- sr-users - kamailio.org

[Serusers] Meaning of and proper response for a 401
by SIP 12 Jul '07

12 Jul '07

There's a UA in our network that responds to a reinvite with a 401. U PROXY.IP.ADDRESS:5060 -> UA.IP.ADDRESS:63718 INVITE sip:1101XXXXXXX@UA.IP.ADDRESS:63718 SIP/2.0. Record-Route: <sip:PROXY.IP.ADDRESS;ftag=as7bbe322e;lr=on>. Via: SIP/2.0/UDP PROXY.IP.ADDRESS;branch=z9hG4bK9159.cd2a95b4.0. Via: SIP/2.0/UDP PSTN.GW.IP:5090;branch=z9hG4bK39c17f01;rport=5090. From: <sip:13154476314@our.domain.com>;tag=as7bbe322e. To: "User One" <sip:1101XXXXXXX@our.domain.com>;tag=202d9942. Contact: <sip:13154476314@PSTN.GW.IP:5090>. Call-ID: 6b3f44351f5fab945ac212a2141df93a(a)UA.IP.ADDRESS. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 314. P-hint: LR. . v=0. o=root 15241 15243 IN IP4 216.143.130.90. s=session. c=IN IP4 216.143.130.90. t=0 0. m=audio 17192 RTP/AVP 0 8 18 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U UA.IP.ADDRESS:63718 -> PROXY.IP.ADDRESS:5060 SIP/2.0 410 Gone. Via: SIP/2.0/UDP PROXY.IP.ADDRESS:5060;branch=z9hG4bK9159.cd2a95b4.0. Via: SIP/2.0/UDP PSTN.GW.IP:5090;branch=z9hG4bK39c17f01;rport=5090. From: <sip:13154476314@our.domain.com>;tag=as7bbe322e. To: "User One" <sip:1101XXXXXXX@our.domain.com>;tag=202d9942. Contact: <sip:1101XXXXXXX@UA.IP.ADDRESS:63718>. Call-ID: 6b3f44351f5fab945ac212a2141df93a(a)UA.IP.ADDRESS. CSeq: 102 INVITE. Content-Length: 0. First off, I thought a 410 was supposed to be unauthorised (what is 'Gone' ?). Secondly... is the UA requesting our server to authorise its reinvite, or is something else going on here I clearly don't follow? N.

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[OpenSER-Users] query: how to deal with crazy clients
by Klaus Darilion 12 Jul '07

12 Jul '07

Hi! I once again bother about a stupid SIP client which is misconfigured and sending REGISTER messages like crazy to my SIP proxy (~20 requests/second). How do you deal with that stupid clients. Manual blocking does not work as the client has dynamic IP addresses. I once also tried the pike module but had problems to find out a proper configuration (threshold value and sampling interval). Further, I think blocking at IP level (e.g. openser injects iptable rules) would be more useful. I would be happy to hear how you handle such situations.

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[OpenSER-Users] SER query regarding mysql
by Parminder.Nayyar＠relianceada.com 12 Jul '07

12 Jul '07

hi, we are facing following problem while using SER with mysql data base. The problem statements are as follows: 0(0) find_mod_export: found <db_update> in module mysql [/export/home/mins/SER/lib/ser/modules/mysql.so] 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /export/home/mins/SER/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /export/home/mins/SER/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /export/home/mins/SER/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /export/home/mins/SER/lib/ser/modules/rr.so record_route 0(0) fixing /export/home/mins/SER/lib/ser/modules/rr.so loose_route 0(0) fixing /export/home/mins/SER/lib/ser/modules/tm.so t_relay 0(0) fixing /export/home/mins/SER/lib/ser/modules/registrar.so lookup 0(0) fixing /export/home/mins/SER/lib/ser/modules/registrar.so lookup 0(0) fixing /export/home/mins/SER/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /export/home/mins/SER/lib/ser/modules/tm.so t_relay 0(0) fixing /export/home/mins/SER/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /export/home/mins/SER/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /export/home/mins/SER/lib/ser/modules/auth_db.so www_authorize 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /export/home/mins/SER/lib/ser/modules/auth.so www_challenge 0(0) fixing /export/home/mins/SER/lib/ser/modules/uri_db.so check_to 0(0) fixing /export/home/mins/SER/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /export/home/mins/SER/lib/ser/modules/auth.so consume_credentials 0(0) fixing /export/home/mins/SER/lib/ser/modules/registrar.so save 0(0) fixing /export/home/mins/SER/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /export/home/mins/SER/lib/ser/modules/auth_db.so proxy_authorize 0(0) get_connection(): Connection not found in the pool 0(0) release_connection(): Removing connection from the pool 0(0) fixing /export/home/mins/SER/lib/ser/modules/auth.so proxy_challenge 0(0) fixing /export/home/mins/SER/lib/ser/modules/auth.so consume_credentials 0(0) fixing /export/home/mins/SER/lib/ser/modules/registrar.so lookup 0(0) fixing /export/home/mins/SER/lib/ser/modules/registrar.so lookup 0(0) fixing /export/home/mins/SER/lib/ser/modules/sl.so sl_send_reply 0(0) INFO: udp_init: SO_RCVBUF is initially 65536 0(0) DEBUG: udp_init: trying SO_RCVBUF: 131072 0(0) DEBUG: setting SO_RCVBUF; set=131072,verify=131072 0(0) DEBUG: udp_init: trying SO_RCVBUF: 262144 0(0) DEBUG: setting SO_RCVBUF; set=262144,verify=262144 0(0) INFO: udp_init: SO_RCVBUF is finally 262144 0(0) ERROR: tcp_init: bind(5, e83bc, 16) on 10.128.16.254: Address already in use 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(20185) shm_mem_destroy 0(20185) destroying the shared memory lock The ser.cfg file used is attached with the mail. Plz sugest me some solution for mentioned problem... thanx 'n' regds, Parminder Nayyar

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[OpenSER-Users] BLA errors
by Jeremy George 12 Jul '07

12 Jul '07

Hello All, I have BLA implemented, with errors. An ngrep trace of relevant packets showing errors follows. The configurations of both main proxy and bla presence server are attached. I stripped the main proxy to the bone. No outside routing at all. The bla presence server is almost all from the sample. 75.56.10.115 is a Cisco 7960 with one line registered: jeremy/private 75.56.10.116 is a Polycom 650 with two lines registered: jessica/private and helen/shared 75.56.10.117 is a Polycom 330 with two lines registered: didama/private and helen/shared 128.91.3.19 is the main proxy running openser v1.2.1 . 128.91.2.215 is the bla presence server running openser svn 2413 . This error: "1(1184) PUA:subs_cback_func: no reply message found" is being produced repeatedly on the bla presence server. Finally, I'm getting these error messages on the bla presence server: 0(1284) PUA_BLA: handle_notify: ERROR Notify in a non existing dialog 0(1284) WARNING: script writer didn't release transaction From time-to-time pua entries eat all of available memory. Curiously, they were all for the Polycom 330 and none were for the Polycom 650. Any help is greatly appreciated. Thanks. - Jeremy # U 128.91.2.215:5060 -> 75.56.10.117:5060 SUBSCRIBE sip:helen@75.56.10.117 SIP/2.0..Via: SIP/2.0/UDP 128.91.2.215;bra nch=z9hG4bKd187.ae440494.0..To: sip:helen@75.56.10.117;tag=9D362AB1-962A397 2..From: sip:helen@128.91.2.215;tag=45cdb744f974e73ed68392c080e87bb4-2684.. CSeq: 11 SUBSCRIBE..Call-ID: 7df1b9fa-810@128.91.2.215..Content-Length: 0.. User-Agent: OpenSER (1.3.0-dev7-tls (i386/linux))..Event: dialog;sla..Conta ct: <sip:helen@128.91.2.215>..Expires: 1810.... # U 75.56.10.117:5060 -> 128.91.2.215:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist..Via: SIP/2.0/UDP 128.91.2. 215;branch=z9hG4bKd187.9e440494.0..From: <sip:helen@128.91.2.215>;tag=45cdb 744f974e73ed68392c080e87bb4-2684..To: <sip:helen@75.56.10.117>;tag=9D362AB1 -962A3972..CSeq: 11 SUBSCRIBE..Call-ID: 7df1b9fa-810@128.91.2.215..Event: d ialog;sla..User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049..Content- Length: 0.... # # U 128.91.2.215:5060 -> 75.56.10.117:5060 SUBSCRIBE sip:helen@75.56.10.117 SIP/2.0..Via: SIP/2.0/UDP 128.91.2.215;bra nch=z9hG4bK0de3.730cc385.0..To: sip:helen@75.56.10.117..From: sip:helen@128 .91.2.215;tag=45cdb744f974e73ed68392c080e87bb4-be09..CSeq: 10 SUBSCRIBE..Ca ll-ID: 7df1ba94-810@128.91.2.215..Content-Length: 0..User-Agent: OpenSER (1 .3.0-dev7-tls (i386/linux))..Event: dialog;sla..Contact: <sip:helen@128.91. 2.215>..Expires: 1820.... # U 75.56.10.117:5060 -> 128.91.2.215:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 128.91.2.215;branch=z9hG4bK0de3.730cc385.0 ..From: <sip:helen@128.91.2.215>;tag=45cdb744f974e73ed68392c080e87bb4-be09. .To: <sip:helen@75.56.10.117>;tag=8C7BE007-7BDCD3A4..CSeq: 10 SUBSCRIBE..Ca ll-ID: 7df1ba94-810@128.91.2.215..Contact: <sip:helen@75.56.10.117>..Event: dialog;sla..User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049..Expire s: 1820..Content-Length: 0.... # # U 128.91.3.19:5060 -> 75.56.10.116:5060 SUBSCRIBE sip:helen@75.56.10.116 SIP/2.0..Record-Route: <sip:128.91.3.19;lr =on;ftag=36887A42-5425822F>..Via: SIP/2.0/UDP 128.91.3.19;branch=z9hG4bK27d 4.3e70f5b4.2..Via: SIP/2.0/UDP 75.56.10.117;branch=z9hG4bKa5b69571873BC26E. .From: "Helen" <sip:helen@128.91.3.19>;tag=36887A42-5425822F..To: <sip:hele n@128.91.3.19:5060>..CSeq: 1 SUBSCRIBE..Call-ID: aa7d279b-1806c1f8-b6acbe85 @75.56.10.117..Contact: <sip:helen@75.56.10.117>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..Ev ent: call-info..User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049..Max -Forwards: 70..Expires: 1800..Content-Length: 0.... # U 75.56.10.117:5060 -> 128.91.3.19:5060 SIP/2.0 482 Loop Detected..Via: SIP/2.0/UDP 128.91.3.19;branch=z9hG4bK27d4. 3e70f5b4.0..Via: SIP/2.0/UDP 75.56.10.117;branch=z9hG4bKa5b69571873BC26E..F rom: "Helen" <sip:helen@128.91.3.19>;tag=36887A42-5425822F..To: <sip:helen@ 128.91.3.19:5060>..CSeq: 1 SUBSCRIBE..Call-ID: aa7d279b-1806c1f8-b6acbe85@7 5.56.10.117..Contact: <sip:helen@75.56.10.117>..Record-Route: <sip:128.91.3 .19;lr=on;ftag=36887A42-5425822F>..Event: call-info..User-Agent: PolycomSou ndPointIP-SPIP_330-UA/2.1.2.0049..Content-Length: 0.... # U 75.56.10.116:5060 -> 128.91.3.19:5060 SIP/2.0 501 Not Implemented..Via: SIP/2.0/UDP 128.91.3.19;branch=z9hG4bK27d 4.3e70f5b4.1..Via: SIP/2.0/UDP 75.56.10.117;branch=z9hG4bKa5b69571873BC26E. .From: "Helen" <sip:helen@128.91.3.19>;tag=36887A42-5425822F..To: <sip:hele n@128.91.3.19:5060>;tag=BC1615E5-7F8E8B8C..CSeq: 1 SUBSCRIBE..Call-ID: aa7d 279b-1806c1f8-b6acbe85@75.56.10.117..Contact: <sip:helen@75.56.10.116>..Rec ord-Route: <sip:128.91.3.19;lr=on;ftag=36887A42-5425822F>..Event: call-info ..User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078..Expires: 1800..Co ntent-Length: 0.... # # U 75.56.10.117:5060 -> 128.91.2.215:5060 NOTIFY sip:helen@128.91.2.215 SIP/2.0..Via: SIP/2.0/UDP 75.56.10.117;branch =z9hG4bK70de934cAD6A9A99..From: <sip:helen@75.56.10.117>;tag=8C7BE007-7BDCD 3A4..To: <sip:helen@128.91.2.215>;tag=45cdb744f974e73ed68392c080e87bb4-be09 ..CSeq: 1 NOTIFY..Call-ID: 7df1ba94-810@128.91.2.215..Contact: <sip:helen@7 5.56.10.117>..Event: dialog;sla..User-Agent: PolycomSoundPointIP-SPIP_330-U A/2.1.2.0049..Subscription-State: active;expires=1818..Max-Forwards: 70..Co ntent-Type: application/dialog-info+xml..Content-Length: 156....<?xml versi on="1.0"?>..<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version ="0" state="full" entity="sip:helen@128.91.3.19:5060">..</dialog-info> # U 128.91.2.215:5060 -> 75.56.10.117:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 75.56.10.117;branch=z9hG4bK70de934cAD6A9A9 9..From: <sip:helen@75.56.10.117>;tag=8C7BE007-7BDCD3A4..To: <sip:helen@128 .91.2.215>;tag=45cdb744f974e73ed68392c080e87bb4-be09..CSeq: 1 NOTIFY..Call- ID: 7df1ba94-810@128.91.2.215..Server: OpenSER (1.3.0-dev7-tls (i386/linux) )..Content-Length: 0.... # U 128.91.2.215:5060 -> 75.56.10.116:5060 NOTIFY sip:helen@75.56.10.116 SIP/2.0..Via: SIP/2.0/UDP 128.91.2.215;branch =z9hG4bK8d7b.ab428dd3.0..To: sip:helen@128.91.3.19;tag=FBC67234-A17D6FE3..F rom: sip:helen@128.91.2.215;tag=10.810.1184176700.14..CSeq: 3 NOTIFY..Call- ID: a023d5f7-3f2d49fe-76f2717d@75.56.10.116..Content-Length: 156..User-Agen t: OpenSER (1.3.0-dev7-tls (i386/linux))..Max-Forwards: 70..Event: dialog;s la..Contact: <sip:128.91.2.215:5060>..Subscription-State: active;expires=32 83..Content-Type: application/dialog-info+xml....<?xml version="1.0"?>.<dia log-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="15" state="ful l" entity="sip:helen@128.91.3.19:5060">.</dialog-info>. # U 75.56.10.116:5060 -> 128.91.2.215:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist..Via: SIP/2.0/UDP 128.91.2. 215;branch=z9hG4bK8d7b.ab428dd3.0..From: <sip:helen@128.91.2.215>;tag=10.81 0.1184176700.14..To: <sip:helen@128.91.3.19>;tag=FBC67234-A17D6FE3..CSeq: 3 NOTIFY..Call-ID: a023d5f7-3f2d49fe-76f2717d@75.56.10.116..Event: dialog;sl a..User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078..Content-Length: 0.... # U 128.91.2.215:5060 -> 75.56.10.116:5060 NOTIFY sip:helen@75.56.10.116 SIP/2.0..Via: SIP/2.0/UDP 128.91.2.215;branch =z9hG4bK06ea.193b0e22.0..To: sip:helen@128.91.3.19;tag=D976CBBC-2176183F..F rom: sip:helen@128.91.2.215;tag=10.810.1184176890.21..CSeq: 2 NOTIFY..Call- ID: e11fdddb-9530f65a-3123a455@75.56.10.116..Content-Length: 156..User-Agen t: OpenSER (1.3.0-dev7-tls (i386/linux))..Max-Forwards: 70..Event: dialog;s la..Contact: <sip:128.91.2.215:5060>..Subscription-State: active;expires=34 73..Content-Type: application/dialog-info+xml....<?xml version="1.0"?>.<dia log-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="15" state="ful l" entity="sip:helen@128.91.3.19:5060">.</dialog-info>. # U 75.56.10.116:5060 -> 128.91.2.215:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist..Via: SIP/2.0/UDP 128.91.2. 215;branch=z9hG4bK06ea.193b0e22.0..From: <sip:helen@128.91.2.215>;tag=10.81 0.1184176890.21..To: <sip:helen@128.91.3.19>;tag=D976CBBC-2176183F..CSeq: 2 NOTIFY..Call-ID: e11fdddb-9530f65a-3123a455@75.56.10.116..Event: dialog;sl a..User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078..Content-Length: 0.... # U 128.91.2.215:5060 -> 75.56.10.116:5060 NOTIFY sip:helen@75.56.10.116 SIP/2.0..Via: SIP/2.0/UDP 128.91.2.215;branch =z9hG4bKdcf1.0c60f511.0..To: sip:helen@128.91.3.19;tag=18F7DEB-D3FFDEAA..Fr om: sip:helen@128.91.2.215;tag=10.810.1184176985.23..CSeq: 2 NOTIFY..Call-I D: a7685c76-af0100e1-7b4a7078@75.56.10.116..Content-Length: 156..User-Agent : OpenSER (1.3.0-dev7-tls (i386/linux))..Max-Forwards: 70..Event: dialog;sl a..Contact: <sip:128.91.2.215:5060>..Subscription-State: active;expires=356 8..Content-Type: application/dialog-info+xml....<?xml version="1.0"?>.<dial og-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="15" state="full " entity="sip:helen@128.91.3.19:5060">.</dialog-info>. # U 75.56.10.116:5060 -> 128.91.2.215:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 128.91.2.215;branch=z9hG4bKdcf1.0c60f511.0 ..From: <sip:helen@128.91.2.215>;tag=10.810.1184176985.23..To: <sip:helen@1 28.91.3.19>;tag=18F7DEB-D3FFDEAA..CSeq: 2 NOTIFY..Call-ID: a7685c76-af0100e 1-7b4a7078@75.56.10.116..Contact: <sip:helen@75.56.10.116>..Event: dialog;s la..User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078..Content-Length: 0....

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[OpenSER-Users] OT: Asterisk mailing lists down
by Klaus Darilion 12 Jul '07

12 Jul '07

Hi! Am I the only one having problems with the Asterisk lists? thanks klaus

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[Serusers] Help needed
by Azeem Mehmood 12 Jul '07

12 Jul '07

Dear Sir/Madam How are you? I hope you are fine. Can anybody help me installing SER from scratch. With Best Regards Azeem Mehmood ( President) Oskar-Tahir Sp z o.o ul. Regatowa 3A 03-639 Warsaw Poland Phone : +48604826332 WebSite:- www.voip-prof.com Email: voip-prof(a)voip-prof.com

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[OpenSER-Users] OpenSER training course for admins
by Bogdan-Andrei Iancu 12 Jul '07

12 Jul '07

Hi, We have a good opportunity of organizing a one-day training course for OpenSER, specially for people looking for how to configure OpenSER. So, the targeted people are the OpenSER administrators that are looking to learn, via interactive sessions, based on script examples, how to configure different OpenSER features or services. We will try to mainly focus on the most complex and requested configuration cases, like: NAT traversal covering all cases Accounting - basic, extra accounting, multi-leg accounting General frame for routing (sequential request routing) Security (authentication, GW protection, blacklists) Integration to Asterisk (media services) Interdomain peering HA support (replication, GW redundancy) Now,about the course - as I said, it will be one day course and it will be hosted by the VoN Italy in Milan, on 26 September, from 9:30 to 18:00 hours. The package includes breakfast and lunch at the training location and the cost is estimated to be not more than 400 Euro per person. The event is organized in collaboration with Pulvermedia and right now we are looking to see how many people would be interested in this offer (technical side from openser.org and organization from Pulvermedia). Please feel free to contact me for any details/question. As I said, the first step is to see how many people are looking for such event, so let me know if you are interested in it. This survey will be ending Sunday 8th 24:00 CET. All this information will be posted on the openser project web page and forum. So, please give us the feedback regarding this opportunity to see if we go for it or not (also any suggestions are welcomed). Regards, Bogdan

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[Serusers] 3-way conferences
by inge 12 Jul '07

12 Jul '07

Hi all, With two incoming calls, I'm able to initiate a 3-ways conference with my SPA-941 using the internal functionality. But, for ATAs which don't provide this function, is it possible to do that with SER ? Thanks for your support Adrien .L

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[OpenSER-Users] Rewrite SDP for music of hold
by Iñaki Baz Castillo 12 Jul '07

12 Jul '07

Hi, I'm starting using OpenSer with Asterisk. In calls passing through Asterisk the caller or callee get music on hold of Asterisk when is put on hold. But of course, direct calls between OpenSer users don't get this music. I'm thinking in the posibility of detecting "on hold" reinvites (maybe reading the "a=sendonly" of SDP body) and rewriting the SDP IP contact to a music on hold server. I have no idea if this is possible. The first issue is how to set up a RTP music streamer for pointing there the SDP c header. Does "this" RTP music streamer exist? of course Asterisk is not a solution since I don't want to manage INVITE's, just rewrite the SDP. Second issue is: could be rewriting the SDP always sucessfull? what about cases with more than one RTP session as audio+video? So, is there any solution for offering music on hold with OpenSer? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] quick question regarding pattern matching
by Zahid Mehmood 11 Jul '07

11 Jul '07

Hi all, I'm still trying to get a handle on using regular expression in openser. I can't seem to find any documentation on openser wiki on regular expression usage. I have the following in my openser.cfg and a call to sip: 48911(a)ourdomain.com is logged as emergency call. What am I doing wrong? if (uri=~"^sip:(99)|(9911)|(45555)|((93)?911)@") { xlog ("L_INFO","Emergency call \n"); } I am using this as a work around: if ((uri=~"^sip:(99)|(9911)|(45555)|(93911)@") || (uri=~"^sip: 911@")) Thanks in advance for your help. -- Zahid

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[OpenSER-Users] Performance metric
by Pande, Piyush 11 Jul '07

11 Jul '07

Hi, Does anybody know what is the benchmark for calls/sec and number of simultaneous SIP calls that OpenSER can support?

1 0

Re: [Serusers] sl module functions in failure route
by Jiri Kuthan 11 Jul '07

11 Jul '07

At 16:34 11/07/2007, JF wrote: >Hi all, > >Is there any particular reason why sl module functions cannot be used >in failure routes? Yes. Failure_route is only invoked from within stateful processing. If a request was processed statelessly, SER does not really know if it failed. -jiri >I want to send a stateless reply in failure route. > >JF >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] Fw: Radius - prevent start record for ACK ?
by G.Jacobsen 11 Jul '07

11 Jul '07

Just discovered an acc parameter which does the trick and suppresses ACK accounting:modparam("acc", "report_ack", 0)Sorry for not reading properly.Still I am confused why radius receives start records with ACK before INVITE start records.CheersGerry----- Original Message ----- From: "G.Jacobsen" <g_jacobsen(a)yahoo.co.uk> To: <users(a)openser.org> Sent: Wednesday, July 11, 2007 8:06 PM Subject: Radius - prevent start record for ACK ? > I am using freeradius with openser 1.0.1. > > My problem is that openser is emitting TWO radius start records: One for the > INVITE with SIP method = 1 (INVITE) and also a second start record for SIP > Method = 4 (ACK) > > Is is normal that a start record is also emitted for the ACK ? > > As long as the INVITE record is sent before the ACK record that is no real > problem. > > However every-now-and-then (no pattern observed) Openser sends the Radius > start record with SIP Method = 4 (ACK) before the Record with SIP Method = 1 > for the INVITE. > > The problem is that the Radius record for the ACK does not contain the > username from the credentials so that the username is not correctly set in > the Radius record, instead of the correct username the callerid is used as > the username. > > How do I prevent that openser sends start records for the ACK ? Quite > frankly I do not understand why a start record for ACK is emitted since I > set the radius_flag only for INVITEs. > > the relevant parts from my configuration: > > modparam("acc", "radius_config", > "/usr/local/etc/radiusclient-ng/radiusclient.conf") > modparam("acc", "radius_flag", 1) > modparam("acc", "log_flag", 1) > modparam("acc", "service_type", 15) > > route{ > > ..... > # If this is an INVITE and not RE-INVITE start accounting > if(is_method("INVITE") && !has_totag()){ > # set the acc flags > setflag(1); > }; > > > if (is_method("ACK")) { #hop-by-hop ACKs are absorbed here > t_newtran(); > # drop broken ACKs > exit; > }; > > > Any hints are welcome. > > TIA > > Gerry > ___________________________________________________________ Inbox full of spam? Get leading spam protection and 1GB storage with All New Yahoo! Mail. http://uk.docs.yahoo.com/nowyoucan.html

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[OpenSER-Users] duplicate phplib_id_key
by TCB 11 Jul '07

11 Jul '07

I have just installed OSER v 1.2.1 and i can only add one entry into my subscriber table after that i get the following error when i run openserctl add .... I can add directly into the db with phppgadmin either . I;m using pgsql. ERROR: duplicate key violates unique constraint "phplib_id_key" any ideas ? -- TC

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[Serusers] nathelper timer error out of memory
by Kostas Marneris 11 Jul '07

11 Jul '07

Hello all, I have a setup with ser-0.9.6 on a x86-64 Debian Etch system with 8GB RAM. We use SIPP in order to stress test the system. As a first test we try to register about 150,000 users (the Registration requests arrival rate in not our concern at this time). We manage indeed to register 150,000 users and SER can stop/start with this location DB size. My setup concerning memory : config.h : #define PKG_MEM_POOL_SIZE 512*1024*1024 serctl file: STARTOPTIONS='-m 128' Although, at about 15,000-20,000 contacts at location DB, I get the following error, every 30 seconds : ERROR: nathelper::timer: out of memory natelper module config : modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) Searching the mailing list I found the following : http://lists.iptel.org/pipermail/serusers/2004-March/006828.html We actually changed the PKG_MEM_POOL_SIZE to 512*1024*1024 and recompile but we still get this error. thanks for any help, Kostas

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[OpenSER-Users] enum_query from failure route
by JF 11 Jul '07

11 Jul '07

Hi, Is there any particular reason why enum_query cannot be called from FAILURE_ROUTE? Thanks, JF

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[Serusers] sl module functions in failure route
by JF 11 Jul '07

11 Jul '07

Hi all, Is there any particular reason why sl module functions cannot be used in failure routes? I want to send a stateless reply in failure route. JF

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Re: [OpenSER-Users] DLG_count error
by [ACTIVENDY] NOC 11 Jul '07

11 Jul '07

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Re: [Serusers] How to configure Trunk capacity
by roayu 11 Jul '07

11 Jul '07

Hi, Klaus! Thanks for your reply. In that case, is there any software that can be integrated with SER to function as SIP trunking? Thanks. Best regards, Roa Yu -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Tuesday, July 10, 2007 2:34 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] How to configure Trunk capacity What do you mean by "trunk capacity"? The number of calls which may be sent to a certain gateway concurrently? If yes - you can't configure this in ser. ser is is transaction statefull, thus it does not count calls at all. Usually you just send the call to the gateway, and if the gateway is busy it will reponse with a certain response code. Then you configure ser to activate a failure to another gateway. regards klaus roayu wrote: > Hi, there! > > > > Can anyone tell me where can I configure the Trunk capacity for SER? > > > > Thank you. > > > > > > Best regards, > > Roa Yu > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Users] openser 1.1.1 available for several embedded platforms
by Ovidiu Sas 10 Jul '07

10 Jul '07

Hi, openser 1.1.1 is now available in binary format for several embedded patforms: - Linksys NSLU - Synology DS-101(g+) - Iomega NAS 100d - Freecom FSG-3 - Qnap TurboStation ... and all routers supported by the oleg or dd-wrt distributions. Check out http://www.nslu2-linux.org Regards, Ovidiu Sas

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[OpenSER-Users] mediaproxy module init error
by Rizwan Hisham 10 Jul '07

10 Jul '07

Hi all, im a new user of openser. how can i use mediaproxy found at www.ag-projects.com to destribute load on my 2 asterisk servers. I have configured a simple openser server which routes every request to asterisk. now i want to install another asterisk server and openser will be used to destribute calls between the 2. openser is running fine, i need to learn how mediaproxy works along with openser. i have installed the mediaproxy but the following error shows up when starting openser: 0(0) ERROR:mediaproxy:mod_init: Bad config - nat ping enabled, but not nat bflag set in usrloc module 0(0) init_mod(): Error while initializing module mediaproxy ERROR: error while initializing modules 0(0) INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill here are the modules and modules parameters: loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "xlog.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("rr", "enable_full_lr", 1) modparam("domain", "db_url", "mysql://openser:openser@192.168.0.7/openser") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "natping_interval", 60) I have no idea what above statements are doing excpet they are bothering me. i use the following link for help http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy -- Best Regards Rizwan Hisham Software Engineer AXVOICE Inc. www.axvoice.com

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[Serusers] ser mem usage
by Arun Kumar 10 Jul '07

10 Jul '07

HI I've noticed strange thing about my ser(0.9.6) day by day memory usage is getting increas and even after stopping my ser its not releasing the memory. thanks Arun

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[OpenSER-Users] radius authentication and local users database...
by liran tal 10 Jul '07

10 Jul '07

Hey everyone, I've been using the radius authentication just fine but I have a general question... Is it possible to configure it so that openser will first look if the user is found on the local users/registrar database and if not found it will then use the radius mechanism? So, is it possible to integrate both radius and local user lookup or it's one or the either? Thanks, Lir.

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[Serusers] Getting rid of SIP URI in Caller ID
by Charles Ulrich 10 Jul '07

10 Jul '07

Hi all, I'm fairly new to SER (just heard about it this afternoon), but I've already run up against a problem that's vexing me. We have a Polycom phone sitting behind a Linksys WRT54G router/firewall with DD-WRT firmware. This version of DD-WRT is the "voip" edition, meaning it has SER and rtpproxy installed to help ease NAT issues with SIP. No configuration of SER is necessary, we enter the WRT54G's IP address into the phone's outbound proxy field and it just works. The only drawback is that when using this method, the caller ID on an incoming call from the PSTN contains a SIP URI (in the form of sip:5551212@207.179.x.x) instead of a simple telephone number. Our end-users are pretty voip illiterate and thus haven't the slightest clue how VoIP works. They are only going to get confused when a SIP URI shows up on their caller ID instead of the actual PSTN phone number of the person who called them. How can this be remedied? I can't tell what version of ser is on this device because the -h and -V flags return nothing. (Probably to make it slightly smaller for the embedded platform.) The 'strings' utility wasn't helpful either. Attached is the ser.cfg that comes with DD-WRT. Any help is greatly appreciated. Let me know if I need to provide more information. Thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com

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[OpenSER-Users] next_gw() function
by Bruno Machado 10 Jul '07

10 Jul '07

Hi all. I'm using Openser 1.1.0 and I'm trying to create a failure route using LCR. All the calls must be routed to the same IP, but when the OpenSER execute the next_gw(), I receive the error "503 Service not available, no gateways found". When I change the IP column gw.ip_addr, it works fine. I think that is not possible to route the messages through the same IP. But I need it! All of my calls must to pass through this IP. openser ----> proxy B ----> the main GW (isn't answering) Using the failure route: openser ----> proxy B ----> the secundary GW (is answering) This 'proxy B' is important and I can't remove it at all. So, how can I use the next_gw() or another function to route the call throught it? There are a parameter that I can set to help? I wouldn't like to use a static route wrote in the script file. And the Openser's version 1.2? I didn't try it yet. Is it possible using this version? Any comments are welcome. Thanks a lot. Bruno Machado ____________________________________________________________________________________ Novo Yahoo! Cadê? - Experimente uma nova busca. http://yahoo.com.br/oqueeuganhocomisso

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