sr-users February 2007

sr-users@lists.kamailio.org

158 participants
285 discussions

by Klaus Darilion

Hi! What are the return values of avp_db_load? E.g. is it FALSE on problems accessing the DB or FALSE if it did not found any AVP to load? I couldn't find this documented somewhere. regards klaus -- Klaus Darilion nic.at

17 years, 9 months

[Serusers] Anyone using CDRTool ?

by Lokesh Kumar

Hello All, Is there anyone using CDRTool in conjugation with ser, asterisk, radius setup. Can anyone give me a hand to setup it properly? Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.411 / Virus Database: 268.17.27/671 - Release Date: 2/5/2007

17 years, 9 months

[Serusers] Request for Daniel-Constantin Mierla

by Giovanni Chirico

Hi Daniel, Im trying to use the dispatcher module. Im using the call forwarding ser.cfg file of the SER Getting Started <http://www.iptel.org/ser/doc/gettingstarted> pack. I have inserted the function ds_select_dst("1", "0"); In the main route() and the function forward(uri:host, uri:port); in the PSTN Handler, route(5). How I can made the failure route, so if a gateway is down another gateway is load? In attachment there are my ser.cfg and dispatcher.list files. Please help me :-( Bye, Giò

17 years, 9 months

[Serusers] automated response

by Myron Groening

Vacation Alert !! I will be out of the office, and will not be checking my message till FEB. 26 2007 Thanks Myron Groening

17 years, 9 months

[Serusers] new version of radiusclient-ng

by Jan Janak

Hello, I just released a new version of radiusclient-ng library. The new version, 0.5.4, fixes problems on 64-bit architectures. You can download the new release from http://ftp.iptel.org/pub/radiusclient-ng/radiusclient-ng-0.5.4.tar.gz Credits go to Peter Nixon. Jan.

17 years, 9 months

[Users] ERROR: module version mismatch for,/usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls,(i386/freebsd); module: openser 1.1.1-notls (i386/freebsd)

by Hansi Lao

Hello All, I have installed openser-1.1.1 with tls on my box. However, when I try to run openser. It returns me this error: ERROR: module version mismatch for /usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls (i386/freebsd); module: openser 1.1.1-notls (i386/freebsd). Would someone be kind enough to help me how to compile a mysql.so which supports tls? Thank you.

17 years, 9 months

[Users] migrating openser v1.1.x to v1.2.x

by Daniel-Constantin Mierla

Hello, since we got in testing phase for 1.2.0 release, I think is better to start early a document with migration guidelines. Issues encountered during upgrade should be gathered together in the next page, to share and help each others: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x It is just the skeleton and few stuff there. Hopefully the content will grow quickly. Everybody can contribute. Cheers, Daniel

17 years, 9 months

[Users] How can i make my OpenSER server with presence module work properly with the db_text database?

by Kc Ayuba

Hi, I'm trying to use db_text database with the presence module of OpenSER(1.2.x ). User agent is X-Lite(3.0). When I try to send a REGISTER message to OpenSER, OpenSER still works. But if i sent a SUBSCRIBE or a PUBLISH message, then openser crashes. If i use the MySQL database, then everything works just fine. But i prefer to use the db_text database. Here is the core file that was generated when OpenSER crashed. Core generated: Program terminated with signal 11, Segmentation fault. #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) bt #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 #1 0x080a3f6c in timer_ticker () at timer.c:141 #2 0x0806b51f in main_loop () at main.c:724 #3 0x0806d43b in main (argc=3, argv=0xaff0a2e4) at main.c:1438 (gdb) frame 0 #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) list 605 606 subs.from_domain.s = row_vals[from_domain_col].val.str_val.s; 607 subs.from_domain.len = strlen(row_vals[from_domain_col].val.str_val.s); 608 609 subs.event_id.s = row_vals[event_id_col].val.str_val.s; 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); 611 612 subs.to_tag.s = row_vals[to_tag_col].val.str_val.s; 613 subs.to_tag.len = strlen(row_vals[to_tag_col].val.str_val.s); 614 (gdb) 615 subs.from_tag.s = row_vals[from_tag_col].val.str_val.s; 616 subs.from_tag.len = strlen(row_vals[from_tag_col].val.str_val.s); 617 618 subs.callid.s = row_vals[callid_col].val.str_val.s; 619 subs.callid.len = strlen(row_vals[callid_col].val.str_val.s); 620 621 subs.contact.s = row_vals[contact_col].val.str_val.s; 622 subs.contact.len = strlen(row_vals[contact_col].val.str_val.s); 623 624 subs.cseq = row_vals[cseq_col].val.int_val; I really hope that someone can help me with this problem. Thanks in advance. Kind regards, Kc

17 years, 9 months

RE: [Users] Asterisk OpenSER and Exchange 2007 Unified Messaging

by Jon Webster

The significant difference of debug from a good call and openser is next. Below that are the full debug logs of both the openser and goodproxy call with an ngrep of openser's call in between. significant differences ======================== Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> VERSES --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: asterisk debug of openser call =========== --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 180 Ringing FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- Called openser/8886000 -- SIP/openser-082966e8 is ringing pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 200 OK FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM> CONTENT-LENGTH: 197 CONTENT-TYPE: application/sdp SERVER: RTCC/2.0.6017.0 v=0 o=- 0 0 IN IP4 exchangeUM s=Microsoft Exchange Speech Engine c=IN IP4 exchangeUM t=0 0 m=audio 30660 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> <-- SIP read from openser:5060: BYE sip:3149@asterisk SIP/2.0 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f TO: <sip:3149@asterisk>;tag=as4aa53078 CSEQ: 1 BYE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 69 Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825 VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 CONTENT-LENGTH: 0 USER-AGENT: RTCC/2.0.6017.0 P-hint: outbound --- (13 headers 0 lines) --- Sending to openser : 5060 (non-NAT) Transmitting (no NAT) to openser:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f To: <sip:3149@asterisk>;tag=as4aa53078 Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:3149@asterisk> Content-Length: 0 ngrep of above call ====================== # U +0.000616 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.067855 openser:5060 -> asterisk:5060 SIP/2.0 200 OK. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 30660 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U +32.353338 openser:5060 -> asterisk:5060 BYE sip:3149@asterisk SIP/2.0. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. TO: <sip:3149@asterisk>;tag=as4aa53078. CSEQ: 1 BYE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 69. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825. VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. CONTENT-LENGTH: 0. USER-AGENT: RTCC/2.0.6017.0. P-hint: outbound. . # U +0.000734 asterisk:5060 -> openser:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser. Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. To: <sip:3149@asterisk>;tag=as4aa53078. Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:3149@asterisk>. Content-Length: 0. . good call from working proxy ======== We're at asterisk port 17508 Video is at asterisk port 11130 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to goodproxy:5060: INVITE sip:6000@goodproxy SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 05 Feb 2007 14:55:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 381 381 IN IP4 asterisk s=session c=IN IP4 asterisk t=0 0 m=audio 17508 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called exchange12/6000 pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport;received=goodproxy From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: M-Networks USR/1.0 Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE Content-Length: 0 --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- SIP/exchange12-082b8e18 is ringing pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTACT: <sip:goodproxy:5060> CONTENT-LENGTH: 197 CONTENT-TYPE: application/sdp SERVER: RTCC/2.0.6017.0 v=0 o=- 0 0 IN IP4 exchangeUM s=Microsoft Exchange Speech Engine c=IN IP4 exchangeUM t=0 0 m=audio 47688 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: ACK sip:goodproxy:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK6c3f8b53;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/exchange12-082b8e18 answered SIP/jonlaptop-082b0fb8 -- Attempting native bridge of SIP/jonlaptop-082b0fb8 and SIP/exchange12-082b8e18 Scheduling destruction of call '109c82647d6340be2a5ab19e7eb18a14@asterisk' in 32000 ms set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set de5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK47d7f855;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (local, 7776000, 3) exited non-zero on 'SIP/jonlaptop-082b0fb8' pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK47d7f855;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;tag=f51839ab98;epid=BD-70-82-06-F9 CSEQ: 103 BYE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- Destroying call '109c82647d6340be2a5ab19e7eb18a14@asterisk' pbx*CLI>

17 years, 9 months

[Users] CVS frozen - testing phase of 1.2.0

by Daniel-Constantin Mierla

Hello, starting with today the development for 1.2.0 is closed. No new features should be added until the release. The next month will be dedicated to testing, integration of submitted patches and fixing reported bugs. We have a lot of new code to test, your help is very much appreciated. I want to list the major changes you should focus on while testing: - usage of pseudo-variables and transformations is script (http://openser.org/dokuwiki/doku.php/transformations:devel) - TM timers (millisecond accuracy) - IP blacklists - DNS failover - automatic error handling - flags system (script flags, message flags, branch flags) - usrloc enhancements (licking system refurbished, database content optimization via DB fetch support) - new management interface (mi_fifo, mi_xmlrpc) - SIP Servlet Java Application server interoperability (seas module) - PERL AGI (perl module) - SNMP support via snmpstats module - Presence support (modules presence, pua, pua_usrloc, pua_mi) - xmpp IM gateway (xmpp module) - sip session timer support (module sst) - accounting clean up - federation based peering via domainpolicy module If I forgot something, please contribute. In addition, great help would be with creating 1.2.0 variants of documentation and dokuwiki pages (where is the case), testing the tools (openserctl, DB creation scripts) and developing scripts to migrate database from 1.1.x to 1.2.0. To keep track of discovered issues, please register them on the tracker -- it this way you make sure it is not lost in mail threads: http://sourceforge.net/tracker/?group_id=139143&atid=743020 Have nice testing sessions! Your feedback is always helpful! Cheers, Daniel

17 years, 9 months

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sr-users February 2007