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[Users] Openser generated strange SIP packets with LCR Fallback on a second gateway
by mailinglist-openser-users 07 May '07

07 May '07

gallois:/var/log# sudo /usr/bin/ngrep -W byline "" "port 5060" |more filter: (ip or ip6) and ( port 5060 ) # U phone:5060 -> openser:5060 INVITE sip:xxxxxxxx@sip1.e-qual.fr SIP/2.0. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 70. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U openser:5060 -> phone:5060 SIP/2.0 100 Giving a try. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. To: <sip:xxxxxxxx@sip1.e-qual.fr>. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Server: OpenSER (1.2.0-notls (i386/linux)). Content-Length: 0. . # U openser:5060 -> gateway1:5060 INVITE sip:xxxxxxxxx@gateway1:5060;transport=udp SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.0. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U gateway1:5060 -> openser:5060 SIP/2.0 488 Not acceptable here. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.0;received=openser. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. To: <sip:xxxxxxxx@sip1.e-qual.fr>;tag=as714f5f56. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:xxxxxxxxx@gateway1>. Content-Length: 0. . # U openser:5060 -> gateway1:5060 ACK sip:xxxxxxxxx@gateway1:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.0. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. To: <sip:xxxxxxxx@sip1.e-qual.fr>;tag=as714f5f56. CSeq: 1 ACK. User-Agent: OpenSER (1.2.0-notls (i386/linux)). Content-Length: 0. . # U openser:5060 -> gateway2:5060 INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U openser:5060 -> gateway2:5060 INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U openser:5060 -> gateway2:5060 INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv. # U openser:5060 -> gateway2:5060 INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0. Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>. Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone. Contact: <sip:phone>. Content-Length: 221. Content-Type: application/sdp. CSeq: 1 INVITE. From: <sip:sip1.e-qual.fr>;tag=c0a80843-16. Max-Forwards: 69. Session-GUID: 875967334-858992696-842282596-875782963. Supported: 100rel. To: <sip:xxxxxxxx@sip1.e-qual.fr>. User-Agent: Quintum/1.0.0. Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1. Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c. Quintum: 0c01030b0237370501001301000f0b413031322d323032334332. P-hint: outbound PSTN. . v=0. o=Quintum 17 17 IN IP4 phone. s=VoipCall. c=IN IP4 phone. t=0 0. m=audio 10274 RTP/AVP 4 101. c=IN IP4 phone. a=rtpmap:4 g723/8000/1. a=ptime:30. a=rtpmap:101 telephone-event/8000/1. a=sendrecv.

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[Users] lcr: load_gws(): from_uri to large
by Papadopoulos Georgios 07 May '07

07 May '07

Hello all, I get the following error from lcr module load_gws(): from_uri to large This happens when the rpid is exactly 128 characters long. If I make it 127 long it works fine. This is on the 1.1.x branch. Has anyone else noticed this? thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[Users] Openser used to route call between PSTN to/from remote voice gateways with routing different in each direction ( originate from PSTN or originate from remote voice gateway )
by mailinglist-openser-users 07 May '07

07 May '07

Version: Openser 1.2.0 Platform: PSTN <-> Central Gateways <-- OpenSer --> Remote Gateways <-- >PABX We have 4 central Voice Gateway connected to local PSTN and several tens of customer remote voice gateways connected to PABX. On our central voice gateways, we have several PSTN provider, and these providers have different prices for different location ( so calling a cell phone should be rather made by a voice gateway connected to a particular PSTN provider ). As well, we provide SDA numbers for incoming call from PSTN on our central gateway and that should be routed to a remote customer voice gateway. But if customerA call the SDA of customerB, customerA should go through PSTN by our central voice gateway and then our central voice gateway route the call to CustomerB. So all remote gateways should not know any of the SDA route, only our central gateway should use this routes. So basically we have 2 kinds of routing to be done by OpenSer: o For our Central Gateway - For SDA11, try gateway Customer1 - For SDA12, try gateway Customer1 - For SDA21, try gateway Customer2 - For SDA22, try gateway Customer2 - For any other unknown number, no gateway should be returned ( I don't want a wrong number routed to our gateway from PSTN to generate another call on PSTN ) o For our remote customer gateways, I should have routing like this: - for cell phone, try gateway 1, if that gateway1 fail then try randomly gateway 2,3 or 4 ( and if possible the less loaded of these 3 gateways ) - for spain try gateway2, if that gateway2 fail, then try randomly gateway 1,3 or 4 ( and if possible the less loaded of these 3 gateways ) - for all other routes, try randomly gateway 1,2,3,4 ( and if possible the less loaded of these 4 gateways ) So I have several constraint: - two kind of routing should be made either if the call is coming from the central gateway or from a remote customer gateway ( differenlty said: routing different if call originate from PSTN or originate from remote voice gateway ) - for customer remote gateways, there are some preferred central gateway for certains location (e.g international call ) but if these gateways fail, fallback should occur to others central gateway - the less is hardcoded in the configuration file, the better is is. - how to try first the currently less loaded gateway when several gateways are available ( is it possible in openser has it is only dialog statefull and not call statefull ? ) What is the best option in OpenSer to achieve this ? Is the LCR module with "from_uri" and route by "prefix" can handle all this ? Where can we find complex examples of the LCR module ? Regards, Jerome. -- Jerome

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[Users] do I need openser FIFO?
by Joao Pereira 07 May '07

07 May '07

Hello I'm using SER/OpenSER for a long time and I always used the FIFO. My question is, do I really need to use the FIFO? What OpenSER functionalities rely in the FIFO? Just Serweb? Thanks regards Joao Pereira -- ______________________________________________ João Gomes Pereira FCCN Av. do Brasil, nº 101 1700-066 Lisboa tel: +351 218 440 100 - fax: +351 218 472 167 email|SIP: joao.pereira(a)fccn.pt http://www.fccn.pt ______________________________________________ --- Aviso de Confidencialidade Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha recepcionado indevidamente esta mensagem, solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218 440 100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218 440 100 and delete it immediately. ---

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[Serusers] SER/PSTN Gateway conf
by flavio 07 May '07

07 May '07

Hi to all, I'm newbie for SER and I'm testing my own configuration to send/receive calls through my pstn gateway, according to follow simple scenario: Phone --> PSTN --> Gateway --> SER --> SIP Phone I send and receive calls successfully, but when Phone over PSTN hangs up SER reply to BYE message with 404 Not Found.Here is my ser.cfg: debug=3 fork=no log_stderror=yes listen=10.28.19.202 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://root@10.28.19.202/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://root@10.28.19.202/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/usr/local/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/usr/local/etc/ser/permissions.deny") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("16")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { # if ((method=="INVITE" || method=="REFER") && !has_totag()) { # sl_send_reply("403", "Forbidden"); # break; # }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(3); route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:0[0-9]*@") { # calls to PSTN #strip(0); route(4); break; }; if(uri=~"^sip:3[0-9]*@") { calls to Mobile route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; route(1); } route[4]{ #----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("10.28.52.105"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(1); } Any suggestions about my issue? Thanks 4 all, Flavio -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

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[Users] HOWTO submit a patch in a correct way?
by Peter Lemenkov 06 May '07

06 May '07

Hello All! I need a simple feature in OpenSER - parameter in Jabber-module to set up jabber's priority. So I took a look into the sources and created a simple patch (attached). Are there any formal procedures that I must follow for accepting my patch? -- With best regards!

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[Users] E1 config for chile
by laurent schweizer 04 May '07

04 May '07

Hello, I try to configure a Patton smart node with a E1 for the Chile. I need to know wich parameters I must set for Chile. If someone have informations it's welcoms. Laurent

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[Users] Passing predefined string-value to RADIUS accounting server
by Peter Lemenkov 04 May '07

04 May '07

Hello All! I've encountered some difficulties. I need to pass predefined string value ("REALM" actually) to accounting RADIUS server. I want to did it in way similar to: modparam("acc", "radius_extra", "User-Name=$fU; Sip-User-Realm=$ar") But I need to set realm for all radius-accounting requests to some value ("MYREALM" for example). I tried set $ar explicitly before accounting this way $ar = "MYREALM"; but failed with error "invalid left operand in assignment". Does $ar read-only variable? Another my try was to pass new realm value in module configuration section: modparam("acc", "radius_extra", "User-Name=$fU; Sip-User-Realm=MYREALM") It failed with message "parse failed in <Sip-User-Realm=MYREALM> around position ..." I can't set up custom variable before route{} sections so I can't define my $var(realm) into module configuration line. Are any ways to solve my problem exists? -- With best regards!

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[Users] forcing tcp transport
by Will Quan 04 May '07

04 May '07

I am setting up the proxy so the transport protocol is forced to tcp when packet size is > 1300b. The problem is when the originator abandons the call, the CANCEL msg sent over UDP and I really want it to use the TCP transport, as the INVITE did. Is there a way to achieve this?--will

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[Users] OpenSER 1.2 and Flatstore Accounting
by Brandon Armstead 04 May '07

04 May '07

Hello my fellow Flatstore Accounting users, There seems to be some issues with Flatstore and Accounting in OpenSER 1.2, it does not seem to properly account/log extras? Perhaps I am incorrectly using it, but the syntax still remains modparam("acc", "log_extra", "foo=$pvar"). Is there anything I'm missing? Thanks! Regards, Brandon.

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