sr-users August 2013

sr-users@lists.kamailio.org

93 participants
177 discussions

Re: [SR-Users] siremis installation
by Daniel-Constantin Mierla 02 Aug '13

02 Aug '13

Hello, have you checked the options to create the database? Do you see any messages in the http server logs? Cheers, Daniel On 8/1/13 11:44 AM, Premchandiran wrote: > > Hi all, > > I am trying to install siremis 4.0.0 in linux, after installation when > I try to login a pop window opens with SQLSTATE[HY000] [2002] No such > file or directory and in > DEBUG.log below line is written > > '08/01/2013','02:33:30','DEBUG','SIREMIS','install module siremis > sql','… [View More]

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Re: [SR-Users] Fix at IMC module
by Daniel-Constantin Mierla 02 Aug '13

02 Aug '13

Hello, thanks for troubleshooting and fix. I would rather change the define, like: #define IMC_MEMBER_SKIP (1<<4) It should be safer for similar cases in future usages. I'll look into the code soon. Cheers, Daniel On 8/2/13 10:40 AM, Shankar wrote: > > Hello, > > While testing group chat via IMC module, we found that owner of a chat > room could not destroy the room or invite new users to a chat room > after few message exchange. > > We had modified the … [View More]

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Fix at IMC module
by Shankar 02 Aug '13

02 Aug '13

Hello, While testing group chat via IMC module, we found that owner of a chat room could not destroy the room or invite new users to a chat room after few message exchange. We had modified the code at imc_cmd.c to fix this issue. Fix, member->flags &= ~IMC_MEMBER_SKIP; à member->flags &= ~(IMC_MEMBER_SKIP); where ~IMC_MEMBER_SKIP is defined as #define IMC_MEMBER_SKIP 1<<4. Here ~ takes precedence over the << operator. With this … [View More]

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Asterisk and Kamailio as a sip router
by Rafael Carvallo 02 Aug '13

02 Aug '13

Hey everyone, currently i have this configuration set on kamailio: modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 0) #forzado de la reescritura direccion de destino modparam("dispatcher", "flags", 2) #banderas de funcionamiento, 2 significa "soporte para failover" modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher",… [View More] "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 5) #tiempo que transcurre antes de verificar nuevamente una salida inactiva modparam("dispatcher", "ds_probing_threshhold", 5) #Numero de intentos antes de marcar una salida como inactiva modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) #loadmodule "dispatcher.so" ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { if (is_method("SUBSCRIBE")){ if (src_ip == 192.168.2.1 | src_ip == 192.168.2.2){ t_relay(); } route(REGISTRAR); } if (src_ip == 192.168.2.1 | src_ip == 192.168.2.2){ t_relay(); } else{ route(ASTERISK); } } route[ASTERISK]{ ds_select_dst("1", "8"); t_relay(); exit(); } Everything else is left with the same basic configuration it had when i installed the software. I want to use it kinda of a sip router, so what Kamailio does is just forward the packets betwen my Asterisk boxes and the Sip Phones. Currently it seems to work (partially) but i have doubts about if this is correctly done (it's the first time using Kamailio and i need this working withing a week at most). As you can see i use the module dispatcher for failover/failback (this is the purpose of using Kamailio, a failover/failback setup). One major problem i've found with this setup is, if the phones are currently connected and working with one of my 2 asterisk boxes and if that box fails, Kamailio starts sending the traffic to the second box (as intended), but the phones don't try to subscribe to the new asterisk box, rather they just keep sending traffic (and obviously kamailio forwarding it). Sometimes one of the phones subscribe to the new box, but that's not always the case, the packets reach the new Asterisk box, but since the phones aren't registered to it, they can't make calls. Other times the behavior of the setup is rather weird, i can call some extensions and some others i can not (even though they are registered within the asterisk box) The traffic gets to the Asterisk box (as shown in the asterisk logs) but the call is shown as "service unavailable". I've checked a lot of times the Asterisk setup and it seems to be fine so i think it has something to do with Kamailio. I know this might be a really bad configuration file, but it's been at most 3 days since i started using Kamailio which i sometimes find kinda hard to understand and i really need this working within a week. To summarize all i want is kamailio forwarding packets between the currently active server and the phones so if it fails, then the packets go to the second one, the phones must re-subscribe to the new active server. [View Less]

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getting error db_mysql/kamailio does not export db_use_table function
by Rama Shankar 02 Aug '13

02 Aug '13

Hi, We are trying to install and run kamailio for video call , we have installed it successfully . When we are running Kamailio without mysql support it gets start and we are able to make video call with help of soft phone. But when we are running it with mysql support it gets exit with following error , Please suggest me where I'm doing mistake. db_check_api(): module db_mysql/kamailio does not export db_use_table function Kind regards Rama Shankar DISCLAIMER AND PRIVILEGE NOTICE: This … [View More]

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XCAP client to test org.openmobilealliance.user-profile
by Shankar 02 Aug '13

02 Aug '13

Hello, Anyone aware of free/opensource client which can be used to test org.openmobilealliance.user-profile of xcap server in kamailio module. Please help. Regards, Shankar

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RTPProxy doesn't change port in T38 ReINVITE
by Федор Косилов 01 Aug '13

01 Aug '13

Hello everyone, I'm experiencing the problem described in subject. My UAC is Vega Europa 50 with t38 enabled, and it sends different port numbers for audio in initial request and for image in t38 reinvite, but when my script calls force_rtp_proxy(), rtpproxy replies with same port numbers for both requests. My script routes requests to Asterisk box, and when my Vega Europa starts t38 rtp, Asterisk doesn't treat this traffic as a valid udptl. Here is a dialog between Vega and Kamailio http://… [View More]

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Re: [SR-Users] Siremis 4.0. login issue
by Salman Zafar 01 Aug '13

01 Aug '13

Make sure you are preparing and getting the apache config by using make command in siremis folder. PS: Please do not create new email threads on each reply. On Thu, Aug 1, 2013 at 6:21 PM, Premchandiran < premchandiran.marimuthu(a)plintron.com> wrote: > Hi Salman , **** > > I tried the #a2enmod rewrite and restarted the apache2 still the same > issue . **** > > I changed AllowOverride All to AllowOverride None since I was getting 43 > forbidden . **** > > … [View More]

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mediaproxy-ng: error rewriting SDP during a video call
by Alexey Rybalko 01 Aug '13

01 Aug '13

Hi! Few days ago I was lucky to establish calls between Chrome and SIP UA. Thanks to new rtpproxy developers! That was for audio only because many UAs lack for VP8 support. To verify a video I tried to involve Jitsi into the tests. Mediaproxy can't rewrite answer SDP from Jitsi. Probably the issue is related to SDP but I'm not sure to which part in particularly. *Original offer SDP (ICE attributes are skipped) * v=0 o=- 6784329718950523193 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio … [View More]video a=msid-semantic: WMS 0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP m=audio 59996 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126 c=IN IP4 10.xx.xx.xx a=rtcp:59996 IN IP4 10.xx.xx.xx .... a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VZhAArr96fxUtzvuNPpE+OSa09wchsmTD9pdrdBv a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 CN/48000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:579751405 cname:n2gMzhLDYdqpWwP7 a=ssrc:579751405 msid:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP 0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpPa0 a=ssrc:579751405 mslabel:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP a=ssrc:579751405 label:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpPa0 m=video 59996 RTP/SAVPF 100 116 117 c=IN IP4 10.xx.xx.xx a=rtcp:59996 IN IP4 10.xx.xx.xx ... a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=sendrecv a=mid:video a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VZhAArr96fxUtzvuNPpE+OSa09wchsmTD9pdrdBv a=rtpmap:100 VP8/90000 a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 goog-remb a=rtpmap:116 red/90000 a=rtpmap:117 ulpfec/90000 a=ssrc:1190198390 cname:n2gMzhLDYdqpWwP7 a=ssrc:1190198390 msid:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP 0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpPv0 a=ssrc:1190198390 mslabel:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpP a=ssrc:1190198390 label:0dKFIAlsVMexmIc0qjieYgfPDvBnSLG5lmpPv0 *Original answer SDP* v=0 o=user2 0 0 IN IP4 10.xx.xx.xx s=- c=IN IP4 10.xx.xx.xx t=0 0 m=audio 5052 RTP/AVP 0 8 126 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 m=video 5054 RTP/AVP 100 a=rtpmap:100 VP8/90000 *Mediaproxy logs* (for answer SDP) mediaproxy-ng[14907]: Got valid command from 127.0.0.1:38310: answer .... mediaproxy-ng[14907]: [7dmpb0qdja07p87pa1ji - ] Got LOOKUP, but no usable callstreams found mediaproxy-ng[14907]: Error rewriting SDP mediaproxy-ng[14907]: Protocol error in packet from 127.0.0.1:38310: Error rewriting SDP:... However the *audio calls work well*. Here's the answer SDP from Jitsi: v=0 o=user2 0 0 IN IP4 10.xx.xx.xx s=- c=IN IP4 10.xx.xx.xx t=0 0 m=audio 5048 RTP/AVP 0 8 126 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 Comments and suggestions would be greatly appreciated. regards, Alexey [View Less]

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Two servers with shared registration database
by Eugene Prokopiev 01 Aug '13

01 Aug '13

Hi, Is it possible to configure two Kamailio servers with shared registration database? Any SIP UA can be registered on any server but it must transparently call another SIP UA which can be registered on the same or on another Kamailio server. Is where any example of this configuration? -- WBR, Eugene Prokopiev

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sr-users August 2013