sr-users October 2006

sr-users@lists.kamailio.org

189 participants
346 discussions

[Serusers] memory allocation error
by Lol Zimmerli 27 Oct '06

27 Oct '06

Hello, In my ser.cfg I would like to add a lot of peers in this form: ... if (uri=~"^sip:\*123[0-9]*@.*"){ xlog("L_INFO", "Forwarding *123 to 'other.sip.com': %fu (%ua - %is) -> %tu, %mf\n"); strip(4); rewritehost("other.sip.com"); } ... If I put four of them, everything is okay, ser starts and the calls are correctly redirected. But if I put 1000 of them: Oct 27 02:55:57 ser ser: ERROR:lex:addstr: memory allocation error Oct 27 02:55:57 ser last message repeated 71 times Oct 27 02:55:57 ser ser: ERROR: mk_action: memory allocation failure Oct 27 02:55:57 ser ser: ERROR: mk_action: memory allocation failure Oct 27 02:55:57 ser ser: ERROR:lex:addstr: memory allocation error Oct 27 02:55:57 ser last message repeated 12 times Oct 27 02:55:57 ser ser: ERROR: mk_action: memory allocation failure Oct 27 02:55:57 ser ser: ERROR: mk_action: memory allocation failure Oct 27 02:55:57 ser ser: ERROR:lex:addstr: memory allocation error Oct 27 02:55:57 ser last message repeated 20 times Oct 27 02:55:57 ser ser: ERROR: mk_elem: memory allocation failure Oct 27 02:55:57 ser ser: ERROR:lex:addstr: memory allocation error I tried to change #define SHM_MEM_SIZE 32 into #define SHM_MEM_SIZE 128 in config.h, but same result. Also, in some old doc, a config item called 'shared_mem_size', but adding shared_mem_size=256 in ser.cfg just leads to a Oct 27 03:04:16 ser ser: parse error (30,1-16): syntax error I am quite sure it's just a number to change, but where? Thanks in advance -- # Lol Zimmerli // S y s C o ® // http://www.sysco.ch/ Keep it right when you make it faster. - The Elements of Programming Style (Kernighan & Plaugher)

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[Users] OpenSER summit - registration will be closed Monday, Nov 30
by Daniel-Constantin Mierla 27 Oct '06

27 Oct '06

Hello everybody, the registration for OpenSER summit will be closed Monday, November 30. Afterwards, you can register via VoN, the access will be free for those having access to VoN conference or exhibition, if there is space in the summit room. There are over 80 registered people, about 20 special guests, any many other executives present at VoN were invited to visit us. The agenda is now in final shape (http://www.openser.org/index.php?option=com_content&task=view&id=60&Itemid=…). This was quite a successful campaign, within a short time, and the event promises a lot. In about 3 weeks we managed to have everything in place. Special thanks to sponsors and those which helped to have it possible. Summit page: http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… Cheers, Daniel

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[Users] Problems with Mediaproxy. Bad Port.
by Ricardo Martinez 27 Oct '06

27 Oct '06

Hello list. I'm having a problem with my mediaproxy box. I have a NAT'd client (a softphone) calling to a SIP-H323 conversor through my OpenSER-Mediaproxy box and then to a Cisco Call Manager (it's a very weird configuration). Anyway, the problem presents when the 200 - OK message arrives from my conversor, the SDP part indicate the media address (10.10.148.230) and port (16952) where the remote endpoint will listen RTP, despite of this the mediaproxy open a session for the port 17360. The escenario is something like this. 100.100.148.248 ------- 100.100.148.246 ----- 100.100.154.36 ------ 100.100.154.119 ---- 100.100.148.230 Softphone OpenSer/Mediaproxy Proxy Sip siph323 conversor Call Manager.H323 Please check the log from the mediaproxy and the execution of the "sessions.py" utility, plus the OK message. U 100.100.154.36:5060 -> 100.100.148.246:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.148.246;branch=z9hG4bK67ea.aa12dc97.0. Via: SIP/2.0/UDP 192.168.1.101;received=100.100.148.248;rport=46096;branch=z9hG4bKc0a801650000001d4533a0ee00007c2900000011. Record-Route: <sip:100.100.154.36;ftag=2214787970;lr=on>. Record-Route: <sip:7072408196@100.100.148.246:5060;nat=yes;ftag=2214787970;lr=on>. From: "unknown"<sip:5502408196@sip.desa.mydomain.net>;tag=2214787970. To: <sip:7072408196@sip.desa.mydomain.net>;tag=2260729709. Call-ID: A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101. Cseq: 2 INVITE. Date: Mon, 16 Oct 2006 15:10:37 GMT. Server: siph323. Content-Type: application/sdp. Content-Length: 162. Contact: sip:7072408196@100.100.154.119. . v=0. o=100.100.154.119 1 505533324 IN IP4 100.100.154.119. s=SIP Library call. c=IN IP4 100.100.148.230. t=3370000237 0. m=audio 16952 RTP/AVP 0. a=rtpmap:0 PCMU/8000. Oct 16 12:12:00 sip mediaproxy[18289]: lookup A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101 100.100.148.230:16952:audio 100.100.154.36 sip.desa.mydomain.net local sip.desa.mydomain.net unknown siph323 info=from:5502408196@sip.desa.mydomain.net,to:7072408196@sip.desa.mydomain.net,fromtag:2214787970,totag:2260729709 Oct 16 12:12:00 sip mediaproxy[18289]: execution time: 0.48 ms Oct 16 12:12:00 sip mediaproxy[18289]: session A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101: caller signed in from 100.100.148.248:49154 (RTP) (will return to 100.100.148.248:49154) Oct 16 12:12:00 sip mediaproxy[18289]: session A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101: called signed in from 100.100.148.230:17360 (RTP) (will return to 100.100.148.230:17360) [root@sip ]# /usr/src/mediaproxy-1.7.2/mediaproxy/sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------------------------------------------------------ 100.100.148.248:49154 - 100.100.148.246:60022 - 100.100.148.230:17360 active 0'39" G711u Audio 336.91k/669.53k/332.62k Total traffic: 79.69kbps/78.12kbps/157.81kbps (in1/in2/out) Session count: 1 Proxy version: 1.7.2 Doing more debugs i discover the RTP traffic coming from the 100.100.148.230 it has source port 17360, but the listening port is not the same (i guess is the one indicated in the SDP). Why mediaproxy assumes the 17360 port is for listening too and ommit the SDP 16952 port ? Is this maybe an asymmetric regarding the RTP media?. If so, i added this line into the rtp-asymmetric-clients file: # # Lines starting with a '#' character or empty lines are ignored. # Put a User-Agent name or regular expression per line. # Check is case insensitive. # # This file should only list SIP clients that are asymmetric regarding # the RTP media streams. For clients that are asymmetric regarding the SIP # signaling, use the equivalent SIP version of this file. # Clients that are asymmetric for both SIP signaling as well as RTP media # streams should go in both files. # siph323 But not solve the problem. Can someone help me here? Thanks in advance Regards Ricardo Martinez.- RedVoiss.

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[Users] Q:ABOUT:400 Bad Request RESPONESE
by Alejandro Sanchez 27 Oct '06

27 Oct '06

Hi evreybody My sip server sends this message to the Phone. INVITE sip:192.168.1.10:7191 SIP/2.0 Record-Route: <sip:192.168.1.20;lr=on;ftag=1604fbda> Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bK2f9.039444a7.0 Via: SIP/2.0/UDP 192.168.1.9:7191;branch=z9hG4bK71Fso4gTlg Max-Forwards: 69 Content-Type: application/sdp To: <sip:2109075@192.168.1.20> From: "4422109087" <sip:4422109087@192.168.1.20>;tag=1604fbda Call-ID: 5f1e5301-40b07670(a)192.168.1.9 CSeq: 1596803196 INVITE Allow: INVITE, ACK, CANCEL, MESSAGE, OPTIONS, BYE, UPDATE, REFER, SUBSCRIBE, NOTIFY, SERVICE User-Agent: URMET-UserAgent Contact: "4422109087" <sip:4422109087@192.168.1.9:7191> P-hint: usrloc applied v=0 o=ntzum 16015906 936435728 IN IP4 192.168.1.9 s=- c=IN IP4 192.168.1.9 b=CT:31 t=0 0 m=audio 49648 RTP/AVP 4 96 97 98 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:96 AMR/8000 a=rtpmap:97 X-G723-7200/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-15 m=video 49650 RTP/AVP 99 100 34 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 packetization-mode=1 a=rtpmap:100 H263-1998/90000 a=rtpmap:34 H263/90000 and i have like response a 400 Bad Request. Any hava an idea why? Thank's in Advanced. Alejandro Sanchez ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Serusers] variables in log messages
by Mark Price 26 Oct '06

26 Oct '06

Is there a method to print variables into log messages? For example, something like this: instead of saying: if(method=="SUBSCRIBE") { log(1,"SUBSCRIBE\n"); } do this log(1,method) which obviously doesn't work. It would be very helpful to be able to print out things like headers. Perhaps the select or more likely AVP functions will work in 0.10? Is there anything available now? Mark

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[Users] uri or to_uri
by Gregoire 26 Oct '06

26 Oct '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi the list! I am just wondering the difference between the variables "uri" and "to_uri". I saw that "to_uri" refers to the value of the field "To", but I can not see at which field refers "uri". Does someone know something about that? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFQHp6I8gmGeMTr0sRAgbuAKCmISnqxdAvCQeWTsJjxtNoqoCY8gCeJA8c kW4+/zwvtIJXofKjTTVwrGA= =j0qD -----END PGP SIGNATURE-----

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[Users] BAD REQUEST
by Jose Gil Navarrete 26 Oct '06

26 Oct '06

Hi. My problem is the next. When my sip server send the message INVITE to the phone, the phone always respond with 400 Bad Request, I think is something wrong in the message, in the UDP message the Checksum always is corrupted by the sip server, is it the problem? or is something else in the sip message? How I can correct this? this is the trace: User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 7191 (7191) Source port: 5060 (5060) Destination port: 7191 (7191) Length: 1057 Checksum: 0x879f [incorrect, should be 0xb0cd] Session Initiation Protocol Request-Line: INVITE sip:192.168.1.8:7191 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 9 Message Header Record-Route: <sip:192.168.1.20;lr=on;ftag=db79a2e2> Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bKdfe7.2eeeaa64.0 Via: SIP/2.0/UDP 192.168.1.7:7191;branch=z9hG4bKN2IVzrueyM Max-Forwards: 69 Content-Type: application/sdp To: <sip:2109075@192.168.1.20> From: "4422109087" <sip:4422109087@192.168.1.20>;tag=db79a2e2 Call-ID: eea4130a-277bd7f8(a)192.168.1.7 CSeq: 1503273304 INVITE Allow: INVITE, ACK, CANCEL, MESSAGE, OPTIONS, BYE, UPDATE, REFER, SUBSCRIBE, NOTIFY, SERVICE User-Agent: PHONE-UserAgent Contact: "4422109087" <sip:4422109087@192.168.1.7:7191> P-hint: usrloc applied Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): ntzum 717537891 1011557901 IN IP4 192.168.1.7 Session Name (s): - Connection Information (c): IN IP4 192.168.1.7 Bandwidth Information (b): CT:29 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 49648 RTP/AVP 4 96 97 98 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:96 AMR/8000 Media Attribute (a): rtpmap:97 X-G723-7200/8000 Media Attribute (a): rtpmap:98 telephone-event/8000 Media Attribute (a): fmtp:98 0-15 Media Description, name and address (m): video 49650 RTP/AVP 99 100 34 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:99 H264/90000 Media Attribute (a): fmtp:99 packetization-mode=1 Media Attribute (a): rtpmap:100 H263-1998/90000 Media Attribute (a): rtpmap:34 H263/90000 Regards! --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Users] Mysql table version problem in cvs version
by Pletli Antal 26 Oct '06

26 Oct '06

Hi, I have a problem with openser cvs version. Before the installing the cvs version i dropped the previous openser database. I have installed the new version, and after make dbinstall the "version" table contains the following record for version silo: | silo | 5 | but the debug log shows this problem: 0(0) table_version(): Invalid type (2) or nul (0) version columns for silo 0(0) MSILO:mod_init: Wrong version v-1 for table <silo>, need v5 0(0) init_mod(): Error while initializing module msilo What did I wrong? Thanks in advance, Antal

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[Users] pbxnsip and openser
by Nick De Cristofaro 26 Oct '06

26 Oct '06

Anyone here using pbxnsip as their BB user agent with OpenSER. I'm wondering what steps would be required to make it work. Is it a complicated process? Thanks Nick

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[Users] Replication ACK problem
by Dioris Moreno 26 Oct '06

26 Oct '06

Hi everybody I have a working installation of OpenSER (Proxy A). In this installation, the ACK messages are delivery without any problem through loose_route() and t_relay(). Then, I installed a second OpenSER proxy (Proxy B) and configure both servers to replicate REGISTER messages with t_replicate(), and it works fine. I made a lot of captures to check the replication process was working fine and also to check the NAT pings; each proxy takes care of the NATed clients it receives sending OPTIONS message periodically. I'm using mediaproxy to support NATed clients; to avoid each proxy starting a media session when it find a NATed client I used the following rule: a proxy receiving messages from another proxy never starts a media session, the proxy that sends the message takes care of that. Having two registered clients (Client A in Proxy A and Client B in Proxy B), I start making calls from Client A to Client B, but I found something in the captures. All the messages flow correctly, except the final ACK message sent by Client A to Client B. Proxy A receives this message and pass it to Proxy B, but Proxy B never deliver it to Client B. Of course, the media session never starts. I was searching info about this situation and I found an email respond from Bogdan about using t_newtran() to process ACK messages, I tried, but it didn't work. Has anybody ever had the same problem? Is there a special way to deal with ACK messages when we're working with replicated proxies? I'll appreciate any help.

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sr-users October 2006