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37454 discussions

[Kamailio-Users] Cluecon
by Bogdan-Andrei Iancu 05 Aug '08

05 Aug '08

Hi, For people interested in meeting and chatting around the kamilio - opensips subject, in the next 3 days I will be present at ClueCon, in Chicago. So feel free to drop me an email ;) Best regards, Bogdan

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[Kamailio-Users] I don't know why
by Tseveendorj Ochirlantuu 05 Aug '08

05 Aug '08

Dear developers, I don't know what is the reason they will go to two ways. /---- Kamailio OpenSER \---- openSIPS I hope these two project going to powerful. We need OpenSER related projects. I like them. Best regards, Tseveendorj

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[Kamailio-Users] TLS with openser
by Ali Jawad 04 Aug '08

04 Aug '08

hi all Got some issues with tls my pstn gw does not support tls ..but I am using it between my clients and openser. The traffic is being sent encrypted to the pstn gw. I am using sethostport to forward traffic. any ides ? Thx

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[Serusers] cannot create /tmp/ ser_fifo
by 静于 04 Aug '08

04 Aug '08

Hello all, I am trying to add a user to mysql database with the command: serctl add 1000 1000 1000(a)ser.com but I get a error: cannot create /tmp/ ser_fifo Any pointers to where to start debuging this error? Thanks. Best Regards. Ivy --------------------------------- 雅虎邮箱，您的终生邮箱！

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[Kamailio-Users] INVITE branch handling for multiple contacts.
by Alex Balashov 03 Aug '08

03 Aug '08

Greetings, Is there a place somewhere that expounds upon or otherwise illustrates the inner workings of call forking for multiple concurrent contacts? I can only find a patchwork of hints by perusing the documentation for the 'tm' and 'registrar' modules, and they don't provide me entirely with the understanding that I am looking for. Failing that, I am seeking answers and clarifications to the following general questions. All of them concern handling of an INVITE in a situation where the registrar module allows multiple concurrent contacts, and multiple contacts are in fact registered, and the "append_branches" modparam to "registrar" is set to 1. 1. What is officially supposed to happen if multiple contacts to which the INVITE is relayed all answer the call instantaneously, whether via 183 early media or 200 OK -- but not OOB ringing events? What I get seems a little strange, although perhaps there is an explanation that I am missing. Both far-endpoints send a 200 OK w/SDP and I hear both media streams simultaneously when I call, interleaved by clicks. After a while, the second (most recent) contact's media stream drops off because the UAC decides that the call has timed out because it has not received ACK replies to its 200 OKs; at the same time, OpenSER appears to generate a CANCEL for that second call leg. The CANCEL is replied to with a 200 OK (not a 487 Request Terminated?) although this seems a little bizarre since the dialog's state is already established -- but since the 200 OKs are never replied to, I suppose it is not necessary to receive a BYE in order to terminate that request. Meanwhile, 200 OKs in response to INVITE keep coming from the contact that was slower to pick up and was CANCEL'd (why? this is Asterisk 1.4), but the ACKs from them keep being routed to the first contact (the one that remains), which must be understandably confused as to why they're there although processed as retransmissions. I have a packet capture I'd love to send someone for interpretation, but would rather do it privately off-list. 2. Is there a shared REPLY-ROUTE for replies on each branch? Or is there a way to fragment particular reply routes for each branch? The goal I am ultimately looking to achieve is surely commonplace enough; I would like multiple concurrent registrants to be able to be rung at one request URI. But at present, it seems some very strange things are happening when this occurs that confuse the far-end UACs. That's what governs my intuition that I just need to properly understand how branching is supposed to work. Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] multidomain test for domains 192.168.0.3 and 192.168.0.32
by Zen Kato 03 Aug '08

03 Aug '08

I have installed openser-1.3.2-nontls on 192.168.0.3. The phones are 0103(a)192.168.0.3, 0203(a)192.168.0.3 and 0303(a)192.168.0.3. To test multidomain support, I added modparam("auth_db|auth_diameter|group|uri_db|usrloc","use_domain",1) and changed from "uri==myself" to "is_uri_host_local" in openser.cfg. I added two phones 9101 and 9102 with domain 192.168.0.32. These two phones are registered on mysql db as follows; # openserctl add 9101(a)192.168.0.32 password 9101@localhost # openserctl add 9102(a)192.168.0.32 password 9102@localhost I also set up 192.168.0.32 for eth0:0. eth0 is 192.168.0.3. Then, I check: [root@p4c8 multidomain]# openserctl domain show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd domain_dump 192.168.0.3 192.168.0.32 FIFO command was: :domain_dump:openser_receiver_6329 [root@p4c8 multidomain]# openserctl domain showdb database engine 'MYSQL' loaded Control engine 'FIFO' loaded +----+--------------+---------------------+---------+ | id | domain | last_modified | version | +----+--------------+---------------------+---------+ | 2 | 192.168.0.3 | 2008-04-09 16:13:12 | 0 | | 4 | 192.168.0.32 | 2008-01-01 00:00:00 | NULL | +----+--------------+---------------------+---------+ [root@p4c8 multidomain]# openserctl ul show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=3 max_slot=1 AOR:: 0203(a)192.168.0.3 Contact:: sip:0203@192.168.0.102 Q= Expires:: 295 Callid:: 91741a3a146e34bb(a)192.168.0.102 Cseq:: 101 User-agent:: Grandstream BT110 1.0.8.23 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.0.3:5060 Methods:: 5695 AOR:: 0303(a)192.168.0.3 Contact:: sip:0303@192.168.0.103 Q= Expires:: 294 Callid:: 4c0a9450bb744c33(a)192.168.0.103 Cseq:: 101 User-agent:: Grandstream BT100 1.0.6.8 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.0.3:5060 Methods:: 5695 AOR:: 0103(a)192.168.0.3 Contact:: sip:0103@192.168.0.101 Q= Expires:: 295 Callid:: 60507d20c32ac791(a)192.168.0.101 Cseq:: 101 User-agent:: Grandstream BT110 1.0.8.23 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.0.3:5060 Methods:: 5695 FIFO command was: :ul_dump:openser_receiver_6424 openserctl ul show does not show 9101(a)192.168.0.32 and 9102(a)192.168.0.32 !? [root@p4c8 multidomain]# openserctl ul show 9101(a)192.168.0.32 database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_show_contact location 9101(a)192.168.0.32 404 AOR not found FIFO command was: :ul_show_contact:openser_receiver_6394 location 9101(a)192.168.0.32 What else shoud I set to test multidomain in this situation? Regards, Zen

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[Kamailio-Users] error adding users using openserctl
by muhammad akl 03 Aug '08

03 Aug '08

hey all , i have successfuly built openser-1.2.2-tls from source and till now all things seems to be fine faced a problem with openser_mysql.sh and i have solved it but when i tried to add users to the DB using openserctl i got this message : database engine 'MYSQL' loaded Control engine 'FIFO' loaded /sbin/openserctl: no: not found /sbin/openserctl: no: not found usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] is_user: user counter= Error: user 'zzzz' already exist however i havn't added any users before so any help are more welcomed BR muhammad

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[Serusers] LCR question
by Deepak 03 Aug '08

03 Aug '08

Hi can someone please explain me how LCR works. Specifically, scenario: prefix 1514 handled by gw_1 and gw_2 all other prefixes handled by gw_3 1) Call comes w/ RURI = 1514111111. 2) load_gws is called. In the database, prefix 1514 is supposed to be handled by gw_1 and gw_2; 3) next_gw() is called. RURI is now 1541111111@gw_1 4) call routed to gw_1; gw_1 rejects call 5) we go to failure route and call load_gws() again -- [QUESTION: what happens here?? how does SER know that we need to replace URI w/ 151411111@gw_2 ] 6) next_gws() called. I assume call will go to 1541111111@gw_2 7) gw_2 rejects call. We again go to failure_route. Here again we do a load_gws(). [QUESTION: How does SER know that both gw_1 and gw_2 options have been utilized?? Does load_gws return an error here or does it load gw_3? ] My problem is that in my case, gw_3 is being used after gw_1 and gw_2 have been used. As per my understanding this should NOT happen since prefix 151411111 is not handled by gw_3. Any pointers appreciated.. Running ser 0.9.6

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[Kamailio-Users] Debugging Openser v1.4 scripts
by Jim Burke 02 Aug '08

02 Aug '08

Hi All, We are working to get our script working on Version 1.4 and are having problems debugging the script. Previously if I saw the following error in the logs it would highlight the line and colomb numbers were the error existed. Can someone help out? Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, line 9459656, column 1-1: syntax error Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, line 9459656, column 1-1: bad command!) Jul 29 22:29:18 [19064] ERROR:core:main: bad config file (2 errors) Thanks, Jim

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[Kamailio-Users] T.38 not relayed by RTPproxy
by Jens Carl 02 Aug '08

02 Aug '08

Hello List, I have problem with T.38 and the RTPproxy by one of mine scenarios. The first one works flawless and looks like that: Linksys ATA -> NAT -> OpenSer/RTPproxy -> T.38-GW (Asterisk) The second one on the other hand don't work how it should and looks like this one: T.38-GW -> SIP-Proxy -> Openser/RTPproxy -> NAT -> Linksys ATA. The normal RTP stream is opened and they switch over to T.38, but the RTPproxy isn't sending out the received T.38 packets from the ATA. If the OpenSER is getting a re-INVITE I call the function force_rtp_proxy(). I already tried the function with 'l' parameter but then none of the scenarios are working. Why is the RTPproxy not relaying like in the 1st scenario? Which flag should I use to get the things working? Why is RTPproxy using the same ports like it used for the normal RTP traffic? I thought when I call force_rtp_proxy() a 2nd time I create a new session with different ports? Can I force the RTPproxy to use different ports? If someone needs a PCAP-trace I can provide this. Many thanks Jens

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