sr-users July 2012

sr-users@lists.kamailio.org

64 participants
92 discussions

[Kamailio-Users] BYE message not relayed to caller

by Vikram Ragukumar

Hello, My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy. I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like : No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route: <sip:IPaddr_proxy:7160;lr=on;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" <sip:1234@IPaddr_VoipSwitch>;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0 The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up. Any suggestions on how i can get around this problem ? Thanks and Regards, Vikram.

11 years, 4 months

is kamilio support RFC5626 now ?

by mike

dear kamilio developers : i read the kamilio 3.3.0' s new features , it say it support outbound , and i have place to handle reg-id and sip.instance . so , i wonder if the kamilio3.3.0 support RFC5626 now，so that the kamilio can be used as a edge proxy . i ask why because i can't find where kamilio add flow-token and path for the register request. thanks

11 years, 6 months

kamailio core at qm_status

by Jijo

Hi All, I'm observing a core intermittently at "qm_status (qm=0x786cd000) at mem/q_malloc.c:763" for kamailio version 3.1.0 looking at the backtrace this is occurring while doing a tcp buffer overrun. Please have a look and let me know if anyone observed this issue or how can we debug it. Please find the backtrace and kamailio version osbprod-V2R0:~ # /usr/sbin/kamailio -V version: kamailio 3.1.0 (i386/linux) 21a375 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 15MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 21a375 compiled on 12:38:44 Apr 26 2012 with gcc 4.5.0 #0 qm_status (qm=0x786cd000) at mem/q_malloc.c:763 #1 0x0819c0f3 in qm_debug_frag (qm=0x786cd000, f=<value optimized out>) at mem/q_malloc.c:167 #2 0x0819c352 in qm_free (qm=0x786cd000, p=0x7a6cb6ac, file=0x821cce2 "<core>: tcp_main.c", func=0x822285f "wbufq_run", line=880) at mem/q_malloc.c:457 #3 0x0815f686 in wbufq_run (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:880 #4 handle_tcpconn_ev (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:4141 #5 0x08169bf2 in io_wait_loop_epoll () at io_wait.h:1092 #6 tcp_main_loop () at tcp_main.c:4606 #7 0x080b0404 in main_loop () at main.c:1655 #8 0x080b1f84 in main (argc=9, argv=0xbfb4c9f4) at main.c:2446 (gdb) l 758 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ", 759 "dumping free list stats :\n"); 760 for(h=0,i=0;h<QM_HASH_SIZE;h++){ 761 unused=0; 762 for (f=qm->free_hash[h].head.u.nxt_free,j=0; 763 f!=&(qm->free_hash[h].head); f=f->u.nxt_free, i++, j++){ 764 if (!FRAG_WAS_USED(f)){ 765 unused++; 766 #ifdef DBG_QM_MALLOC 767 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ",

11 years, 6 months

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers

by Samuel Muller

Hello all, I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN. I configured it like this : Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062). Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy. I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. But the audiocodes returns some errors about SIP headers sent by Kamailio : ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected Here you have the example of an INVITE from a SIP phone to the PSTN : ** audiocodes debug ** 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from 77.246.81.132:5060 ---- INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0 To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>> Call-ID: 944d8aec-27503ee6(a)192.168.0.113 CSeq: 102 INVITE Max-Forwards: 49 Contact: "Sam" <sip:0123451010@77.246.81.162:15170> Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 281 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, replaces Content-Type: application/sdp P-Asserted-Identity: <0123451010> Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes v=0 o=- 26933860 26933860 IN IP4 192.168.0.113 s=- c=IN IP4 77.246.81.133 t=0 0 m=audio 35038 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes. When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations. I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ... Someone can confirm or already met the problem ? Many thanks all :) .Sam.

11 years, 7 months

[Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks

by Joe Hart

Hi all, For a project on which I'm currently working, I am having some problems figuring out how to correctly configure Kamailio to communicate with RTP Proxy in order to send media into and out of a network with private IP address ranges. I have a proxy set up to send the SIP traffic, and all of this is working fine. However, I'm having some trouble getting the RTP Proxy set up. Currently, when the call is connected, the offer/answer is made and RTP Proxy seems to be taking over, but I'm having trouble getting my audio to flow in both directions. Examination of the traffic coming into and out of this machine seems to indicate that the IP addresses aren't being mangled correctly. Specifically, it appears the internal IP address isn't being changed to reflect the IP address of the machine on which RTP Proxy is running, so that when the caller tries to send audio back, the IP it's given to reply to is 10.10.x.x, which obviously won't work. I have tried experimenting with specifically setting IP addresses in the rtpproxy_offer() and _answer() methods to no avail, as well as setting various flags in those methods. However, I must admit that I'm not entirely sure what's happening under the hood with these methods, or what rtpproxy is doing with that information when it gets it. Rather than continue to hack at this by trial and error, I'm hoping someone here can point me in the right direction. Any advice, example code or pep talks would be greatly appreciated. Thanks in advance, -- Joe Hart Voice Systems Integrator Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0673

11 years, 7 months

Suggestions for time utils

by Andreas Granig

Hi, I'd like to implement a couple of helper functions for time handling, e.g. checking day of week, day of month etc. from within the kamailio config file. What would you prefer, a new module ("timeutils" maybe?) or adding it to cfgutils (there are already time-based functions there like sleep and usleep)? The idea is to implement time based call-forwards, and an approach could be to provision various time-related values in usr_preferences table, and then check it in the config. For example, when doing call-forwards from Monday to Friday only, I could put this into usr_preferences: attribute: cf_weekday value: [1, 2, 3, 4, 5] (each entry is a separate row in usr_preferences) And in kamailio config, I'd call this: avp_db_load(...); if(is_weekday("$avp(s:cf_weekday)")) { do CF } So is_weekday would iterate over the entries in the avp list and return true if the weekday at the time of routing matches an entry in the list. A module config param could control whether to use gmtime or localtime for matching. Does this make sense? Suggestions for other approaches? I'd rather prefer to do it directly in config instead of using some external interpreter like lua, python etc. Andreas

11 years, 8 months

Possible bad lcr module, after upgrade to 3.3.0 from 3.2

by Alexey Mechanoshin

Hello. After upgrade kamailio and db, lcr module bad routing if entered *any value*to from_uri column in lcr_rule table: mysql> select * from lcr_rule where id=5; +----+--------+--------+-------------------------------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+--------+-------------------------------+-------------+---------+---------+ | 5 | 1 | yyyy | sip:x[xx]\d{2}@xxx.xxx.xx.xx | NULL | 0 | 1 | +----+--------+--------+-------------------------------+-------------+---------+---------+ and load_gw(...) skip this rule, in log: INFO: lcr [lcr_mod.c:1493]: skipping disabled <gw/rule> = <1/5> If remove value from 'from_ui' - rule enabled and all work. Why is turned off rule ? Linux Debian squeeze, upgraded from "deb http://deb.kamailio.org/kamailiosqueeze main" -- "В связи с отсутствием интереса общественности, конец света отменяется" (с) ? С Уважением, Механошин Алексей

11 years, 8 months

Outbound and Registrar. No binding update

by José Luis Millán

Hi, I'm using Kamilio 3.3.0 as registrar server. I`m using an outbound proxy so 'use_path' parameter or 'registrar' module is enabled. According to RFC 5626, a re-registration from a specific combination of AoR, instance_id and reg_id must update the binding. "" If the registrar receives a re-registration for a specific combination of AOR, and instance-id and reg-id values, the registrar MUST update any information that uniquely identifies the network flow over which the request arrived if that information has changed, and SHOULD update the time the binding was last updated. "" In my installation this is not fullfilled as shown: "" AOR:: jmillan Contact:: sip:jmillan@MY_IP;transport=ws;ov-ob=a570655c14 Q= Expires:: 181 Callid:: 1hgq3khalq2rzfr Cseq:: 108 User-agent:: JsSIP 0.1.0 Path:: <sip:OUTBOUND_IP:9090;transport=tcp;lr;ovid=de0c0b9f>,<sip:a570655c14@OUTBOUND_IP :10080;transport=ws;lr;ovid=de0c0b9f;ob> State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:KAM_IP:5060 Methods:: 783 Ruid:: uloc-50119d99-328e-1 Instance:: <urn:uuid:38dce009-ae1f-4fd1-91dc-99ed9affddc1> Reg-Id:: 1 Contact:: sip:jmillan@MY_IP;transport=ws;ov-ob=1dd97b4d51 Q= Expires:: 193 Callid:: 1hgq3khalq2rzfr Cseq:: 110 User-agent:: JsSIP 0.1.0 Path:: <sip:OUTBOUND_IP:9090;transport=tcp;lr;ovid=de0c0b9f>,<sip:1dd97b4d51@OUTBOUND_IP :10080;transport=ws;lr;ovid=de0c0b9f;ob> State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:KAM_IP:5060 Methods:: 783 Ruid:: uloc-50119d99-3290-2 Instance:: <urn:uuid:38dce009-ae1f-4fd1-91dc-99ed9affddc1> Reg-Id:: 1 "" The scenario is such that when the client looses the connection with the Outbound Server, it reconnects and re-registers to Kamailio in order to replace registration and be able to receive in-dialog messages. Any comment is wellcome. Regards,

11 years, 8 months

planning release of v3.3.1

by Daniel-Constantin Mierla

Hello, I am planning to package v3.3.1 out of the latest branch 3.3 this Thursday, August 2. If you are aware of issues that are not listed on the tracker (http://sip-router.org/tracker/), add them there to see what can be sorted out before. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

11 years, 8 months

cannot allocate memory & increase shared memory

by Alex Solt

Hi, We are getting "cannot allocate memory" error in the log file. I was wondering which one is the right way to increase the shared memory: 1) adding the following to /etc/openser/openserctl STARTOPTIONS="-m 128" 2) modify the following in /etc/default/openser MEMORY=1024 Thanks,AS

11 years, 8 months

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sr-users July 2012