sr-users May 2004

sr-users@lists.kamailio.org

154 participants
333 discussions

[Serusers] Problem with REGISTER in cvs version.
by Luis Vazquez 06 May '04

06 May '04

hi all, This is my first message, even I have been reading and learning from this great list for a while. I used the stable branch of SER for 6 months without problems, now I'm playing with the NAT an rtp proxy stuff so I upgraded to the cvs branch. I downloaded, compiled and installed cvs (2004-04-27) with mysql support. After solving some compile problems (fixed in a patch by janakj in the cvs two hours after my first checkout :( I got it running but there is a problem I don't know how to solve. I'm trying to register and call a SIP phone from behind NAT. I get it to register correctly (with the Contact field fixed) but when I try to call it, the lookup("location") fails and I get a "480 Temporarily Unavailable" from ser. The network schema I'm using is (for testing): 192.168.10.* 200.59.203.* [Phone Gstream] .188 ------------>.1 [NAT].75 ----------------- | | .76 .77 [SER] [SIP UA (Asterisk)] I added a gstream1 user to SER and the register block in ser.cfg is: ...... # -- nathelper -- if (nat_uac_test("1")) { if (method == "REGISTER" || !search("^Record-Route:")) { log(2,"LOG: Registro desde una IP privada, re-escribiendo contacto...\n"); # Rewrite contact with source IP of signalling fix_nated_contact(); if (method == "INVITE") { ............. }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; ............. if (uri==myself) { if (method=="REGISTER") { # No auth by now log(2,"Saving url location\n"); save("location"); break; }; ..... The ngrep SIP log for the register sequence is: U 200.59.203.75:61503 -> 200.59.203.76:5060 REGISTER sip:200.59.203.76 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.188:5062;branch=z9hG4bK975f3a4817e63645 ..From: "Grandstream 1" <sip:gstream1@200.59.203.76>;tag=f001e96a008a26a5..To: <sip:gstream1@200.59.203 .76>..Contact: <sip:gstream1@192.168.10.188:5062>..Call-ID: 0776eb9b4143e102@192.168.10.188..CSeq: 102 REGISTER..Expires: 3600..User-Agent: Grandstream BT100 1.0.4.55..Max-Forwards: 70..Allow: INVITE,ACK,CA NCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 200.59.203.76:5060 -> 200.59.203.75:61503 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.188:5062;branch=z9hG4bK975f3a4817e63645;rport=61503;receive d=200.59.203.75..From: "Grandstream 1" <sip:gstream1@200.59.203.76>;tag=f001e96a008a26a5..To: <sip:gstr eam1(a)200.59.203.76>;tag=b27e1a1d33761e85846fc98f5f3a7e58.ce3d..Call-ID: 0776eb9b4143e102(a)192.168.10.188 ..CSeq: 102 REGISTER..Contact: <sip:gstream1@200.59.203.75:61503<>;q=0;expires=3600..Server: Sip EXpres s router (0.8.13-dev-28 (i386/linux))..Content-Length: 0.... Everything seems Ok, but here begins the problems. If I go to the command line and type: # serctl ul show gstream1 I get: 404 Username gstream1(a)ipcontact.com.uy in table location not found But if I type: # serctl ul show Among many other lines I get: ..... ...Record(0x402f46e0)... domain: 'location' aor : 'gstream1(a)200.59.203.76' ~~~Contact(0x402f47c0)~~~ domain : 'location' aor : 'gstream1(a)200.59.203.76' Contact: 'sip:gstream1@200.59.203.75:61503' Expires: 3083 q : 0 Call-ID: '0776eb9b4143e102(a)192.168.10.188' CSeq : 102 replic : 0 State : CS_SYNC Flags : 1 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ............. It seems to my like ser is not matching the domain part properly. On the other side, the record in the mysql location table is: username | domain | contact | expires | q | .... gstream1(a)200.59.203.76 | | sip:gstream1@200.59.203.75:61503 | 2004-05-04 17:20:32 | -1082130.38|... As you see 'domain' field is empty. Could be this the problem? Anyway If I register from Kphone (user luis) running in the same machine as SER, 'domain' is empty again luis(a)ipcontact.com.uy | |sip:luis@200.59.203.76:5062;transport=udp|2004-05-04 16:57:46|1148846.12|... and # serctl url show luis fails too, however I'am able to call this user "luis(a)ipcontact.com.uy" but not "gstream1(a)ipcontact.com.uy". The problem here seems to be gstream1 is registered using the ser server's IP number and not the domain "ipcontact.com.uy", but shouldn't be this handled automagicaly by SER. I don't have any hint to the serctl ul "misterio" (if fails even if I pass the domain or the ip after the user name?) One diference I see with the 0.8.12 stable version I'm using in "production" is after a REGISTER only the username part of the uri is recorded in the 'username' field of table location!! Column 'domain' is empty int this case too, but serctl works. May be this is some king of buggy 'feature' in the cvs? Any hint? Oh, just in case, this is the failed INVITE sequence: U 200.59.203.77:5060 -> 200.59.203.76:5060 INVITE sip:gstream1@ipcontact.com.uy SIP/2.0..Via: SIP/2.0/UDP 200.59.203.77:5060;branch=z9hG4bK186cfac a..From: "SIP Planet(1083694723.15)" <sip:192@ipcontact.com.uy>;tag=as16156ce5..To: <sip:gstream1@ipcon tact.com.uy>..Contact: <sip:192@200.59.203.77>..Call-ID: 653214bf775a279776bc5c2360100a13(a)200.59.203.77 ..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Tue, 04 May 2004 18:18:45 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..UniqueID: 1083694725.16..Content-Type: application/sdp..Content-Length: 21 1....v=0..o=root 16321 16321 IN IP4 200.59.203.77..s=session..c=IN IP4 200.59.203.77..t=0 0..m=audio 12 692 RTP/AVP 0 3 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off U 200.59.203.76:5060 -> 200.59.203.77:5060 SIP/2.0 480 Temporarily Unavailable..Via: SIP/2.0/UDP 200.59.203.77:5060;branch=z9hG4bK186cfaca..From: "SIP Planet(1083694723.15)" <sip:192@ipcontact.com.uy>;tag=as16156ce5..To: <sip:gstream1@ipcontact.com. uy>;tag=b27e1a1d33761e85846fc98f5f3a7e58.251a..Call-ID: 653214bf775a279776bc5c2360100a13(a)200.59.203.77. .CSeq: 102 INVITE..Server: Sip EXpress router (0.8.13-dev-28 (i386/linux))..Content-Length: 0 U 200.59.203.77:5060 -> 200.59.203.76:5060 ACK sip:gstream1@ipcontact.com.uy SIP/2.0..Via: SIP/2.0/UDP 200.59.203.77:5060;branch=z9hG4bK186cfaca.. From: "SIP Planet(1083694723.15)" <sip:192@ipcontact.com.uy>;tag=as16156ce5..To: <sip:gstream1@ipcontac t.com.uy>;tag=b27e1a1d33761e85846fc98f5f3a7e58.251a..Contact: <sip:192@200.59.203.77>..Call-ID: 653214b f775a279776bc5c2360100a13@200.59.203.77..CSeq: 102 ACK..User-Agent: Asterisk PBX..Content-Length: 0.... Thank you very much in advance. Best regards Luis

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[Serusers] PSTN rerouting bug handling CANCEL message
by Raymond Chen 06 May '04

06 May '04

Dear all, I am testing SER CVS head for PSTN rerouting if the first route has failed like example onr.cfg. the problem is the caller cancels the call before the call is pick up but SER didn�t properly terminate it, instead it sends to the second route and the third. It ends up the callee rings multiple times. It didn�t happen in the stable release 0.8.12, but we would like to usualize the new mysql module connection pool in 0.8.13 cvs head. Any one has a solution? U 202.123.123.123:5060 -> 211.123.123.124:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 202.123.123.124;branch=z9hG4bK1908.c8b2c55.0,SIP/2.0/UDP 202.123.123.125:5060.. From: <sip:34293057@202.123.123.125>;tag=C2B5948-13C9..To: <sip:0085298637212@202.123.123.124>..Date: Tue, 04 May 2004 10:48:14 GMT..Call-ID: 971874CF-C0F011D3-8AFA976C-D4B6780B@202.123.123.125..Timestamp: 946888968..Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled..CSeq: 101 INVITE..Content-Type: application/sdp..Session: Media..Content-Length: 140....v=0..o=CiscoSystemsSIP-GW -UserAgent 5444 359 IN IP4 202.123.123.123..s=SIP Call..c=IN IP4 202.123.123.123..t=0 0..m=audio 20150 RTP/AVP 18.. # U 202.123.123.124:5060 -> 202.123.123.125:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 202.123.123.125:5060..From: <sip:34293057@202.123.123.125>;tag=C2B5948-13C9..To : <sip:0085298637212@202.123.123.124>..Date: Tue, 04 May 2004 10:48:14 GMT..Call-ID: 971874CF-C0F011D3-8AFA976C-D4B6780B(a)211.14 7.234.150..Timestamp: 946888968..Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled..CSeq: 101 INVITE..Content-Type: application /sdp..Session: Media..Content-Length: 140....v=0..o=CiscoSystemsSIP-GW-UserAgent 5444 359 IN IP4 202.123.123.123..s=SIP Call..c =IN IP4 202.123.123.123..t=0 0..m=audio 20150 RTP/AVP 18.. # U 202.123.123.125:59047 -> 202.123.123.124:5060 CANCEL sip:0085298637212@202.123.123.124:5060 SIP/2.0..Via: SIP/2.0/UDP 202.123.123.125:5060..From: <sip:34293057@211.147.234. 150>;tag=C2B5948-13C9..To: <sip:0085298637212@202.123.123.124>..Date: Mon, 03 Jan 2000 08:42:48 GMT..Call-ID: 971874CF-C0F011D3 -8AFA976C-D4B6780B@202.123.123.125..CSeq: 101 CANCEL..Max-Forwards: 6..Timestamp: 946888970..Content-Length: 0.... # U 202.123.123.124:5060 -> 202.123.123.123:5060 CANCEL sip:600085298637212@202.123.123.123:5060 SIP/2.0..Record-Route: <sip:202.123.123.124;ftag=C2B5948-13C9;lr=on>..Via: SIP/ 2.0/UDP 202.123.123.124;branch=z9hG4bK1908.c8b2c55.0..Via: SIP/2.0/UDP 202.123.123.125:5060..From: <sip:34293057@211.147.234.1 50>;tag=C2B5948-13C9..To: <sip:0085298637212@202.123.123.124>..Date: Mon, 03 Jan 2000 08:42:48 GMT..Call-ID: 971874CF-C0F011D3- 8AFA976C-D4B6780B@202.123.123.125..CSeq: 101 CANCEL..Max-Forwards: 5..Timestamp: 946888970..Content-Length: 0..P-Hint: GATEWAY. ... # U 202.123.123.124:5060 -> 202.123.123.125:5060 SIP/2.0 200 cancelling..Via: SIP/2.0/UDP 202.123.123.125:5060..From: <sip:34293057@202.123.123.125>;tag=C2B5948-13C9..To: <sip :0085298637212@202.123.123.124>;tag=8ade010e448416c7b1e15c5e4aff46dd-62a5..Call-ID: 971874CF-C0F011D3-8AFA976C-D4B6780B(a)211.147 .234.150..CSeq: 101 CANCEL..Server: Sip EXpress router (0.8.13-dev-29 (i386/linux))..Content-Length: 0..Warning: 392 211.147.23 4.146:5060 "Noisy feedback tells: pid=2909 req_src_ip=202.123.123.125 req_src_port=59047 in_uri=sip:0085298637212@211.147.234. 146:5060 out_uri=sip:600085298637212@202.123.123.123:5060 via_cnt==1".... # U 202.123.123.123:5060 -> 202.123.123.124:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 202.123.123.124;branch=z9hG4bK1908.c8b2c55.0,SIP/2.0/UDP 202.123.123.125:5060..From: <sip:342 93057(a)202.123.123.125>;tag=C2B5948-13C9..To: <sip:0085298637212@202.123.123.124>..Date: Tue, 04 May 2004 10:48:16 GMT..Call-ID: 971874CF-C0F011D3-8AFA976C-D4B6780B@202.123.123.125..Timestamp: 946888970..Content-Length: 0..CSeq: 101 CANCEL.... # U 202.123.123.124:5060 -> 211.123.123.128:5060 INVITE sip:901201185298637212@63.192.114.66:5060 SIP/2.0..Record-Route: <sip:202.123.123.124;ftag=C2B5948-13C9;lr=on>..Via: SIP /2.0/UDP 202.123.123.124;branch=z9hG4bK1908.c8b2c55.2..Via: SIP/2.0/UDP 202.123.123.125:5060..From: <sip:34293057@211.147.234. 150>;tag=C2B5948-13C9..To: <sip:0085298637212@202.123.123.124>..Date: Mon, 03 Jan 2000 08:42:48 GMT..Call-ID: 971874CF-C0F011D3 -8AFA976C-D4B6780B@202.123.123.125..Supported: timer,100rel..Min-SE: 1800..Cisco-Guid: 2484681635-3236958675-2331481964-356872 8075..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 5..Remote-Party-ID: <sip:34293057@202.123.123.125>;party=calling;screen=yes;privacy=off.. Timestamp: 946888968..Contact: <sip:34293057@202.123.123.125:5060>..Expires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: 328..P-Hint: GATEWAY....v=0..o=CiscoSystemsSIP-GW-UserAgent 5757 2194 IN IP4 202.123.123.125.. s=SIP Call..c=IN IP4 202.123.123.125..t=0 0..m=audio 17774 RTP/AVP 18 4 0 8 19..c=IN IP4 202.123.123.125..a=rtpmap:18 G729/8000 ..a=fmtp:18 annexb=yes..a=rtpmap:4 G723/8000..a=fmtp:4 annexa=no..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:19 CN/80 00.. Thanks Raymond __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

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[Serusers] Timer Scenario.
by Atle Samuelsen 06 May '04

06 May '04

Hey Guys! First, thanks for a great mailinglist! How is it possible to either set maximum Reinvites when there is no "reply" from the device, Im am using RecordRouteing..? The reson why I want to do this is because If a user is temporary offline, I want to push him QUICK to a announcement. Anybody had this scenario before? and got an Idea on how to do this? I had 2 toughts, 1: Count invites, and set a flag, if f.eks some reply from phone etc, it will set counter to 0 again, and start looking for reinvites. This idea died because SER does'nt show me the reinvites it sends. 2: Use modparam("tm", "retr_timer2", 8), Tho this did'nt work for some werid reson. - Atle

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[Serusers] Voice loss after establishing call
by Arnd Vehling 06 May '04

06 May '04

Hello, we currently experience "incoming voice loss" after an outgoing call has been established. i.e. after the remote party picks up the call you dont hear anything for 1-2 seconds which results in both sides of the connection saying repeatedly "Hello" until both sides can hear each other. This phenomenon occurs with nated and non natted clients so it doesnt seem to be related to "rtpproxy" or other NAT problems. It occurs between SIP<>SIP and SIP<>PSTN Calls (via a cisco gw) independantly of the UA type used. Has anyone an idea where to look for the bug/where to start debugging? best regards, Arnd

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[Serusers] SIP DIALER
by Nair 06 May '04

06 May '04

Please provide information regarding RFC 3261 compatible SIP dialer. --------------------------------- Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs

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[Serusers] Ringback tone
by GR S 06 May '04

06 May '04

Folks, Sorry to bother you all with such a silly question, but I am facing some issues with ringback tones. I am using SJphones with SER. They work perfectly well, I can make calls to other extensions and talk. I am maintaining a list of valid users in a database table, and have a small module which checks the authennticity of the dialed extension when SER receives INVITEs, and if it is valid, the call is t_relayed to the called extension. My problem is when an extension is dialed, I cannot hear the ringing tone at the callers end. I searched on the archives, but could not find mails addressing such a problem. Obviously I am missing something here. Can anyone help? Thanks for your time... Regards, Girish __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

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[Serusers] isdngw version 0.2 available
by Ulrich Abend 06 May '04

06 May '04

Hi, I am happy to announce, that the isdngw reached beta state, thus I want to encourage everyone interested in testing the PSTN gateway. isdngw is a PSTN2SIP and SIP2PSTN gateway included in SEMS. It runs on Linux and requires only a supported ISDN card (BRI or PRI). A new site for isdngw was launched, too. Find installation instructions and a list of supported cards there: http://www.iptel.org/isdngw Any feedback, may it be bug reports, information on ISDN controllers, any other comments or success stories ;-) are very much appreciated. Regards, Uli.

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[Serusers] DSTATE option in Makefile
by Quenton 06 May '04

06 May '04

Hi i have a short question about the compilation of ser, i want to complie with DSTATE option , normaly i run: make modules make install prefix ="...." How can i declare the DSTATE option during the installation, I take a look in the makefile.defs where I found this flag and remove the comment in front of DSTATE. But it doesn't work . Thanks in advance Quenton

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[Serusers] Re: [Sems] ivr.record and DTMF tone
by gaillac harry 06 May '04

06 May '04

Hi Richard Sorry I can't answer but I think we are looking for the sames problems do you solve theses question could you help me ? I have a few questions about multiple domains and call from/to pstn. I don't want to change the ser.cfg script everytime a new domain is added. How can I support the following features? 1. Every domain has their own 3 or 4 digit dial plan. When they call PSTN, their caller id should be the either a prefix plus their internal number or a generic receptionist number, but not the 3/4 digit internal number. Different domains have different prefix. Is there a way to do this without changing the script for every domain? 2. Should interdomain calls be prohibited? Is it preferred to send the call to PSTN and hairpin back? I know it consumes more resources but would it provide better privacy? If that's the case, how to prevent interdomain calls, i.e. how to verify from and to URI have the same host parts? 3. When PSTN calls come in, should I use prefix2domain in pdt module to convert the PSTN number to a local domain number? Any pitfalls? This list has been great! Thanks for all your help. Le jeu 06/05/2004 à 05:05, Richard a écrit : > Hi, > > Is there a way to record the message and also detect > DTMF tones? For example, after the recording, user can > press # to stop. Then he can play and re-record it if > necessary. This would be very useful to record > personalize greetings. > > Thanks, > Richard > > > > > > __________________________________ > Do you Yahoo!? > Win a $20,000 Career Makeover at Yahoo! HotJobs > http://hotjobs.sweepstakes.yahoo.com/careermakeover > > _______________________________________________ > Sems mailing list > sems(a)lists.iptel.org > http://lists.iptel.org/cgi-bin/mailman/listinfo/sems

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[Serusers] Memory error
by Mook 06 May '04

06 May '04

Thanks, Jiri. When I do these changes, got the following message. 0(6096) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: fm_malloc 0(6096) parse error (31,13-49): failed to load module ERROR: bad config file (1 errors) Mook. At 09:47 AM 5/6/2004, Mook wrote: >I'm running a Linux red Hat 9.0 with all updates, I have 512 MB ram, ser 0.8.12 . >I writes a module and compiles it fine.When ser receive invite request, this module will be executed. >Then I gets the following error message. > >May 6 15:11:58 nw127 /usr/local/sbin/ser[3154]: Error: new_module: memory allocation error >May 6 15:11:58 nw127 /usr/local/sbin/ser[3154]: Error: new_module: execute failed >May 6 15:11:58 nw127 /usr/local/sbin/ser[3154]: ERROR: do_action: memory allocation failure >May 6 15:11:58 nw127 /usr/local/sbin/ser[3154]: rwrite(): Error in do_action >May 6 15:11:58 nw127 /usr/local/sbin/ser[3154]: lookup(): Unable to rewrite Request-URI >May 6 15:11:58 nw127 /usr/local/sbin/ser[3154]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 539 >May 6 15:13:21 nw127 /usr/local/sbin/ser[3118]: WARNING: unhandled signal 12 >May 6 15:15:47 nw127 /usr/local/sbin/ser[3163]: build_rr(): No memory left >May 6 15:15:47 nw127 /usr/local/sbin/ser[3163]: insert_RR(): Error while inserting outbound Record-Route >May 6 15:15:47 nw127 /usr/local/sbin/ser[3163]: record_route(): Error while inserting Record-Route line >May 6 15:15:47 nw127 /usr/local/sbin/ser[3163]: ERROR: do_action: memory allocation failure >May 6 15:15:47 nw127 /usr/local/sbin/ser[3163]: rwrite(): Error in do_action >May 6 15:15:47 nw127 /usr/local/sbin/ser[3163]: lookup(): Unable to rewrite Request-URI >May 6 15:15:47 nw127 /usr/local/sbin/ser[3163]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 589 >May 6 15:15:47 nw127 /usr/local/sbin/ser[3166]: build_rr(): No memory left >May 6 15:15:47 nw127 /usr/local/sbin/ser[3166]: insert_RR(): Error while inserting outbound Record-Route >May 6 15:15:47 nw127 /usr/local/sbin/ser[3166]: record_route(): Error while inserting Record-Route line >May 6 15:15:47 nw127 /usr/local/sbin/ser[3166]: ERROR: mk_proxy: memory allocation failure >I change the value of PKG_MEM_POOL_SIZE by 10240*1024 in config.h and recompile. >But I still has the same memory error message.How can get memory statistics when I send kill -SIGUSR1? This looks like a memory leak in your module (or in SER?) unfortunately. To get memory dumps printed, recompile SER with memory logging (i.e., replace -DF_MALLOC with -DDBG_QM_MALLOC in Makefile.defs) and set configuration option 'memlog' low (e.g., 'memlog=1'). On restart, SER will dump all unallocated fragments. It will be quite a lot -- each process will have many of them, but eventually we should be able to figure out some fragments which will be reocurring too frequently. -jiri

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sr-users May 2004