[SR-Users] p_usrloc registration update problem
Hi, I'm testing the p_usrloc module (Kamailio version 3.3.2, installed via ubuntu package) and it looks like it's working but for some reason a re-registration is causing a new entry being added to the distributed location table instead of updating the existing one. The kamailio configuration looks like this: #!define DBURL mysql://openser:openserrw@localhost/openser loadmodule p_usrloc.so modparam(p_usrloc, db_mode, 3) modparam(p_usrloc, write_db_url, DBURL) modparam(p_usrloc, read_db_url, DBURL) modparam(p_usrloc, write_on_db, 1) modparam(p_usrloc, alg_location, 1) modparam(p_usrloc, reg_db_table, locdb) mysql select * from locdb; +++---+++-+---++ | id | no | url | status | errors | failover| spare | rg | +++---+++-+---++ | 1 | 1 | mysql://root:secret@172.16.10.131/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | | 1 | 2 | mysql://root:secret@172.16.10.132/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | +++---+++-+---++ When using the default usrloc functionality everything is working fine. The only difference I noticed is that the default usrloc is using MYSQL UPDATE to update the location entry and that p_usrloc is using MYSQL REPLACE. Regards Hans ___ 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] Bug in uac_replace_from if used more than once?
Hello,
you have to use it only once -- this is due to how the changes are done
to the sip message headers, but also because of adding a special
parameter to record-route header. It is its well known behavior since
more than 10 years ago.
Wait until you know what is the final version of the header that should
be sent out and then change it. Before that, you can keep it in a config
variable that can be overwritten easily.
Cheers,
Daniel
On 12/17/12 6:29 PM, Daniel Tryba wrote:
Apparantly it should be simple to mangle the From in request:
http://kamailio.org/docs/modules/stable/modules_k/uac.html#id2525538
The task at hand is use the prefix 131 to suppress callerid for this call:
if($rU=~^131[0-9]+$)
{
strip(3);
append_hf(Privacy: id\r\n);
uac_replace_from(anonymous,sip:anonymous@anonymous.invalid:$rp);
uac_replace_to(,sip:$rU@$rd);
}
Just before this replace there is an other uac_replace_from to rewrite an
alphanumeric username to the default callerid for that device. If there are 2
uac_replace_from in the path, things go wrong like below. A uac_restore_from()
before the second uac_replace_from doesn't change this behavior.
device calls 1310123456789:
U 10.0.34.226:1300 - 10.0.35.226:5060
INVITE sip:1310123456789@kamailio.local:5060 SIP/2.0.
From: sip:anonymous@10.0.3.44:5060;tag=26e5ce5850.
To: sip:1310123456789@kamailio.local:5060.
Kamailio forwards request to dispatcher but fails to properly replace the from
uri:
U 10.0.35.226:5060 - 10.0.32.36:5060
INVITE sip:+31123456789@kamailio.local:5060 SIP/2.0.
From: anonymous sip:
+31987654321@10.0.3.44:5060sip:anonymous@anonymous.invalid:5060;tag=26e5ce5850.
To: sip:0123456789@kamailio.local.
Even though the debug log suggests it did the proper replacement:
uac [replace.c:265]: value to store is is 'sip:anonymous@10.0.3.44:5060' and
len is '30'
uac [replace.c:268]: Storing in FROM-AVP (for use in reply):
'sip:anonymous@10.0.3.44:5060' with len '30'
uac [replace.c:324]: uri to replace [sip:anonymous@10.0.3.44:5060]
uac [replace.c:325]: replacement uri is [sip:anonymous@anonymous.invalid:5060]
uac [replace.c:383]: encode
is=AABQXkFeV15BQUcUXF5AUWxpZDo1MDYw len=48
uac [replace.c:265]: value to store is is
'sip:1310619024127@kamailio.local:5060' and len is '54'
uac [replace.c:271]: Storing in TO-AVP (for use in reply):
'sip:1310123456789@kamailio.local:5060' with len '54'
uac [replace.c:324]: uri to replace [sip:1310123456789@kamailio.local:5060]
uac [replace.c:325]: replacement uri is [sip:0123456789@kamailio.local]
uac [replace.c:383]: encode
is=AAEFAAkGAw0BAANxQlYuERJDCRxeHAoXWgYRXRdbAxtNHxwNCgdACxgubmw6NTA2MA--
len=72
Kamailio version: kamailio 3.3.2 (x86_64/linux)
(the latest stable debian/squeeze package).
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
___
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] From sip phone to provider trunk
Hello, look at uac module with its authentication functions. But be aware of its limitations. A proxy cannot authenticate fully RFC compliant in behalf of users, for that you need a b2bua -- however, there is a chance that works with uac workaround. Cheers, Daniel On 12/17/12 7:14 PM, andre second wrote: Still no success. Shall I use auth module maybe? --- On *Fri, 12/14/12, andre second /andrei.be...@yahoo.com/* wrote: From: andre second andrei.be...@yahoo.com Subject: Re: [SR-Users] From sip phone to provider trunk To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List sr-users@lists.sip-router.org, mico...@gmail.com Date: Friday, December 14, 2012, 9:22 AM Hi. Thanks for you reply! Provider is using Login and Password combination for making calls. --- On *Thu, 12/13/12, Daniel-Constantin Mierla /mico...@gmail.com/* wrote: From: Daniel-Constantin Mierla mico...@gmail.com Subject: Re: [SR-Users] From sip phone to provider trunk To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List sr-users@lists.sip-router.org Cc: andre second andrei.be...@yahoo.com Date: Thursday, December 13, 2012, 11:14 AM Hello, is your trunk provider requiring a username/password for the calls sent to it, or it is just IP based peering? Cheers, Daniel On 12/10/12 4:52 PM, andre second wrote: Hi, I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running. What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that? Thank you! ___ 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 -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda -http://www.linkedin.com/in/miconda -Inline Attachment Follows- ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org /mc/compose?to=sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Support
Hello, yes, kamailio/ser support video calls out of the box, because it does not do media stream processing. Cheers, Daniel On 12/17/12 2:18 PM, Murali Krish wrote: Hi this is Murali krishnan i am final student of master degree in Dalhousie University, I am Working on my project, i m need of a sip server which can support H.264 codec video calls...I created an account in iptel sip account when i used the account it doesn't support video calls. If i use SER will this support video calls? -- /Regards Murali Krish/ ___ 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] dlg_manage() and add_contact_alias() problem
Hello, indeed, checking the sources seems to be an issue for the moment. First is related to how the alias is added -- the contact is updaed by chunks, it should be at once, with a special flag to be visible by other modules (like fixed_nated_contact()). But this is not all, the added aliased parameter has to be handled for sending requests, too. You can open an issue on our tracker and we will take care of it. Cheers, Daniel On 12/17/12 12:47 PM, Pavel Miskov wrote: Hi Daniel, sorry for delay, I was out of office. here are dialog attributes for this to cases, with ngrep output and configuration at the end. Thanks Pavel 1. add_contact_alias() ./kamctl fifo profile_list_dlgs caller dialog:: hash=3965:7641 state:: 4 ref_count:: 2 timestart:: 1355743238 timeout:: 60993875 callid:: d4e9b8c4b17c2f8001704dedccc78e4c@0:0:0:0:0:0:0:0 from_uri:: sip:al...@example.load from_tag:: fe895d80 caller_contact:: sip:alice@10.2.5.112:5060;transport=udp;registering_acc=example_load caller_cseq:: 2 caller_route_set:: caller_bind_addr:: udp:11.22.111.222:5060 callee_bind_addr:: udp:11.22.111.222:5060 to_uri:: sip:b...@testing.load to_tag:: e9648c0277e15011i0 callee_contact:: sip:bob@10.2.4.130:5060 callee_cseq:: 2 callee_route_set:: ngrep output: # U 11.22.111.222:5060 - 10.2.5.112:5060 BYE sip:alice@10.2.5.112:5060;transport=udp;registering_acc=example_load SIP/2.0..Via: SIP/2.0/UDP 11.22.111.222;branch=z9hG4bK2bfc.ace765f3.0..To: sip: al...@example.load;tag=fe895d80..From: sip:b...@testing.load;tag=e9648c0277e15011i0..CSeq: 3 BYE..Call-ID: d4e9b8c4b17c2f8001704dedccc78e4c@0:0:0:0:0:0 :0:0..Content-Length: 0..User-Agent: kamailio (3.3.2 (x86_64/linux))..Max-Forwards: 70 # U 11.22.111.222:5060 - 10.2.4.130:5060 BYE sip:bob@10.2.4.130:5060 SIP/2.0..Via: SIP/2.0/UDP 11.22.111.222;branch=z9hG4bK2bfc.bce765f3.0..To: sip:b...@testing.load;tag=e9648c0277e15011i0.. From: sip:al...@example.load;tag=fe895d80..CSeq: 3 BYE..Call-ID: d4e9b8c4b17c2f8001704dedccc78e4c@0:0:0:0:0:0:0:0..Content-Length: 0..User-Agent: kamai lio (3.3.2 (x86_64/linux))..Max-Forwards: 70 2. fix_nated_contact() --- ./kamctl fifo profile_list_dlgs caller dialog:: hash=3688:11727 state:: 4 ref_count:: 2 timestart:: 1355741746 timeout:: 53323007 callid:: e0e6443a46b8448121e731390ec255e2@0:0:0:0:0:0:0:0 from_uri:: sip:al...@example.load from_tag:: 85acc1ab caller_contact:: sip:alice@3.4.5.6:52556;transport=udp;registering_acc=example_load caller_cseq:: 2 caller_route_set:: caller_bind_addr:: udp:11.22.111.222:5060 callee_bind_addr:: udp:11.22.111.222:5060 to_uri:: sip:b...@testing.load to_tag:: 56c9050dcfebc91i0 callee_contact:: sip:bob@7.8.9.0:5060 callee_cseq:: 2 callee_route_set:: ngrep output: # U 11.22.111.222:5060 - 3.4.5.6:52556 BYE sip:alice@3.4.5.6:52556;transport=udp;registering_acc=example_load SIP/2.0..Via: SIP/2.0/UDP 11.22.111.222;branch=z9hG4bKd9ee.f0980c15.0..To: sip:al...@example.load;tag=85acc1ab..From: sip:b...@testing.load;tag=56c9050dcfebc91i0..CSeq: 3 BYE..Call-ID: e0e6443a46b8448121e731390ec255e2@0:0:0:0 :0:0:0:0..Content-Length: 0..User-Agent: kamailio (3.3.2 (x86_64/linux))..Max-Forwards: 70 # U 11.22.111.222:5060 - 7.8.9.0:5060 BYE sip:bob@7.8.9.0:5060 SIP/2.0..Via: SIP/2.0/UDP 11.22.111.222;branch=z9hG4bKd9ee.01980c15.0..To: sip:b...@testing.load;tag=56c9050dcfebc91i 0..From: sip:al...@example.load;tag=85acc1ab..CSeq: 3 BYE..Call-ID: e0e6443a46b8448121e731390ec255e2@0:0:0:0:0:0:0:0..Content-Length: 0..User-Agent: ka mailio (3.3.2 (x86_64/linux))..Max-Forwards: 70 # U 7.8.9.0:5060 - 11.22.111.222:5060 SIP/2.0 200 OK..To: sip:b...@testing.load;tag=56c9050dcfebc91i0..From: sip:al...@example.load;tag=85acc1ab..Call-ID: e0e6443a46b8448121e731390ec255e2@0 :0:0:0:0:0:0:0..CSeq: 3 BYE..Via: SIP/2.0/UDP 11.22.111.222;branch=z9hG4bKd9ee.01980c15.0..Server: Linksys/SPA942-5.1.15(a)..Content-Length: 0 # U 3.4.5.6:52556 - 11.22.111.222:5060 SIP/2.0 200 OK..To: sip:al...@example.load;tag=85acc1ab..Via: SIP/2.0/UDP 11.22.111.222;branch=z9hG4bKd9ee.f0980c15.0..CSeq: 3 BYE..Call-ID: e0e6443a4 6b8448121e731390ec255e2@0:0:0:0:0:0:0:0..From: sip:b...@testing.load;tag=56c9050dcfebc91i0..Contact: Pero Probna sip:alice@10.2.5.112:5060;transp ort=udp;registering_acc=example_load..User-Agent: Jitsi1.0-build.3967Windows 7..Content-Length: 0 3. configuration is sample configuration in v3.3.2 with addition of dialog part #--- dialog --- modparam(dialog, db_url, mysql://xxx:xxx@localhost/xxx)
Re: [SR-Users] recreating kamilio_fifo without restart
Hello, even if you create it, it is not going to work, because the file descriptor in kamailio points to a different address, the solution is to restart kamailio. Perhaps the -v has to be replaced with something else, it seems to be a common issue. Cheers, Daniel On 12/16/12 2:47 PM, Uri Shacked wrote: Hi, I entered the command kamailio -v instead of kamailio -V. It looks like it starts kamailio again even that it was running the problem is that the kamctl_fifo and kamctl_ctl file are gone now. how do i create the files again without restarting kamailio? Thanks, Uri ___ 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Decode contact with user paramer
When using the siputils function encode_contact() on a message that
contains a user parameter e.g.
sip:X00015;phone-context=natio...@10.yyy.1.92;user=phone;tag=88930
it encodes fine e.g.
sip:natted_ua*sip:X00015;phone-context=national**192.168.242.102*5080*u...@10.yyy.70.100
but decode_contact() errors due to the ; (there is an explicit check for
it in the code)
I am currently working round this using
$rU = $(rU{s.replace,;,:});
to remove the ; as the user part is not needed for my application.
Would it be safe to remove the check for ; ?
Gareth
___
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] Bug in uac_replace_from if used more than once?
On Tuesday 18 December 2012 09:21:24 Daniel-Constantin Mierla wrote: you have to use it only once -- this is due to how the changes are done to the sip message headers, but also because of adding a special parameter to record-route header. It is its well known behavior since more than 10 years ago. It is so well known that it isn't documented. May I suggest to add this to the module usage, would have saved me some time wondering what was happening :) I'll change the config to suggested method. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024 ___ 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] Bug in uac_replace_from if used more than once?
Hello, On 12/18/12 11:44 AM, Daniel Tryba wrote: On Tuesday 18 December 2012 09:21:24 Daniel-Constantin Mierla wrote: you have to use it only once -- this is due to how the changes are done to the sip message headers, but also because of adding a special parameter to record-route header. It is its well known behavior since more than 10 years ago. It is so well known that it isn't documented. May I suggest to add this to the module usage, would have saved me some time wondering what was happening :) it is the public secret :-) Feel free to submit a patch with the improvements you want in the documentation. Wiki is also open for contributions, there you just need to create yourself an account and start a tutorial about this behaviour. Perhaps an entry in FAQ would be good as well. Cheers, Daniel I'll change the config to suggested method. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Planning next minor release - v3.3.3
quick note to say that the packaging will start very soon, any change to branch 3.3 should be announced first on sr-dev. After the announcement with release of v3.3.3 is done, commits can go again directly to branch 3.3 Cheers, Daniel On 12/17/12 8:54 PM, Daniel-Constantin Mierla wrote: Hello, I will package v3.3.3 tomorrow afternoon, please make all wanted commits to branch 3.3 before noon GMT. Cheers, Daniel On 12/11/12 10:32 AM, Daniel-Constantin Mierla wrote: Hello, I am considering releasing next minor version from branch 3.3 at the beginning of next week, Monday or Tuesday (Dec 17 or 18). For outstanding issues, please use the tracker or send to sr-dev mailing list. Developers that pushed fixed in the master branch should backport them in 3.3 before the next week starts. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Kamailio v3.3.3 Released
Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Kamailio v3.3.2 Released
Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Question registering sip providers in kamailio and forward all?
Hi all, I.m. a newbi in Kamailio. My problem: I.m. a blind user and i programmed a voicechat in Freeswitch. 3 have 2 sip providers. vodafone.de sipgate.at and freenet.de. The problem: i must register these proviers in kamailio. And forward all incoming calls to Freeswitch, and outgoing calls from freeswitch to kamailio. Dtmf modes are: auto rfc2833 and inband. Then i call: d...@dorf.blindi.net i like to forward these calls to my freeswitch box. the kamailio ip ist: 217.172.180.108 and the freeswitchbox: 217.172.170.120 Can your help me please? Thanks. --- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc ___ 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] obsoleting modules_s/*radius modules
Hello, according to the archive, modules_s/*radius modules were kept because of duplicate and incompatible auth modules at the time there was an attempt to remove them: - http://lists.sip-router.org/pipermail/sr-users/2010-June/064318.html Now there is a single auth modules, thus I guess the modules_s/*radius can be moved to obsolete. If anyone has different opinion, reply with the arguments here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Video conference with many users
Hi, I want to do web (HTML5 + WebRTC) Sip client which can do the video conference with multiple users. Current release of SIPML http://www.sipml5.org/ does 1 to 1 call. I have no idea of conference with many users. Is it the client that we need to modify to accept call and join the conference ? Do I need to send INVITE with extra parameters ? Please advice me. Best Regards, Roy. ___ 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] NAT Traversal issue
Hi All, Problem solved. It was a CODEC issue. Best Regards, Roy. On Tue, Dec 4, 2012 at 3:58 AM, Raj Roy Ghandhi roy.gan...@gmail.comwrote: Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client-LAN client web page call 2. LAN webrtc client - 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy. ___ 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] Queries
Hi, I have newly installed Siremis sip router. But I have no idea How I could start or How do I start calls? If you can provide me any Manual so it would be great. Sincerely Ankit ___ 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
