"/" . Now if I write the following inside event_route[xttp:request] as follows it gives parsing error. if($hu=~"/") { # handle webrtc }

Re: [SR-Users] Writing a module

It's going to be a custom routing module, with our own business logic. It should load the routes from a database and keep refreshing the data every x interval. For SIP-to-PSTN calls (outgoing to carrier) we need to be able to route based on the callerprefix, customer id (which is added as a cu

Re: [SR-Users] Writing a module

On 08/20/2013 07:33 AM, Grant Bagdasarian wrote: Implementing this logic in a stored procedure is quite hard. Why? It's exactly what our Kamailio-based LCR/routing/accounting product does. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur,

Re: [SR-Users] Writing a module

20 aug 2013 kl. 13:33 skrev Grant Bagdasarian : > It's going to be a custom routing module, with our own business logic. > > It should load the routes from a database and keep refreshing the data every > x interval. > For SIP-to-PSTN calls (outgoing to carrier) we need to be able to route bas

Re: [SR-Users] Writing a module

Hello, devel guide is useful for writing own module: - http://www.asipto.com/pub/kamailio-devel-guide/ Also, it was a recent discussion on sr-dev with hits about modules to look at: - http://lists.sip-router.org/pipermail/sr-dev/2013-August/020977.html Cheers, Daniel On 8/20/13 1:33 PM, Gran

Re: [SR-Users] Writing a module

Alright, thanks guys! I'll have a closer look at the existing modules. -Original Message- From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel-Constantin Mierla Sent: Tuesday, August 20, 2013 1:37 PM To: Kamailio (SER) - Users Ma

Re: [SR-Users] Writing a module

There's also a Perl module that can be used to execute a Perl script containing your business logic. Sounds like you are doing lots of database queries and doing that from a C module will always be more difficult than using a DB abstraction layer that the one Perl provides. On Tue, Aug 20, 2013

Re: [SR-Users] Writing a module

On 08/20/2013 07:51 AM, Carlos Ruiz Díaz wrote: There's also a Perl module that can be used to execute a Perl script containing your business logic. Sounds like you are doing lots of database queries and doing that from a C module will always be more difficult than using a DB abstraction layer t

Re: [SR-Users] Writing a module

Sure it's a great option but sometimes your business complexity demands not only database queries but remote sockets connection, complex iteration over result sets, doing some regex over the values, etc. Doing this kind of stuff with Perl feels a lot more comfortable. On Tue, Aug 20, 2013 at 7:

Re: [SR-Users] Writing a module

On 08/20/2013 08:01 AM, Carlos Ruiz Díaz wrote: Sure it's a great option but sometimes your business complexity demands not only database queries but remote sockets connection, complex iteration over result sets, doing some regex over the values, etc. Doing this kind of stuff with Perl feels a l

Re: [SR-Users] Ring all strategy

Hello, what version of kamailio are you using? Older versions didn't support variables in append_branch() parameters. You can use xlog() to print $ru before relaying, for troubleshooting reasons. Also, you can set debug=3 for more verbose output. Cheers, Daniel On 8/18/13 10:59 AM, Volkan

[SR-Users] YNT: Ring all strategy

Hi Daniel, Thank you for your reply, I am using version 4.0.2. I have solved the ring all with this piece of code but I am not sure if this is the right way. I keep members of ring group in avp_ring_groups table. xlog("Ring Group config started"); avp_db_load("$ruri/usern

Re: [SR-Users] File transfer using Kamailio

Hello, On 8/17/13 11:12 AM, Nandini madhu wrote: Dear All, I am looking into feature 'file transfer' between my two SIP clients(IMSDroid). With this regards i found two kamailio modules - XMPP and MSRP. I do read its README files and after i stuck with the following queries: Which module i

Re: [SR-Users] avpops, using avp for regexp

2013/8/16 Jon Bonilla : > Any ideas? Here you are [0]. Looking forward for some comments. [0] http://sip-router.org/tracker/index.php?do=details&task_id=333&project=1 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users

Re: [SR-Users] tls.so: mod_register failed on fedora 18 x86_64

On 08/19/2013 03:34 PM, Roberto Fichera wrote: > On 08/19/2013 12:37 PM, Daniel-Constantin Mierla wrote: > > Hi Daniel, > >> Does it work with tls.so as first loaded module? >> >> If not, can you send the logs messages with debug=3? >> >> Cheers, >> Daniel >> >> On 8/19/13 12:26 PM, Roberto Fichera

[SR-Users] mediaproxy-ng one-way audio (but not all the time)

Hello, I am testing mediaproxy-ng (running on CentOS 6 on Amazon EC2) for WebRTC to non-WebRTC calls and I am getting one-way audio most (but not all) of the time. I always get audio in the WebRTC to non-WebRTC direction. Has anybody had any experience of anything this? I have checked the obvio

[SR-Users] Integrating Kamailio 4.x with Freeswitch using Postgresql

Hello All, I have been following bits of this guide http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc and bits from other posts to get Kamailio 4.0 integrated with FusionPBX & Freeswitch 1.2.12 using Postgres instead of MySQL. My reason for using Postgres is simply because Frees

[SR-Users] DB_CLUSTER - Cluster Definition

Hi All, I'm having difficulties with the above module and can't see what I'm doing wrong. Config as follows: modparam("db_cluster", "connection", "db01=>mysql:// > readwrite:passw...@db01.domain.com/kamailio") > modparam("db_cluster", "connection", "db02=>mysql:// > readonly:passw...@db02.domain

[SR-Users] multihomed Kamailio and enable_double_rr

Hi, I'm having a problem with routing of BYEs in my multi homed Kamailio. My setup is a phone on 172.16.230.1, talking to Kamailio on 172.16.230.128. On the "outside" Kamailio uses 10.64.5.16 and its talking to 41.221.230.60 I'm using the stock Kamailio 4.0.3 kamailio.cfg, with: WITH_NAT defin

Re: [SR-Users] multihomed Kamailio and enable_double_rr

Hello, the problem with the BYE is that the R-URI is the ip address of kamailio, resulting in match for strict routing rather than loose routing (both cases are handled by loose_route() function). My guess of what happens is that 41.221.230.60 detects the invite as coming from behind nat and

[SR-Users] Fwd: multihomed Kamailio and enable_double_rr

On 20 Aug 2013 18:49, "Daniel-Constantin Mierla" wrote: > the problem with the BYE is that the R-URI is the ip address of kamailio, > resulting in match for strict routing rather than loose routing (both cases > are handled by loose_route() function). > > My guess of what happens is that 41.221.

Re: [SR-Users] mediaproxy-ng one-way audio (but not all the time)

On 08/20/13 10:56, Peter Dunkley wrote: Hello, I am testing mediaproxy-ng (running on CentOS 6 on Amazon EC2) for WebRTC to non-WebRTC calls and I am getting one-way audio most (but not all) of the time. I always get audio in the WebRTC to non-WebRTC direction. Has anybody had any experience o

Re: [SR-Users] Fwd: multihomed Kamailio and enable_double_rr

On 08/20/2013 02:11 PM, Steve Davies wrote: But I don't quite understand your suggestion that the proxy on 41.221.230.60 should route the INVITE per the Record-Route. The record-route only says what path reply packets should take? Record-Route indicates the path that _sequential requests_ sh

Re: [SR-Users] Fwd: multihomed Kamailio and enable_double_rr

On 20 August 2013 20:13, Alex Balashov wrote: > On 08/20/2013 02:11 PM, Steve Davies wrote: > > But I don't quite understand your suggestion that the proxy on >> 41.221.230.60 should route the INVITE per the Record-Route. The >> record-route only says what path reply packets should take? >> >

Re: [SR-Users] Fwd: multihomed Kamailio and enable_double_rr

On 20 August 2013 20:16, Steve Davies wrote: > Sorry I didn't write what I meant. But the point was that the OpenSIPs on > 41.221.230.60 is not doing any Contact mangling. > > > Oh . I just looked properly and I'm wrong. Kamailio sends the upstream invite with the Contact still @172.160.230.1.

Re: [SR-Users] multihomed Kamailio and enable_double_rr

On 20 August 2013 18:49, Daniel-Constantin Mierla wrote: > If you cannot control 41.221.230.60 or ask for a change there, the > solution is to use htable in your config to store the contact uri from > invite and replace it in bye before loose_route(). > Let me have a go at doing it this way. F

Re: [SR-Users] multihomed Kamailio and enable_double_rr

On 8/20/13 9:08 PM, Steve Davies wrote: On 20 August 2013 18:49, Daniel-Constantin Mierla > wrote: If you cannot control 41.221.230.60 or ask for a change there, the solution is to use htable in your config to store the contact uri from invite and repl

[SR-Users] Call DNS resolver from a cfg script.

Is there a way to run DNS lookups from cfg script? I need some way to get host IP by DNS name. ___ 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

Re: [SR-Users] Call DNS resolver from a cfg script.

On 08/20/2013 04:39 PM, Sergey Okhapkin wrote: Is there a way to run DNS lookups from cfg script? I need some way to get host IP by DNS name. Yes, but only in the development version. Announcement here: http://www.kamailio.org/w/2013/07/new-dns-operations-in-configuration-file/ Docs here: ht

[SR-Users] Problem in call initiate from PC to Mobile phone using sip client thjrough kamailio and rtpproxy

Hi All, I have installed kamailio-4.0.2 and rtpproxy-1.2.1 in same machine on a private ip (10.150.226.160). First I used PC to PC call by using MicroSip as sip client at both end in LAN. Everything running fine.I can make calls and receive calls on both PC. Then I used SAMSUNG Galaxy star duo a