sr-users November 2004

sr-users@lists.kamailio.org

203 participants
386 discussions

[Serusers] Need complete docbook.dtd to view sgml docs
by msg 26 Nov '04

26 Nov '04

Greetings: Our only (currently available) method to browse the sgml docs is to use Panorama for Win16 (free version) or Novell Dynatext and Sun Answerbook2. I've grabbed a great variety of docbook.dtd files besides the ones included with the above tools; they do not include various elements such as "section id". A search of the oasis site came up negative except for some e-mail traffic. Is there something I can get from iptel.org? Regards, Michael Grigoni Cybertheque Museum

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[Serusers] ser forwarding
by kcassidy＠kakelma.mine.nu 26 Nov '04

26 Nov '04

Hi All, Is it possible for SER to forward SIP traffic to another SER? I have a setup below. Where all SER clients will point to SER-A and SER-A will just forward all request to SER-B. I know how to forward INVITES to SER-B. But is it is possible to forward REGISTER, and all other METHODS? SER-A --> SER-B TIA

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[Serusers] val2str(): Destination buffer too short , pls help
by Zhang Wei 26 Nov '04

26 Nov '04

hi, i have met the same problem with latest version as Peter has met. i loaded the latest version today. In ser.cfg, when I wrote: modparam("usrloc", "db_mode", 1), the user can not registr at all. when I wrote: modparam("usrloc", "db_mode", 2) the user can register but for a moment the debug information shows as the fellow: ---------------------------------------------------------------- entering route[2] - user not online val2str(): Destination buffer too short print_values(): Error while converting value to string submit_query(): Column count doesn't match value count at row 1 insert_row(): Error while submitting query db_insert_ucontact(): Error while inserting contact wb_timer(): Error while inserting contact into database ---------------------------------------------------------------- And the Location table of ser database have no records at all. After a few minutes SER server is down. the debug information shows as the fellow: ---------------------------------------------------------------- val2str(): Destination buffer too short val2str(): Destination buffer too short child process 10639 exited by a signal 11 core was generated INFO: terminating due to SIGCHLD INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received ----------------------------------------------------------------------- where is wrong?

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[Serusers] problem with auth_module
by Todor Milenkov 26 Nov '04

26 Nov '04

Hi All, Could you help me whit "auth_module"? I compiled "radiusclient-0.4.6" and "ser-0.8.14" with auth_radius module. But when i insert module "auth_radius" in ser.cfg file I get this message "Starting SER : cat: /var/run/ser.pid: No such file or directory" Could you advice me, pls. 10x in advance!

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[Serusers] About priority calls in the Ser.
by D2 26 Nov '04

26 Nov '04

Hello all, I want make a call which would be forked with multiple destionation, and each of forked call has a priority number(spcified in SIP RFC, the q field in SIP header.). To do this, I first use the serctl command to add user with specified q number like this: % hsucc- sudo serctl ul show 54760 <sip:054760@140.113.214.205>;q=1.00;expires=1003738934 <sip:54760@140.113.214.205:5060>;q=0.00;expires=3597 But while testing, ser didn't seem to follow the q header field. Could anyone can tell me if I had anything worng about this? P.S. My ser is version 0.8.14. Any advice would be great...... -- Best regards, D2 mailto:leonheart.csie90@nctu.edu.tw

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[Serusers] SER Proxy auth
by E. Versaevel 26 Nov '04

26 Nov '04

Hello, I'm currently trying to setup a SIP environment for VoIP calling for my final school project, so I'm just working with VoIP/SIP for 2 weeks. I'm using SER as a SIP proxy server, but the carrier/gateway I am using for calling to/from PSTN is requiring me to register at their server and authorize outgoing calls, which is something SER won't do. So I got the idea to use asterisk between the PSTN carrier and SER for the authorization, since Asterisk can register and auth itself. SIP --------- SIP --------- SIP --------- PSTN -----| |-----------| |-----------| |------ --------- --------- --------- SER Asterisk Carrier <-- auth stuff --> <-- sip relay --> Has anyone here ever tried a similar setup or is this an impossibility ? Kind regards, E. Versaevel

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[Serusers] UA Replies With 401 -- Has Anyone Ever Seen This Before??
by Java Rockx 26 Nov '04

26 Nov '04

Hello All. This may not be a ser issue. Anyhow, here is a snippet of a SIP conversation from my UTstarcomm iAN-02EX ATA to the PSTN. Once the PSTN gateway sends a re-INVITE message the ATA replies with a 401. Has anyone ever seen this before? Regards, Paul IP Legend --------------- PSTN Gateway - 200.99.100.177 SER Proxy - 67.93.120.33 UTstarcom ATA - 68.84.240.12 U 2004/11/25 21:07:02.138482 200.99.100.177:5060 -> 67.93.120.33:5060 INVITE sip:9990021001@68.84.240.12:60324 SIP/2.0. Via: SIP/2.0/UDP 200.99.100.177:5060;branch=z9hG4bK1fb47bafa1a-f9d0f16. Via: SIP/2.0/UDP 216.229.118.76:4060;branch=z9hG4bKe5004985c3705e02. To: "UTStarcom" <sip:9990021001@200.99.100.177;bgid=100009995;bgt=private>;tag=4159092737. From: sip:4075551212@sip.mycompany.com;tag=0be7a9eb. Call-ID: 824690162(a)172.16.1.43. CSeq: 27980 INVITE. Max-Forwards: 69. Contact: sip:4075551212@216.229.118.76:4060. Record-Route: <sip:200.99.100.177:5060;lr>. Route: <sip:67.93.120.33:5060;ftag=4159092737;lr=on>. Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO. Accept: multipart/mixed, application/sdp, application/isup, application/dtmf, application/dtmf-relay. Supported: timer. Content-Disposition: session;handling=required. Content-Type: application/sdp. Session-Expires: 240;refresher=uac. Content-Length: 228. . v=0. o=Sonus_UAC 10391 17026 IN IP4 216.229.118.76. s=SIP Media Capabilities. c=IN IP4 216.229.118.101. t=0 0. m=audio 15414 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. # U 2004/11/25 21:07:02.139790 67.93.120.33:5060 -> 200.99.100.177:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 200.99.100.177:5060;branch=z9hG4bK1fb47bafa1a-f9d0f16. Via: SIP/2.0/UDP 216.229.118.76:4060;branch=z9hG4bKe5004985c3705e02. To: "UTStarcom" <sip:9990021001@200.99.100.177;bgid=100009995;bgt=private>;tag=4159092737. From: sip:4075551212@sip.mycompany.com;tag=0be7a9eb. Call-ID: 824690162(a)172.16.1.43. CSeq: 27980 INVITE. Server: mycompany SIP Router (0.8.99-dev19 (i386/linux)). Content-Length: 0. Warning: 392 67.93.120.33:5060 "Noisy feedback tells: pid=13383 req_src_ip=200.99.100.177 req_src_port=5060 in_uri=sip:9990021001@68.84.240.12:60324 out_uri=sip:9990021001@68.84.240.12:60324 via_cnt==2". . # U 2004/11/25 21:07:02.140268 67.93.120.33:5060 -> 68.84.240.12:60324 INVITE sip:9990021001@68.84.240.12:60324 SIP/2.0. Record-Route: <sip:67.93.120.33;ftag=0be7a9eb;lr=on>. Via: SIP/2.0/UDP 67.93.120.33;branch=z9hG4bKf19b.7cb468b4.0. Via: SIP/2.0/UDP 200.99.100.177:5060;branch=z9hG4bK1fb47bafa1a-f9d0f16. Via: SIP/2.0/UDP 216.229.118.76:4060;branch=z9hG4bKe5004985c3705e02. To: "UTStarcom" <sip:9990021001@200.99.100.177;bgid=100009995;bgt=private>;tag=4159092737. From: sip:4075551212@sip.mycompany.com;tag=0be7a9eb. Call-ID: 824690162(a)172.16.1.43. CSeq: 27980 INVITE. Max-Forwards: 16. Contact: sip:4075551212@216.229.118.76:4060. Record-Route: <sip:200.99.100.177:5060;lr>. Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO. Accept: multipart/mixed, application/sdp, application/isup, application/dtmf, application/dtmf-relay. Supported: timer. Content-Disposition: session;handling=required. Content-Type: application/sdp. Session-Expires: 240;refresher=uac. Content-Length: 228. . v=0. o=Sonus_UAC 10391 17026 IN IP4 216.229.118.76. s=SIP Media Capabilities. c=IN IP4 216.229.118.101. t=0 0. m=audio 15414 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. # U 2004/11/25 21:07:02.436224 68.84.240.12:60324 -> 67.93.120.33:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 67.93.120.33;branch=z9hG4bKf19b.7cb468b4.0. Via: SIP/2.0/UDP 200.99.100.177:5060;branch=z9hG4bK1fb47bafa1a-f9d0f16. Via: SIP/2.0/UDP 216.229.118.76:4060;branch=z9hG4bKe5004985c3705e02. From: <sip:4075551212@sip.mycompany.com>;tag=0be7a9eb. To: "UTStarcom" <sip:9990021001@200.99.100.177;bgid=100009995;bgt=private>;tag=4159092737. Call-ID: 824690162(a)172.16.1.43. CSeq: 27980 INVITE. WWW-Authenticate: Digest realm="sip.mycompany.com", nonce="e1ee2e70d502f470c9fd35ce2bd2fdcc", opaque="6183bcbc6a0c54d44f910e532e96896d". Content-Length: 0. . __________________________________ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com

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[Serusers] t_relay to local domain?
by serusers＠jtoga.org 25 Nov '04

25 Nov '04

Hi, Not sure if this is simply a dumb thing to do or the configuration is wrong... .8.14 SER is running mostly fine. 1) SER gets an invite to 123(a)domain.com 2) lookup() resolves this to 456(a)domain.com (serctl ul show 123 returns sip:456@domain.com) 3) t_relay is called and eventually get a 408 and the URI indicates 123(a)domain.com Logging has shown that the start of route logic is never called with 456(a)domain.com. It seems like the t_relay() is internally sitting on the location URI. Is this intentional, to stop loops from occurring? I realize that 'aliases' would better serve this but would like to understand what is occurring under the covers. Thanks in advance. Jim

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[Serusers] Parse body of a SIP message in external script
by Klaus Darilion 25 Nov '04

25 Nov '04

Hi all! Is it possible to handle the body of a SIP message (or the complete message) to an external script? I tried the exec module without luck. regards, klaus

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[Serusers] Location table + replicate + strange behavior
by Deepak Lal 25 Nov '04

25 Nov '04

Environment: ser 0.8.14-3 on a AMD64 box Two ethernet interfaces with SER listening only on ONE I have a db_mode of 2 for usrloc. What I do not understand is the once contact information is written to the location db, it does not get flushed at all. I still have "old" contact info in the location db for the UA's. In the log file, I see the recurring message: "Keeping binding '2221111', sip:2221111@X.X.X.X" for replication The message is pretty self-explanatory but I have not configured any replication. Can someone please explain what is happening and how to fix this. Thx in advance __________________________________ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail

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sr-users November 2004