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37553 discussions

have problem with iptel account
by Samhu 16 Jul '14

16 Jul '14

hi, This is Sam from Draytek Crop. We have a IPPBX device with iptel for SIP account as below. User B dial to IPPBX could heart the answer from Auto Attendant of IPPBX. Then user B cannot heart anything (after input the extension of user A) when user A pickup his telephone . At this momemt, user A could hear everything from user B. This Network Architecture worked fine before. And it still works fine now if we select the account of "draytel.org". Please show us how to solve the problem. Best Regards, Sam Hu

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SCA and PRESENCE
by Joel White 15 Jul '14

15 Jul '14

I am having an issue where I enable presence modules and the SCA module stops working. I disable presence and SCA works fine on Polycom phones. Does anyone know what I am missing or doing wrong? Thank you in advance Joel

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function to reset all flags or bflags?
by Juha Heinanen 15 Jul '14

15 Jul '14

there is a function to reset a flag or bflag, but i haven't found one that would reset all flags or bflags. is there a way to do it? if not, is it ok to add such function? -- juha

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Many instances of "new URI shorter than old URI" in log
by Alex Villacís Lasso 15 Jul '14

15 Jul '14

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 [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:49 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI I am not really sure of what this message is supposed to indicate. It looks something related to uac_restore_[from,to] functions, but some debugging statements just before the use show that the message is not originating from there. This message is showing up hundreds of times per day.

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set_rtp_proxy_set behaviour
by Igor Potjevlesch 15 Jul '14

15 Jul '14

Hello, I set two entries of rtpproxy like : modparam("rtpproxy", "rtpproxy_sock", "1 == udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "2 == udp:127.0.0.1:7723") For a particular case, I want to use the second RTPProxy and the first one as the default one. If in a conditional block I set set_rtp_proxy("2"), all other call to rtpproxy_manage will use the first RTPProxy? Or there is a load-balancing introduce by default. Regards, Igor.

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Routing calls to Asterisk using dispatch
by Olli Heiskanen 15 Jul '14

15 Jul '14

Hello, I've started playing with an idea to add multiple asterisk servers and using dispatcher to balance the sip load between them. I added the code according to dispatcher module documentation ( http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but I think there's something off in my setup: kamctl ul output shows 2 AORs for one client: AOR:: 770(a)testers.com Contact:: sip:770@2.2.2.2:64340;rinstance=c634da314e12385f;transport=UDP Q= Expires:: 3221 Callid:: ZTE1MWYwYzM3NGNjNjMxMmEzM2JjYWNmNzQyZTdiNGI. Cseq:: 2 User-agent:: Z 3.2.21357 r21367 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 5087 Ruid:: uloc-53bfe447-35ae-2a2 Reg-Id:: 0 Last-Keepalive:: 1405174150 Last-Modified:: 1405174150 AOR:: 770(a)1.1.1.1 Contact:: sip:770@1.1.1.1:5070 Q= Expires:: 68 Callid:: 327fcf07641f80006e962821112a61b5(a)testers.com Cseq:: 754 User-agent:: Asterisk PBX 11.10.2 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 4294967295 Ruid:: uloc-53bfe447-35af-a82 Reg-Id:: 0 Last-Keepalive:: 1405174477 Last-Modified:: 1405174477 I don't think I should be seeing an AOR for 770 where Contact is the public address of my server (here 1.1.1.1) and User-Agent which is Asterisk. I'm using Asterisk Realtime integration, and by what I can tell the sip messages are going nicely, client authenticates with Kamailio and sends this message to Asterisk (which is on the same machine; Kamailio at 5060 and Asterisk at 5070): 1.1.1.1.sip > 1.1.1.1.vtsas: SIP, length: 374 REGISTER sip:1.1.1.1:5070 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKbc8a.f4473947000000000000000000000000.0 To: <sip:770@1.1.1.1> From: <sip:770@1.1.1.1>;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-d0c1 CSeq: 10 REGISTER Call-ID: 7ffa0191-13742(a)1.1.1.1 Max-Forwards: 70 Content-Length: 0 Contact: <sip:770@1.1.1.1:5060> Expires: 3600 Currently I can make calls from 770 to 123 which is an Asterisk extension that answers, plays hello world and hangs up. However I can't call another sip clients when I route calls through Asterisk, they do work fine if I don't use Asterisk for handling calls, but I'd like Kamailio to be in the role of proxy/loadbalancer and Asterisk to handle calls. My config is the simple default config, added with realtime stuff and then dispatcher according to the documentation. I wonder if there's something wrong in the REGISTER that Kamailio sends to Asterisk, or maybe something else going wrong? Has anyone seen results like this and do you spot something here that needs fixing? Thanks, Olli

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siremis/ser/xrcmds error help
by MARY 15 Jul '14

15 Jul '14

sr-users，hello：　 http://127.0.0.1:8890/siremis/ser/xrcmds error info: [2014-07-14 08:49:30 (GMT)] An exception occurred while executing this script: Error message: #2, require_once(XML/RPC.php): failed to open stream: No such file or directory Script name and line number of error: /xxxx/siremis-4.1.0/siremis/modules/ser/service/asipto/libs/cmds/serxr.php:2 why? can not find the XML/RPC.php file。 2014-07-1417:56:16 MARY

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Re: [SR-Users] Forward UAC Registration with Authentication (like Upper Registration or Passthrough Registration)
by Angelica Juarez 14 Jul '14

14 Jul '14

Anjelicajuarez desde mi iPhone

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To Rewrite SDP when request goes from kamailio to outside network
by Anurag Mishra 14 Jul '14

14 Jul '14

Hi All, I have been struggling with this issue for a long and could not find a way to solve it. I have setup FreeSWITCH and Kamailio on the single server but on a private server with a private IP and fronted by firewall. Firewall has all the ports opened. FreeSWITCH is running with nonat. Kamailio has public IP advertised on port 5060 and private IP on 5070 for phones on LAN. Problem is with outbound calls there is no audio and reason I can see is contact field in SDP has the private IP of the server instead of the public IP. Internal device to device calls is working fine. I have tried to rewrite the SDP and have written below code, but got no success: request_route > { > xlog("L_DBG", "Request Route Enter"); > if (is_method("INVITE")) > { > xlog("L_DBG", "Request Route Enter in Invite"); > if (has_body("application/sdp")) > { > xlog("L_DBG", "Request Route Enter in SDP of > Invite"); > if (search("IN IP4 PRIVATE_IP")) > { > xlog("L_DBG", "Request Route Search IN IP4 > PRIVATE_IP"); > fix_nated_sdp("8", "PUBLIC_IP"); > fix_nated_sdp("2", "PUBLIC_IP"); > } > } > } > } > Can anyone help me to get rid of this. Thanks, Anurag Mishra

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Re: [SR-Users] how to remove non-digits using dp_translate
by Henry Fernandes 11 Jul '14

11 Jul '14

I don't know how to apply the global flag in Kamailio. Can you give me an example? -H From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: <miconda(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Date: Thursday, 3 July, 2014 1:47 AM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] how to remove non-digits using dp_translate Hello, you have to replace [^0-9] with empty, using the global flag. Cheers, Daniel On 02/07/14 23:11, Henry Fernandes wrote: > > I'd like to use the dialplan module to remove all non-digits from a string. > For example, "1-306-555-1234" becomes "13065551234" and "(306) 555-1234" > becomes "3065551234". > > > > > Unfortunately, I can't figure out the regular expressions for subst_exp and > repl_exp. I can easily replace one instance of a non-digit, however I can't > figure out how to replace all non-digits. > > > > > -H > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listi > nfo/sr-users > -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda _______________________________________________ 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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