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[OpenSER-Users] cancel going to 127.0.0.1
by Richard Bennett 16 Nov '07

16 Nov '07

Hi, I seem to remember reading about this problem before, but couldn't find any pointers by searching... On the following setup with Openser 1.1: UA -> Openser/Mediaproxy -> PSTN-Gateway most calls pass just fine and cancel fine, but some don't stop properly. If the UA cancels the call the cancel goes: gateway ip= 1.2.3.204 proxy ip= 1.2.3.201 ua ip = 81.1.2.223 U 81.1.2.223:5071 -> 1.2.3.201:5060 CANCEL sip:0475711111@sipc.example.com SIP/2.0. Via: SIP/2.0/UDP 192.168.1.11:5071;branch=z9hG4bK-2f6ae216. From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0. To: <sip:0475711111@sipc.example.com>. U 1.2.3.201:5060 -> 127.0.0.1:5060 CANCEL sip:0475711111@sipc.example.com SIP/2.0. Record-Route: <sip:1.2.3.201;lr=on;ftag=a27b4fe18ca60f4ao0>. Via: SIP/2.0/UDP 1.2.3.201;branch=z9hG4bK7f9f.b68d8185.0. Via: SIP/2.0/UDP 192.168.1.11:5071;received=81.1.2.223;branch=z9hG4bK-2f6ae216. From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0. To: <sip:0475711111@sipc.example.com>. I 127.0.0.1 -> 1.2.3.201 3:3 ....E..o..@.@...Y............[h.CANCEL sip:0475711111@sipc.example.com SIP/2.0. Record-Route: <sip:1.2.3.201;lr=on;ftag=a27b4fe18ca60f4ao0>. Via: SIP/2.0/UDP 1.2.3.201;branch=z9hG4bK7f9f.b68d8185.0. Via: SIP/2.0/UDP 192.168.1.11:5071;received=81.1.2.223;branch=z9hG4bK-2f6ae216. From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0. To: <sip:0475711111@sipc.example.com>. And then it keeps looping between 127.0.0.1 and 1.2.3.201. For some reason openser sends the CANCEL to the loopback interface instead of on to the gateway. Is this a recognizable problem or should I post a full trace and config file? Thanks for looking, Richard.

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[OpenSER-Users] Newbie issue: Cannot exec HelloWorldServlet on WeSIP
by Alvaro Mari Diego 16 Nov '07

16 Nov '07

Hi everyone, I'm new to WeSIP, so I've got stuck pretty fast, as expected :) I got WeSIP running on OpenSER 1.2.0. I'm trying to run a HelloWorld sip servlet to test WeSIP, which responses a 200 OK and a MESSAGE request saying "Hello, World", to any incoming request to a sip:greetings@amd-openims.test URI. I mounted my sar file, and installed it in wesip webapps with the web manager application, and got WeSIP properly connected to OpenSER' seas. Openser is configured to route any MESSAGE request to WeSIP AS. But when I try to send a message to sip:greetings@amd-openims.test from a client logged as alice(a)amd-openims.test, WeSIP console displays this: ******************************************************************** SeasMessageEvent [SeasRouter] - New message received, from 138.4.7.163:6060 to 138.4.7.163:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 54974,label=620758046 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 138.4.7.163/6060 ExpressMessageChannel [ExpressMChannel[0]] - About to process MESSAGE sip:greetings@amd-openims.test SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< MESSAGE sip:greetings@amd-openims.test SIP/2.0 Route: <sip:138.4.7.163:5080;lr=on>,<sip:iscmark@scscf.amd-openims.test:6060;lr=on;s=1;h=0;d=0> Via: SIP/2.0/UDP 138.4.7.163:6060;branch=z9hG4bKeb6d.ff7360e1.0,SIP/2.0/UDP 138.4.7.163:4060;branch=z9hG4bKeb6d.000ef08.0,SIP/2.0/UDP 138.4.7.149:63926;branch=z9hG4bK-d87543-b370c6252d6ad226-1--d87543-;rport=63926 Max-Forwards: 14 To: "Servicio HelloWorld" <sip:greetings@amd-openims.test> From: "alice" <sip:alice@amd-openims.test>;tag=737cbe08 Call-ID: Y2FiYmY0MmM2NjFhYzU5NWUzNTE2MmM5MWY2ZjdhMGI. CSeq: 2 MESSAGE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,MESSAGE,SUBSCRIBE,NOTIFY,REFER Content-Type: application/unknown? User-Agent: X-Lite release 1011s stamp 41150 P-Charging-Vector: icid-value="P-CSCFabcd4739818c00000007";icid-generated-at=138.4.7.163;orig-ioi="amd-openims.test" Content-Length: 262 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:54974 label:620758046 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bKeb6d.ff7360e1.0] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 138.4.7.163 mapped to: ConvergedEngine[WeSIP_engine].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestMESSAGE, trying match against Rules ConvergedHostValve [SipProcessor[4]] - No servlet-mapping matched this Request. SipResponse [SipProcessor[4]] - >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 138.4.7.163:6060;branch=z9hG4bKeb6d.ff7360e1.0,SIP/2.0/UDP 138.4.7.163:4060;branch=z9hG4bKeb6d.000ef08.0,SIP/2.0/UDP 138.4.7.149:63926;branch=z9hG4bK-d87543-b370c6252d6ad226-1--d87543-;rport=63926 Max-Forwards: 14 To: "Servicio HelloWorld" <sip:greetings@amd-openims.test> From: "alice" <sip:alice@amd-openims.test>;tag=737cbe08 Call-ID: Y2FiYmY0MmM2NjFhYzU5NWUzNTE2MmM5MWY2ZjdhMGI. CSeq: 2 MESSAGE Content-Length: 0 LingerList [SipProcessor[4]] - scheduled transaction z9hG4bKeb6d.ff7360e1.0 for deletion ExpressMessageChannel [SipProcessor[4]] - sending with spiral=false SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] ActionRouter$ActionSender [ActionSender] - Writing 509 bytes into Action Socket SeasHashTable [ExpressMChannel[0]] - Removing hashIdx:54974 label:620758046 from hashTable SeasHashTable [ExpressMChannel[0]] - Transaction with id:z9hG4bKeb6d.ff7360e1.0, hash_idx:54974, label:620758046 removed from the Seas Hash Table ExpressMessageChannel [ExpressMChannel[0]] - Done processing MESSAGE sip:greetings@amd-openims.test SIP/2.0 ******************************************************************** And, as I feared, got a 481 response in my client. I don't know what I'm doing wrong. I wonder it could be a misconfiguration in the WeSIP server.xml, or maybe something wrong in my servlet. I attach you in a zip my HelloWorld Servlet, openser.cfg file, and wesip server.xml file. Thanks in advance, Cheers, -- Alvaro

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[OpenSER-Users] ERROR: init_io_wait: could not alloc epoll array
by Ariadne Ramos 16 Nov '07

16 Nov '07

I have Openser 1.1.1 installed and working fine for long time. But now when I restart the service openser with no changes made in configuration this doesn't start. I have to restart many times to get the service up again. The error that appear in logs is the following: Nov 15 11:32:06 prxc /usr/local/sbin/openser[22658]: ERROR: init_io_wait: could not alloc epoll array Nov 15 11:32:06 prxc /usr/local/sbin/openser[22658]: ERROR: tcp_receive_loop: exiting... Do you know why this is happening? I appreciate your help Thanks in advanced Ariadne

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[OpenSER-Users] q-value
by Daniel 16 Nov '07

16 Nov '07

Hi, Is it possible to change q parameter from Client side? I'm using xlite and Sjphone, but I don't be able to change this value so, Proxy, by default, set as 1 every q-value. any clue? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] [OT] Do ITSP SIP_PSTN gateways allow REFER?
by Iñaki Baz Castillo 16 Nov '07

16 Nov '07

Hi, I will connect my OpenSer with a Nortel (SIP to PSTN gateway) for incoming/outgoings call to/from PSTN. I have a doubt: if an OpenSer user (200(a)mydomain.org) receives an incoming call from gateway, could he REFER the call to 201(a)mydomain.org? This is: do ITSP gateways allow REFER to SIP users subcribers to my OpenSer? I understand they could not accept REFER's to PSTN number because accounting issues (gateway UAC receving a REFER and calling to PSTN?). Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[Serusers] rtpproxy and rtcp
by Miles Scruggs 15 Nov '07

15 Nov '07

Just wondering if there is any way to collect rtcp information from rtpproxy? Thanks Miles

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[OpenSER-Users] attention!!! -- mailing list domains
by Daniel-Constantin Mierla 15 Nov '07

15 Nov '07

Hello, as I see that some send emails to xyz(a)lists.openser.org and cc to xyz(a)openser.org. I want to make sure that everybody understands that only the new domain "lists.openser.org" MUST be used now. First is confusing and, also, we do not need duplicates. If you get problems with lists.openser.org, please report. The old domain "openser.org" for mailing lists will stay in place about one more month from now on, to be sure that most of discussion threads started before migration get closed. The reason at migration time was not to create chaos by not being able to reply, or not having enough time to learn about the new domain. The old mailing lists domains are just aliased to the new ones, the messages end in the same archive, same server. Thanks, Daniel

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[OpenSER-Users] Permissions module addition
by Daryl G. Jurbala 15 Nov '07

15 Nov '07

I'm looking for an addition to the permissions module. I'm using the address table to match valid originators. Each originator needs to be placed into a specific route tree (in carrierroute). Right now, I can do something like: $var(orig) = "unauth"; if (allow_source_address("<fist route tree>")) { $var(orig) = "<first route tree>" } if (allow_source_address("<second route tree>")) { $var(orig) = "<second route tree>" } <etc., etc. for each route tree I have> if $var(orig)=="unauth" { sl_send_reply("403", "Unauthorized IP"); exit; } else { ....process the call..... What I'd like to see is a permissions function that would allow me to do something like: $var(orig) = allow_source_address_group() if $var(orig)=="-1" { sl_send_reply("403", "Unauthorized IP"); exit; } else { ....process the call..... Where the proposed allow_source_address_group() function would search all IPs in the address table and return either a -1 if no matching address is found, or return the value of the group column if one is found. Anyone out there willing to make these changes? If so, please contact me. I'm willing to put a bounty on this. Thanks, Daryl G. Jurbala Director of Network Operations Global Convergence Solutions Tel: +1 732 853 0513

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[Serusers] ACK Problem : Not routed to NAT'd port
by Ricardo Martinez 15 Nov '07

15 Nov '07

Hello list. I'm facing a problem with my SER configuration and a call to a NAT'd endpoint. When I call a NAT'd endpoint registered in my SER proxy, the call is established, but the last ACK message is not properly routed by SER. You can see this in the next messsage flow. 100.100.100.50 : Callee 100.100.100.99 : SER 100.100.100.231 : Called U 100.100.100.50:5060 -> 100.100.100.99:5060 INVITE sip:008991000404@sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 100.100.100.50;rport;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000010f. Contact: <sip:551100000121@100.100.100.50;user=phone>. U 100.100.100.99:5060 -> 100.100.100.50:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. U 100.100.100.99:5060 -> 100.100.100.231:62447 INVITE sip:551100000684@100.100.100.231;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000121@100.100.100.50;user=phone>. U 100.100.100.231:62447 -> 100.100.100.99:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0,SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. (The call is answered) U 100.100.100.231:62447 -> 100.100.100.99:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0,SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.99:5060 -> 100.100.100.50:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.50:5060 -> 100.100.100.99:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Via: SIP/2.0/UDP 100.100.100.50;rport;branch=z9hG4bK404c94810000005a43b2b7f00000182a00000112. Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. U 100.100.100.99:5060 -> 100.100.100.231:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=0. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7f00000182a0000 0112. U 100.100.100.231:62447 -> 100.100.100.99:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0,SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.99:5060 -> 100.100.100.50:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.50:5060 -> 100.100.100.99:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Via: SIP/2.0/UDP 100.100.100.50;rport;branch=z9hG4bK404c94810000005a43b2b7f000001d6e00000114. Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. . U 100.100.100.99:5060 -> 100.100.100.231:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=0. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7f000001d6e0000 0114. (.........) (Timeout and the call is released) U 100.100.100.231:62447 -> 100.100.100.99:5060 BYE sip:551100000121@100.100.100.50;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.20;rport;branch=z9hG4bKc0a801140000002643b2a9be000036e10000001f. Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. . U 100.100.100.99:5060 -> 100.100.100.50:5060 BYE sip:551100000121@100.100.100.50;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=93794163011;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bK15c7.924c3535.0. Via: SIP/2.0/UDP 192.168.1.20;received=100.100.100.231;rport=62447;branch=z9hG4bKc0a801140000 002643b2a9be000036e10000001f. U 100.100.100.50:5060 -> 100.100.100.99:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bK15c7.924c3535.0,SIP/2.0/UDP 192.168.1.20;rport=62447;received=100.100.100.231;branch=z9hG4bKc0a801140000 002643b2a9be000036e10000001f. U 100.100.100.99:5060 -> 100.100.100.231:62447 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.20;rport=62447;received=100.100.100.231;branch=z9hG4bKc0a801140000 002643b2a9be000036e10000001f. I don't understand why the the ACK message is routed to port 5060 and not to the port 62447 like the INVITE or the final OK. Is this maybe a configuration problem?. My ser.cfg is based on the onsip.org configuration documents plus some modifications by me. Can someone help me here Thanks in advance Ricardo Martinez.-

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[OpenSER-Users] Illegal characteres in SIP URI?
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, could somebody tell me which characteres are allowd in a SIP URI? which RFC defines it? I ask it because I do a DB query with $ru so a SQL injection it's possible if RURI contains single or double ' So at the begining of the script I'd like to reject a message if the RURI contains illegal symbols. Could be useful a core function for this? Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] Aliases table and db aliases
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

El Thursday 15 November 2007 16:08:27 escribió: > *Hi,* Better replies to the mailist ;) > Iñaki Baz Castillo wrote: > > * alias with DB_ALIASES module: > > > > table dbaliases: > > alias_username: 1234 > > alias_domain: IP_PROXY > > username: bob > > domain: IP_PROXY > > > > db_aliases_lookup() will rewrite URI when it's 1234@IP_PROXY and it'll > > become bob@IP_PROXY > > > > Later lookup("location") will look for the contact of that AOR. > > *Ok. > db_aliases talble is filled in by myself through "openserctl db_alias > add" command.* > > > * alias with ALIASES table: > > > > table aliases: > > username: 1234 > > domain: IP_PROXY > > contact: sip:bob@IP_PROXY > > > > lookup("aliases") will rewrite URI when it's 1234@IP_PROXY and it'll > > become bob@IP_PROXY > > > > Later lookup("location") will look for the contact of that AOR. > > *so, is ALIASES table also created by myself through "openserctl" > command*s ? No idea. In fact I fill these tables via SQL. > > ok now? > > *ok, thank you very much. > daniel > * Regards. -- ilimit... *Iñaki Baz Castillo* ibc(a)in.ilimit.es ÀREA SISTEMES 0034 937 333 375 VOLTA 1, PIS 5 08224 TERRASSA.BCN Aquest enviament és confidencial i està destinat únicament a la persona a qui s'ha enviat. Pot contenir informació privada sotmesa al secret professional, la distribució de la qual està prohibida per la legislació vigent.

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Re: [OpenSER-Users] Aliases table and db aliases
by Daniel 15 Nov '07

15 Nov '07

Thanks. But I don't understand the use of aliases table. I have defined some subscriber users, with username (like BOB) and password (BOBPWD). So, if I wanted to access the system, I would send a REGISTER message with correct username and password (authorization required). Then my proxy address will be "sip:BOB@IP_PROXY" and my Contact will be "sip:BOB@MY _IP_ADRRESS". These information are saved into USER LOCATION table (save("location") function). Now, through DB_aliases module and dbaliases table, I can define some aliases for BOB (like "sip:1234@IP_PROXY", etc...), so if an INVITE for bob has "sip:1234@IP_PROXY" as R-URI, with db_aliases_lookup() function I can rewrite R-URI with Bob's URI ("bob@IP_PROXY") and then call the lookup("location") function. But I don't understand how I can use the ALIASES table. Are there any example explaining its use ? Sorry. Regards, daniel Iñaki Baz Castillo wrote: El Thursday 15 November 2007 14:31:52 Daniel escribi�: Hi all, what is the different between Aliases table and DB aliases table? What is their use ? "aliases" table is same as "location" table but it should be used to look for alias with the function lookup("aliases"). Of course, aliases are simpler than the complex location or aliases tables that manage more info about users, so in order to implement alias system in a simpler way the module ALIAS_DB exists and its table dbaliases. Note that dbaliases table is much simpler (it's just valid for aliases), Look at the module doc: http://www.openser.org/docs/modules/1.3.x/alias_db.html -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[Serusers] Implicit Registration Set
by Magnus Sörman 15 Nov '07

15 Nov '07

Hi, How do you/could you configure SER to use implicit registration set? If e.g. a user with a sip URI registers, then both that address and his tel-uri should be registered. Thanks, /Magnus Sorman

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[OpenSER-Users] Aliases table and db aliases
by Daniel 15 Nov '07

15 Nov '07

Hi all, what is the different between Aliases table and DB aliases table? What is their use ? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] Presence module help
by Stanisław Pitucha 15 Nov '07

15 Nov '07

Hello Can someone tell me couple of details about presence code? I.e.: 1. There's PUBL_TYPE and WINFO_TYPE define. They are used as flags, but then they are just marking winfo / other packages. What's the effect of PUBL_TYPE on event package exactly? 2. What is the meaning of 'get_rules_doc', 'apply_auth_nbody', 'get_auth_status' funcs? Unfortunately I wasn't able figure it out from source completely. Thanks Stanisław Pitucha

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[OpenSER-Users] OpenSER with multidomain configuration
by satish patel 15 Nov '07

15 Nov '07

Dear all I am going to install openser with multidomain support do anyone have sample .cfg file or configuration guide i have find many document but it is very hard to implement so just suggest me how to test and configure it ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Never miss a thing. Make Yahoo your homepage.

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[Serusers] SER Dispatcher Module
by Arun Kumar 15 Nov '07

15 Nov '07

Hi Here is my setup Users ----> SER ----> Asterisk ----> PSTN works fine. when ever user dial able to hear balance message and calls goes fine. Just now I replaced my setup to use Dispatcher module Users -----> SER (Dispatcher module) ----------> Asterisk Boxes -----> PSTN when user makes calls not able to hear balance message all the times but able to make PSTN calls ? In my config I just changed replace IP of my asterisk box with dispatcher config: Old config: strip(1); rewritehostpost("x.x.x.x:5060"); route(4); NAT Detet. New conifg: strip(1); ds_select_dst("1", "0"); forward(uri:host, uri:port); route(4); dispatcher file: 1 sip:x.x.x.x:5060 1 sip.y.y.y.y:5060 Need help on above. Thanks, Arun

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[Serusers] DTMF Problem
by Arun Kumar 15 Nov '07

15 Nov '07

Hi Here is my setup: USER ------> PSTN -----> Asterisk A ----> IAX2 Trunk ----> Asterisk B ---------> SER --------> Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun

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[OpenSER-Users] MyISAM or InnoDB in MySQL tables?
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, MySQL tables of OpenSer created by the script are "MyISAM". Is there any reason for that instead of using InnoDB? As I know: MyISAM: - Is faster reading and writting (except writting various entries to same table since it blocks the entire table). - No transactions or rollbacks. - Small disc space requeriments. InnoDB: - Transactions, rollbacks, restrictions in foreing keys. - Faster writting various entries in same table (entry lock instead of table lock). - More disc space and RAM requeriments. Yes, it can be obvious that for OpenSer the engine MyISAM can be a good choice but I would like to know if there is more data for this choice. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [Serusers] Voicemail setup on SER v0.9.6
by roayu 15 Nov '07

15 Nov '07

Thanks Stefan! I need to confirm that after the installation, I will be having 2 SERs + 1 SEMS, rite? ( 1 SER for normal sip, 1 SER as SIP stack, 1 SEMS) Thanks. Roa Yu -----Original Message----- From: Stefan Sayer [mailto:stefan.sayer@iptego.de] Sent: Tuesday, November 13, 2007 6:27 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 Hello, roayu wrote: > Which version of SEMS should I download? Is it included the SER as well? > I would prefer a stable version =) I would recommend the stable SEMS 0.10.0 release. Bundled with its SER its available at ftp://ftp.iptel.org/pub/sems/sems-0.10.0-bundle.tar.gz You will find all information at http://iptel.org/sems. Regards Stefan Sayer > > Thanks. > > Roa Yu > > -----Original Message----- > From: SIP [mailto:sip@arcdiv.com] > Sent: Monday, November 12, 2007 10:53 AM > To: roayu > Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 > > You CAN have those on the same box. Although it's not required. You > could have one SER on one box and then the other SER/SEMS on another box > of you preferred. > > > > roayu wrote: >> Hi Eneref, >> >> Thanks for your reply. So, do u mean that I'm gonna have the follwing >> applications in the SAME box? >> - 2 SER servers ( 1 existing SER + 1 new SER for SEMS ) with different > port >> - 1 SEMS >> >> Thanks. >> >> Roa Yu >> >> >> -----Original Message----- >> From: Eneref [mailto:eneref@arcdiv.com] >> Sent: Friday, November 09, 2007 4:14 PM >> To: roayu >> Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 >> >> Yes. You need either SEMS or Asterisk or some such. Meshing SEMS with >> SER, though it LOOKS complex, is really not that bad. >> >> The easiest thing to do is to run another SER server to handle voicemail >> (on a different port from the first SER if it's on the same box) and >> forward messages to it if voicemail is required (on time outs, or >> whatever conditions you set). There are some good instructions on how >> to do that in the SEMS docs. >> >> This also makes it quite easy to swap out with Asterisk or any other >> sort of voicemail-capable server should the need arise in future. >> >> N. >> >> roayu wrote: >> >>> Hi there! >>> >>> >>> >>> Could anyone let me know how to setup Voicemail on SER v.0.9.6 ? Do I >>> need to install SEMS as well? >>> >>> >>> >>> Thanks >>> >>> >>> >>> >>> >>> Regards, >>> >>> Roa Yu >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[OpenSER-Users] pua_usrloc
by Venter, Andries <andries＠sun.ac.za> 15 Nov '07

15 Nov '07

listen=udp:196.211.9.222 port=5065 loadmodule "presence.so" loadmodule "avpops.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "domain.so" loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_usrloc.so" modparam("usrloc|auth_db|uri_db|acc", "db_url", mysql://openser:password@localhost/openser") modparam("presence|pua", "db_url", "mysql://openser:password@localhost/openser") modparam("pua_usrloc", "default_domain", "196.211.9.222") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:196.211.9.222:5065") if( is_method("PUBLISH|SUBSCRIBE")) { route(4); } if (method=="REGISTER") { xlog("L_INFO", "Kyk of ek moet PUBLISH........................\n"); if(!search("^User-Agent: X-Lite")) { xlog("L_INFO", "JA - ek moet PUBLISH........................\n"); pua_set_publish(); } save("location"); exit; }; # presence handling route route[4] { xlog("L_INFO","Route 4........................................\n"); sl_send_reply("100","trying"); # absorb retransmissions if (! t_newtran()) { xlog("L_INFO", "Not new transaction.................................\n"); sl_reply_error(); exit; }; # append_to_reply("Contact: <sip:196.211.9.222:5065>\r\n"); if(is_method("PUBLISH")) { xlog("L_INFO", "Handling PUBLISH.................................\n"); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "Handling SUBSCRIBE.................................\n"); handle_subscribe(); t_release(); }; exit; }

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Re: [OpenSER-Users] pua_usrloc
by Venter, Andries <andries＠sun.ac.za> 15 Nov '07

15 Nov '07

My config is as follows listen=udp:196.211.9.222 port=5065 loadmodule "presence.so" loadmodule "avpops.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "domain.so" loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_usrloc.so" modparam("usrloc|auth_db|uri_db|acc", "db_url", mysql://openser:password@localhost/openser") modparam("presence|pua", "db_url", "mysql://openser:password@localhost/openser") modparam("pua_usrloc", "default_domain", "196.211.9.222") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:196.211.9.222:5065") if( is_method("PUBLISH|SUBSCRIBE")) { route(4); } if (method=="REGISTER") { xlog("L_INFO", "Kyk of ek moet PUBLISH........................\n"); if(!search("^User-Agent: X-Lite")) { xlog("L_INFO", "JA - ek moet PUBLISH........................\n"); pua_set_publish(); } save("location"); exit; }; # presence handling route route[4] { xlog("L_INFO","Route 4........................................\n"); sl_send_reply("100","trying"); # absorb retransmissions if (! t_newtran()) { xlog("L_INFO", "Not new transaction.................................\n"); sl_reply_error(); exit; }; # append_to_reply("Contact: <sip:196.211.9.222:5065>\r\n"); if(is_method("PUBLISH")) { xlog("L_INFO", "Handling PUBLISH.................................\n"); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "Handling SUBSCRIBE.................................\n"); handle_subscribe(); t_release(); }; exit; }

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[OpenSER-Users] pua_usrloc
by Venter, Andries <andries＠sun.ac.za> 15 Nov '07

15 Nov '07

I would like to get presence published for all phones. I call pua_set_publish() before doing save ("location"), but no PUBLISH messages are generated when the REGISTER request comes through. I am using OpenSER 1.2.2. Any ideas?

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[Serusers] tcp branching errors
by Tomasz Zieleniewski 15 Nov '07

15 Nov '07

Hi, I have the following situation. I have two ser rel_2_0_0 instances one acting as a boundary proxy the other on as internal proxy (registrar, location). I have the following situation I have a user which has registered two contacs one woth tcp transport the other one with udp. All messages coming to and outgoing from my network travels through boundary sip proxy. When internal ser perform lookup_location it get to contacts and the destination set contains two records: Nov 15 11:03:13 sen ser[3673]: route[FORWARD]: destination set: Contact: sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP, sip:xlite@192.168.0.164:7060 Then internal ser forward message to boundary SIP proxy by changing the destination of the message and with the usage of t_relay(). I see from log that there are two branches involved: Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP> t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:xlite@192.168.0.164:7060;rinstance=cfca8523e3747cb0> t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) both INVITES reaches boudary SIP proxy. Because first contact was unreachable there was an tcp error: ser[3516]: ERROR: tcp_blocking_connect: poll error: flags 18 Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Nov 15 11:03:13 sen ser[3516]: ERROR: tcpconn_connect: tcp_blocking_connect failed Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_send: connect failed Nov 15 11:03:13 sen ser[3516]: msg_send: ERROR: tcp_send failed Nov 15 11:03:13 sen ser[3516]: ERROR: t_send_branch: sending request on branch 0 failed after which ser invokes sl_reply_error() which results in 477 next hop error reponse message sent to internal ser instance. The problem arises when boudary sip proxy processes second INVITE which correspond to second destination. when boundary SIP proxy receives it doesn't send it to user agent but instead reponses to internal ser with 500 I'm terribly sorry, server error occurred (1/SL) the only thing I see in the boundary SIP proxy log is: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Did I miss something here or this is an error? Waiting for Your feedback Best regards Tomasz

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[OpenSER-Users] OpenSER CVS
by Bogdan-Andrei Iancu 15 Nov '07

15 Nov '07

Hi, If there are good arguments against, Wednesday 14th, 12:00 AM GMT, the CVS repository from SF will be disabled. Regards, Bogdan

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