sr-users November 2010

sr-users@lists.kamailio.org

75 participants
111 discussions

VIA header x-route-tag
by alex pappas 01 Dec '10

01 Dec '10

Dear All, We have some cisco gw that they send to the Kamailio in the first invite and inside the VIA header an x-route-tag="The Gateway that the call entered". I'm trying to add this to the acc module in order to have it recorded but without sucess. My config is this: if (search("x-route-tag")) { avp(s:xroutetag) = $sel(@via); xlog("----- We have a x-route-tag in the SIP msg avp: $avp(s:xroutetag) -----"); }; I'm trying to add the VIA header to an avp so I could manipulate the string but i cannot implement this. Does anyone can suggest a way to add the *x-route-tag* value to an AVP? Thank you in advance Alex sample VIA header: Via: SIP/2.0/UDP 10.0.0.15:5060;* x-route-tag="tgrp:TANDEM-KOR"*

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Redirect w/ Xlite?
by Eric Hiller 01 Dec '10

01 Dec '10

Hello all, I have seen some other complaining about this issue, but the posts were over 4 years ago so I would think xlite would be fixed at this point. I wanted to ensure my formatting was correct before I started trying other phones. All I am trying to do at the moment is a simple redirect. My config is: xlog("tu= $tu || $(tU{s.select,0,-}) \n");rewritehost("pbx.newhost.com");xlog("host is now $rd; all is $ru\n");sl_send_reply("300","Redirect");exit; Insanely simple, however xlite is not redirecting the actual ip of the next invite/register/whatever. It only changes the RURI, but not the actual ip. Per the RFC, it looks like I am doing this right, the new host is in the contact field of the 300 and the client should then redirect its next packet to the new host. Any ideas what I am doing wrong here? Or is this a known redirect issue. And given that, how supported is redirect? Should any system be based upon it? Thanks for the help!-Eric

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too many calls, no calls going thru
by Vic Jolin 30 Nov '10

30 Nov '10

Hi ALL, Just wanted to ask why is it that when I have too many calls, there is no call going thru at all? I mean if there are too many calls coming in, I could not make any other call. Is it my config?

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has_sdp() missing
by Jeremya 30 Nov '10

30 Nov '10

Hi, I've been following some examples using has_sdp() function but it is not supplied by 3.0.x TEXTOPS I've seen some discussion on mailing lists about has_sdp() being included since 1.5. But checking the online documentation is doesn't exist in any version. I use has_body("application/sdp") as an alternative but perhaps the has_sdp() function could be included, or references and examples using it altered or removed? Thanks Jeremy

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Re: [SR-Users] SIP between IPv4 and IPv6 agents
by Komáromi Péter 30 Nov '10

30 Nov '10

Hi again, My kamailio 1.5 is working well and I'm able to create the session between an IPv4 and an IPv6 UA, but there is no RTP session. That's the point where the rtpproxy comes in. I installed it, and made a bridging (I hope it is): /usr/local/bin/rtpproxy -F -l /192.168.124.146 -6 FEC0:0:0:0:0:0:0:2 and started rtpproxy -F start. But there is no RTP traffic yet. I read here ( https://www1.ethz.ch/id/people/allid_list/armin/SIP-IPv4-IPv6.pdf ) I have to make two tables named â01Clocation _inet4â01D and â01Clocation_inet6â01D instead of the only â01Clocationâ01D. If it's true, were can I add these? Because it says: "If mySQL is used, the mySQL initialization script (/usr/local/sbin/ser_mysql.sh) has to be modified to create two additional tables" But there is no /usr/local/sbin/ser_mysql.sh on my computer. Where else can I add these tables? Thanks, Peter

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Tutorial: FreeSwitch as Media Server and SBC for Kamailio 3.1
by Daniel-Constantin Mierla 30 Nov '10

30 Nov '10

Hello, a short note to inform that I updated my tutorial about using FreeSwitch and Kamailio together for large VoIP platforms. Besides upgrade to use latest Kamailio major stable release, v3.1.0, there are couple of new features added in the architecture of the VoIP platform: - along with providing media services (voicemail, conferencing, a.s.o.), FreeSwitch is used now also as SBC for topology hiding and media relay (this helps for transconding needs, palying audio messages during the early session or NAT traversal) - Kamailio config has added features to detect DoS/DDoS and scanning attacks, secure SIP communication over TLS, IP authentication and a bunch of neat things that help for an easier maintenance and update of parameters such as database connectivity details, local IPs, etc. The link to tutorial is: * http://asipto.com/u/kfsb Hope it will be useful for many of you. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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rtpproxy_offer() and rtpproxy_answer() missing in 3.0.x
by Jeremya 30 Nov '10

30 Nov '10

http://kamailio.org/docs/modules/3.0.x/modules_k/nathelper.html refers to functions rtpproxy_offer() and rtpproxy_answer() But I get an error when using them in my script. Are these not available in 3.0.x? Regards Jeremy

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Re: [SR-Users] Settings adjustments
by Javier Gallart 30 Nov '10

30 Nov '10

Hi thanks for the reply, I will try with those values... Javi On Mon, Nov 29, 2010 at 6:24 PM, Alex Balashov <abalashov(a)evaristesys.com>wrote: > On 11/29/2010 11:57 AM, Javier Gallart wrote: > > Could these settings cause any "collateral damage"? >> > > Not really. They will result in increased memory usage on a small scale; > they are preprocessor constants that are used to give static sizes to > character arrays inside dynamically allocated structures. But it shouldn't > be a big deal. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 1170 Peachtree Street NE > 12th Floor, Suite 1200 > Atlanta, GA 30309 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ >

1 0

Fwd: Need Clarification on sst module
by Suresh Arunachalam 30 Nov '10

30 Nov '10

Hi Alex, Thanks for your reply, I had same doubt as you, I have double checked the script for exit call. I do not have copy of the configuration to prove myself, for your information I just copied the example from sst module readme file. Thanks Suresh On Mon, Nov 29, 2010 at 9:32 PM, Alex Balashov <abalashov(a)evaristesys.com>wrote: > On 11/29/2010 06:57 AM, Suresh Arunachalam wrote: > > i have configured kamailio server with sst module and server is sending >> 422 error for Min-SE negotiation. Along with 422 response I see 404 >> response coming from the server which is causing my application to >> remove the transaction. Please let know am I missing something in the >> configuration, let me know the fix for the same. >> > > That would be difficult to determine without seeing the configuration. > > My guess is that at the point in the script where the 422 response is sent, > you do not have an 'exit' or 'return' statement, and so the script keeps > executing until it also reaches a 404 stateless reply. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 1170 Peachtree Street NE > 12th Floor, Suite 1200 > Atlanta, GA 30309 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

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Failover with SRV records
by Eric Hiller 30 Nov '10

30 Nov '10

Sorry all for the second question of the night as I go through working out bugs in my setup. I want to use SRV records to loadbalance across hosts. This works great, when all the hosts listed in the SRV are up. However, I want kamailio to use the next host if the current hosts fails. I tried just setting up a new branch, but kamailio keeps using the same SRV entry over and over on that transaction, only a new transaction seems to give kamailio a chance to use a different SRV entry. Any ideas as to how I can force kamailio to try the next SRV entry if the first one fails? For example I t_relay to 2.domain.com which has a _sip._udp.2.domain.com entry for hostA and hostB. If hostB is down, but kamialio decides to send to hostB it just keeps doing so even though hostA is perfectly up it doesn't try for it. Is there a way to remove hostB from the kamailio try list? Or as an alternative to do a manual SRV lookup on 2.domain.com and then put then in a variable and go through them one by one manually? Ideas? Thanks-Eric

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sr-users November 2010