- sr-users - kamailio.org

[Users] to terminate a parallel forking
by Matteo Piazza 14 Dec '05

14 Dec '05

Hi, If I perform a parallel forking (in a failure route after that the first call downs in timeout) I see that if one of the recever decline the call the other destination continues to ring. Is there a command to terminate all parallel calls. Matteo

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[Serusers] Netcall - Job opportunity
by Edgar Barbosa 14 Dec '05

14 Dec '05

Netcall - Job opportunity Netcall (www.netcall.pt) is the leading VoIP service provider in Portugal. Netcall is also the new kid on the block, we are fighting the established old fashioned telco's. We are currently searching for an experienced systems engineer, highly specialized on VoIP services, to join our team in Porto, Portugal. We offer you a great career opportunity in a young and dynamic company, where your contribution would have a direct impact on the business definition and strategy. You will work as a member of the development team, designing and developing next generation VoIP services. Candidates should have: - Deep knowledge on networking and VoIP related protocols (SIP, IAX, ENUM, etc.) - Experience on SER, Asterisk and MySQL - Strong troubleshooting, debugging, problem solving, and analytical skills So if you want to get for a period experience abroad and also work in a challenging area where you can express yourself with your knowledge... please make the time to address us an email with CV... If you're interested and/or would like to get additional info, please feel free to contact us: emprego(a)netcall.pt

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[Serusers] PSTN guard
by Nikola Stojanoski 14 Dec '05

14 Dec '05

I'm trying to setup PSTN gateway for termination and i would like to use SER as a guard + to collect CDR's. but my consurn is the BYE messages i don't know how many failed BYE messages i'll get in realtime operation because i can't get correct cdr's without BYE. Is there a way to use mediaproxy rtp timeout so when i don't receive BYE i the call will be terminated let's say in 30 sec. What kind of message does mediaproxy send after that idle time CANCEL? And what is better way to do billing? Set cron to check balance on 5 min. or to use exec module and do mysql query after every BYE or CANCEL message? Using cron will make me lose maximum of 5 min. * number of PSTN lines and with query after every call i will lose only charge for active calls. Regards

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RE: [Users] Tricky Question
by Douglas Garstang 14 Dec '05

14 Dec '05

Samuel, I guess I can't call t_replicate more than once. Dammit! Doug. -----Original Message----- From: samuel [mailto:samu60@gmail.com] Sent: Wednesday, December 14, 2005 10:27 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Tricky Question Have you tried t_replicate instead of t_relay??? This t_replicate function should send the REGISTER to the other registrar and absorve the answers. However this t_replicate has (or had??) an inconvenient: it can be used with only one secondary host...so it's not possible to use it with several asterisks. Please, correct me if I am wrong. Samuel. 2005/12/14, Douglas Garstang <dgarstang(a)oneeighty.com>: > Here's a tricky question for the list. > > I have phones registering with OpenSER. After authenticating and saving the location, it forwards each registration request onto multiple Asterisk systems who send back Trying, OK, etc. OpenSER forwards these back to the phones eventhough they don't need to be. I want to stop them from getting back to the phones, because from the phone's perspective they've already registered, and now these extra Trying, Ok messages are coming back. Furtunately it doesn't seem to be causing any problems for the Polycom phones, but I'd still like ton get rid of them. > > How....? Maybe call setflag() when I get a registration, and then look for messages coming back with the flag? > > Doug > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Serusers] re:Changing Sip Port
by Juan 14 Dec '05

14 Dec '05

I still can't register with another port. This is what I am getting from ngrep. interface: \ (10.0.1.0/255.255.255.0) # # U 10.0.1.105:8080 -> 10.0.1.99:8080 REGISTER sip:10.0.1.99 SIP/2.0..Via: SIP/2.0/UDP 10.0.1.105:8080;rport;bran ch=z9hG4bKA2546AEF22FA4766A172D9533D938D55..From: Juan <sip:677403479814529 @10.0.1.99>;tag=2853267163..To: Juan <sip:677403479814529@10.0.1.99>..Conta ct: "Juan" <sip:677403479814529@10.0.1.105:8080>..Call-ID: 1BEBA6527D3A4797 93E960D259872D75@10.0.1.99..CSeq: 41059 REGISTER..Expires: 1800..Max-Forwar ds: 70..User-Agent: X-Lite release 1103m..Content-Length: 0.... # I 10.0.1.99 -> 10.0.1.105 3:3 ....E...}@.........i...c........REGISTER sip:10.0.1.99 SIP/2.0..Via: SIP/2. 0/UDP 10.0.1.105:8080;rport;branch=z9hG4bKA2546AEF22FA4766A172D9533D938D55. .From: Juan <sip:677403479814529@10.0.1.99>;tag=2853267163..To: Juan <sip:6 77403479814529(a)10.0.1.99 <mailto:77403479814529@10.0.1.99>..Contact> >..Contact: "Juan" <sip:677403479814529@10.0.1.105: 8080>..Call-ID: 1BEBA6527D3A479793E960D259872D75@10.0.1.99..CSeq: 41059 REG ISTER..Expires: 1800..Max-Forwards: 70..User-Agent: X-Lite release 1103m..C ontent-Length: 0.... # U 10.0.1.105:8080 -> 10.0.1.99:8080 .. # I 10.0.1.99 -> 10.0.1.105 3:3 ....E...}B.........i...c.......... I don't know why SER (10.0.1.99) is not sending a 100 trying but instead is sending like the echo of the REGISTER request to 10.0.1.105 3:3 ? Can some one enlight me? THANKS Juan

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[Serusers] Using SER together with Asterisk
by bharat.sarvan＠wipro.com 14 Dec '05

14 Dec '05

Hello all, I am trying to integrate SER and Asterisk. So could anybody please guide me with the changes and the configuration that I need to in either SER or Asterisk for achieving it. Like for suppose I want to achieve a call being placed from a User 1 registered to SER to User 2 who is registered to Asterisk.. So how do I move ahead to achieve the same. And are there any other new things that one can come up with using SER and Asterisk together which can't be achieved using SER or Asterisk alone? Kindly do let me know. Thanks and Regards, Bharat Sarvan The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[Users] Tricky Question
by Douglas Garstang 14 Dec '05

14 Dec '05

Here's a tricky question for the list. I have phones registering with OpenSER. After authenticating and saving the location, it forwards each registration request onto multiple Asterisk systems who send back Trying, OK, etc. OpenSER forwards these back to the phones eventhough they don't need to be. I want to stop them from getting back to the phones, because from the phone's perspective they've already registered, and now these extra Trying, Ok messages are coming back. Furtunately it doesn't seem to be causing any problems for the Polycom phones, but I'd still like ton get rid of them. How....? Maybe call setflag() when I get a registration, and then look for messages coming back with the flag? Doug

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RE: [Users] RE: failure_route.... I could cry...
by Douglas Garstang 14 Dec '05

14 Dec '05

Klaus, Adding a second nic can't hurt. Asterisk can bind to multiple interfaces in a system. We don't have to have the info in OpenSER. We're using OpenSER as a means by which we can 'copy' the registrations to all the Asterisk systems, as Asterisk has no way to do this. By nature of the fact that OpenSER is also doing the registration authentication, it's interfacing to MySQL and for now I'm saving the location info there as well. IMO most other people's setup's look a little under-engineered. Most people seem to be happy with having a hot or even cold spare. We're a telco (CLEC) so downtown is a deal breaker. Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:41 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] RE: failure_route.... I could cry... If you already have 2 asterisk boxes, I do not think that adding a second NIC in each asterisk will increase availability. Does Asterisk bind to both interfaces? Which interface will be used by asterisk for sending requests? Which src_IP will be used? What if this interface is broken? Why do you need the location info in Asterisk and openser? IMO your setup looks like a little bit over-engineered. regards klaus Douglas Garstang wrote: > Klaus, > > Yes. It's necessary to replicate 4 times. We're implementing a 'farm' of Asterisk hosts. Asterisk has a serious limitation where there's no way to share contact info for phones between them. OpenSER seemed like it may be able to address this issue, by accepting registrations from the phones and then 'fanning' them out to ALL the Asterisk boxes. In this way, each Asterisk box knows where every phone is. Up to this point I had just been calling forward() multiple times, once to each Asterisk system and it seemed to work well. With the addition of extra ethernet ports in each Asterisk box (for extra redundancy), it got more complicated as I didn't want to register twice to the same Asterisk system. While I don't think that would cause a problem, I was just trying to make it as bullet-proof as I could. > > Doug. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 9:06 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] RE: failure_route.... I could cry... > > > That's a difficult setup. I've never did this yet and do not know if it > is possible at all. you can use t_replicate but this works only with 1 > host. I think someone solved this using forward_tcp. > > is it really necessary to replicate 4 times? > > regards > klaus > > Douglas Garstang wrote: > >>Oh dear... it's failing because as far as t_relay() is concerned, the packet has been delivered, and the transaction is complete, right? If that's the case, how can I also send the REGISTRATION packet to another destination? Can I use forward() with failure_route? Arrgh!. >> >> >> >>>-----Original Message----- >>>From: Douglas Garstang >>>Sent: Tuesday, December 13, 2005 6:01 PM >>>To: 'users(a)openser.org' >>>Subject: failure_route.... I could cry... >>> >>>I am trying to have OpenSER forward registration requests to multiple Asterisk boxes with failover. Each Asterisk box has two ip addresses. My logic tries to relay to the first IP on the first Asterisk box (192.168.10.100). If that fails, it tries to relay to the second IP on the first Asterisk box (192.168.10.17). It then goes on and repeats the process with another Asterisk box who's IP's are 192.168.10.10.101 and 192.168.10.8. >>> >>>Actually the first IP address on each Asterisk system is fake (100 and 101)... that's how I am testing that it will fail to the second one. >>> >>>The openser.cfg below is causing this to be logged to messages: >>>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route10: Asterisk-1, NIC-1 >>>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: Asterisk-2, NIC-1 >>>Dec 13 17:50:08 bil-pdev-3 openser[22318]: ERROR: t_newtran: transaction already in process 0xb6126560 < --- Why? >>>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: t_relay returned error <--- Why? >>>Dec 13 17:50:09 bil-pdev-3 openser[22328]: failure21: Failed to register with Asterisk-2, NIC-1 >>>Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: Asterisk-2, NIC-2 >>>Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: End of routine >>>Dec 13 17:50:10 bil-pdev-3 openser[22322]: User local found >>>Dec 13 17:50:10 bil-pdev-3 openser[22320]: User local found >>> >>>Why am I getting the t_newtran and t_relay errors in there? What am I doing wrong? I'm just calling rewritehostport(next-ip), followed by append_branch() and t_relay. In fact something really weird happens. When I have the code there for route[20], failure_route[21] and route[22], it causes the code for the first three routing blocks, route[10], failure_route[11] and route[12] to act differently. It seems to make route20 fail to connect eventhough it comes AFTER it in the code. >>> >>>I'd really appreciate some help! Am I doing something wrong with rewritehostport and append_branch, because quite frankly the docs are terrible. >>> >>> >>> if ( method == REGISTER ) { >>> if ( !www_authorize("voip.com", "subscriber") ) { >>> www_challenge("voip.com", "0"); >>> }; >>> save("location"); >>> route(10); >>> route(20); >>> >>># >>># First Asterisk System. >>># >>> >>># Try first IP on this system (fake - will fail) >>>route[10] { >>> xlog ("L_INFO","route10: Asterisk-1, NIC-1"); >>> t_on_failure("11"); >>> rewritehostport("192.168.10.100:5060"); >>> append_branch(); >>> if ( !t_relay() ) { >>> xlog ("L_INFO","route10: t_relay returned error"); >>> } >>>} >>> >>>failure_route[11] { >>> xlog ("L_INFO","failure11: Failed to register with Asterisk-1, NIC-1"); >>> route(12); >>>} >>> >>># Try second IP on this system (real - should work) >>>route[12] { >>> xlog ("L_INFO","route12: Asterisk-1, NIC-2"); >>> t_on_failure("13"); >>> rewritehostport("192.168.10.17:5060"); >>> append_branch(); >>> if ( !t_relay() ) { >>> xlog ("L_INFO","route12: t_relay returned error"); >>> } >>> xlog ("L_INFO","route12: End of routine"); >>> return; >>>} >>> >>># >>># Second Asterisk System >>># >>> >>># Try first IP on this system (fake - will fail) >>>route[20] { >>> xlog ("L_INFO","route20: Asterisk-2, NIC-1"); >>> t_on_failure("21"); >>> rewritehostport("192.168.10.101:5060"); >>> append_branch(); >>> if ( !t_relay() ) { >>> xlog ("L_INFO","route20: t_relay returned error");> >>> } >>>} >>> >>>failure_route[21] { >>> xlog ("L_INFO","failure21: Failed to register with Asterisk-2, NIC-1"); >>> route(22); >>>} >>> >>># Try second IP on this system (real - should work) >>>route[22] { >>> xlog ("L_INFO","route22: Asterisk-2, NIC-2"); >>> t_on_failure("23"); >>> rewritehostport("192.168.10.8:5060"); >>> append_branch(); >>> if ( !t_relay() ) { >>> xlog ("L_INFO","route22: t_relay returned error"); >>> } >>> xlog ("L_INFO","route22: End of routine"); >>> return; >>>} >>> >>>Doug >>> >>> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

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RE: [Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Let me ask this question... Does the RURI get consumed when a message is sucessfully delivered, or when a message is attempted to be delivered, but fails? -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:48 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] More Routing.... I do not know if it also works this way. I always used the version I sent in the previous email and it always worked. Thus I never had the need to change it. klaus Douglas Garstang wrote: > Ok...so... why not just call append_branch("192.168.10.8:5060") without first calling rewritehostport("192.168.10.8")? It seems like redundant information. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 9:01 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] More Routing.... > > > Where's the problem? in route[1] or in the failure route? > > You need append branch in failure route > > klaus > > Douglas Garstang wrote: > >>Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! >> >>route(1); >> >>route[1] { >> rewritehostport("192.168.10.7:5060"); >> t_on_failure("2"); >> t_relay(); >>} >> >>failure_route[2] { >> rewritehostport('192.168.10.8:5060"); >> t_relay(); >>} >> >>Doug. >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

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RE: [Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Ok...so... why not just call append_branch("192.168.10.8:5060") without first calling rewritehostport("192.168.10.8")? It seems like redundant information. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:01 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] More Routing.... Where's the problem? in route[1] or in the failure route? You need append branch in failure route klaus Douglas Garstang wrote: > Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! > > route(1); > > route[1] { > rewritehostport("192.168.10.7:5060"); > t_on_failure("2"); > t_relay(); > } > > failure_route[2] { > rewritehostport('192.168.10.8:5060"); > t_relay(); > } > > Doug. > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

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RE: [Users] RE: failure_route.... I could cry...
by Douglas Garstang 14 Dec '05

14 Dec '05

Klaus, Yes. It's necessary to replicate 4 times. We're implementing a 'farm' of Asterisk hosts. Asterisk has a serious limitation where there's no way to share contact info for phones between them. OpenSER seemed like it may be able to address this issue, by accepting registrations from the phones and then 'fanning' them out to ALL the Asterisk boxes. In this way, each Asterisk box knows where every phone is. Up to this point I had just been calling forward() multiple times, once to each Asterisk system and it seemed to work well. With the addition of extra ethernet ports in each Asterisk box (for extra redundancy), it got more complicated as I didn't want to register twice to the same Asterisk system. While I don't think that would cause a problem, I was just trying to make it as bullet-proof as I could. Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:06 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] RE: failure_route.... I could cry... That's a difficult setup. I've never did this yet and do not know if it is possible at all. you can use t_replicate but this works only with 1 host. I think someone solved this using forward_tcp. is it really necessary to replicate 4 times? regards klaus Douglas Garstang wrote: > Oh dear... it's failing because as far as t_relay() is concerned, the packet has been delivered, and the transaction is complete, right? If that's the case, how can I also send the REGISTRATION packet to another destination? Can I use forward() with failure_route? Arrgh!. > > >> -----Original Message----- >>From: Douglas Garstang >>Sent: Tuesday, December 13, 2005 6:01 PM >>To: 'users(a)openser.org' >>Subject: failure_route.... I could cry... >> >>I am trying to have OpenSER forward registration requests to multiple Asterisk boxes with failover. Each Asterisk box has two ip addresses. My logic tries to relay to the first IP on the first Asterisk box (192.168.10.100). If that fails, it tries to relay to the second IP on the first Asterisk box (192.168.10.17). It then goes on and repeats the process with another Asterisk box who's IP's are 192.168.10.10.101 and 192.168.10.8. >> >>Actually the first IP address on each Asterisk system is fake (100 and 101)... that's how I am testing that it will fail to the second one. >> >>The openser.cfg below is causing this to be logged to messages: >>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route10: Asterisk-1, NIC-1 >>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: Asterisk-2, NIC-1 >>Dec 13 17:50:08 bil-pdev-3 openser[22318]: ERROR: t_newtran: transaction already in process 0xb6126560 < --- Why? >>Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: t_relay returned error <--- Why? >>Dec 13 17:50:09 bil-pdev-3 openser[22328]: failure21: Failed to register with Asterisk-2, NIC-1 >>Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: Asterisk-2, NIC-2 >>Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: End of routine >>Dec 13 17:50:10 bil-pdev-3 openser[22322]: User local found >>Dec 13 17:50:10 bil-pdev-3 openser[22320]: User local found >> >>Why am I getting the t_newtran and t_relay errors in there? What am I doing wrong? I'm just calling rewritehostport(next-ip), followed by append_branch() and t_relay. In fact something really weird happens. When I have the code there for route[20], failure_route[21] and route[22], it causes the code for the first three routing blocks, route[10], failure_route[11] and route[12] to act differently. It seems to make route20 fail to connect eventhough it comes AFTER it in the code. >> >>I'd really appreciate some help! Am I doing something wrong with rewritehostport and append_branch, because quite frankly the docs are terrible. >> >> >> if ( method == REGISTER ) { >> if ( !www_authorize("voip.com", "subscriber") ) { >> www_challenge("voip.com", "0"); >> }; >> save("location"); >> route(10); >> route(20); >> >># >># First Asterisk System. >># >> >># Try first IP on this system (fake - will fail) >>route[10] { >> xlog ("L_INFO","route10: Asterisk-1, NIC-1"); >> t_on_failure("11"); >> rewritehostport("192.168.10.100:5060"); >> append_branch(); >> if ( !t_relay() ) { >> xlog ("L_INFO","route10: t_relay returned error"); >> } >>} >> >>failure_route[11] { >> xlog ("L_INFO","failure11: Failed to register with Asterisk-1, NIC-1"); >> route(12); >>} >> >># Try second IP on this system (real - should work) >>route[12] { >> xlog ("L_INFO","route12: Asterisk-1, NIC-2"); >> t_on_failure("13"); >> rewritehostport("192.168.10.17:5060"); >> append_branch(); >> if ( !t_relay() ) { >> xlog ("L_INFO","route12: t_relay returned error"); >> } >> xlog ("L_INFO","route12: End of routine"); >> return; >>} >> >># >># Second Asterisk System >># >> >># Try first IP on this system (fake - will fail) >>route[20] { >> xlog ("L_INFO","route20: Asterisk-2, NIC-1"); >> t_on_failure("21"); >> rewritehostport("192.168.10.101:5060"); >> append_branch(); >> if ( !t_relay() ) { >> xlog ("L_INFO","route20: t_relay returned error");> >> } >>} >> >>failure_route[21] { >> xlog ("L_INFO","failure21: Failed to register with Asterisk-2, NIC-1"); >> route(22); >>} >> >># Try second IP on this system (real - should work) >>route[22] { >> xlog ("L_INFO","route22: Asterisk-2, NIC-2"); >> t_on_failure("23"); >> rewritehostport("192.168.10.8:5060"); >> append_branch(); >> if ( !t_relay() ) { >> xlog ("L_INFO","route22: t_relay returned error"); >> } >> xlog ("L_INFO","route22: End of routine"); >> return; >>} >> >>Doug >> >> > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

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RE: [Users] Routing Examples
by Douglas Garstang 14 Dec '05

14 Dec '05

And you need to call append_branch().... why? The RURI is consumed? Correct? Not sure I understand when it gets consumed. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:32 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Routing Examples Then try the same without AVPs - no difference in the basic prinicples: failure_route[1] { sethostport("1.2.3.4:5061"); append_branch(); t_relay(); } Douglas Garstang wrote: > Ugh... avps....I absolutely can't follow the docs on avp's. I have -no idea- what that's all about. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 8:58 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] Routing Examples > > > Douglas Garstang wrote: > >>I'm having a terrible time trying to get failure routes to work. Can someone point me to some USEFUL examples please? The examples that come with OpenSER are trivial. They all use append_branch("sip:user@domain") which in the real world is barely useful. I need to try sending messages to a specified host with the current user's URI. >> >>For example, is this the correct usage for trying to connect in sequence to multple destinations? >> >>rewritehostport("192.168.10.7:5060"); >>append_branch(); >>t_on_failure("2"); >>t_relay(); >> >>failure_route[2] { >> rewritehostport("192.168.10.7:5060"); >> append_branch(); >> t_relay(); >>} >> >>The second route is never tried. In general, what should I be doing here? >>Which is correct? > > > > I use: > > failure_route[1] { > > # send to voicemail in format sip:<extension>@1.2.3.4:5061 > avp_pushto("$ruri/username","$extension"); > sethostport("1.2.3.4:5061"); > append_branch(); > t_relay(); > } > > klaus > > >>rewritehostport(ip-addr) >>append_branch() >>t_relay() >> >>or maybe... >>append_branch(ip-addr) >>t_relay() >> >>or maybe... >>append_branch(ip-addr1) >>append_branch(ip-addr2) >>t_relay() >> >>Do you get my point? The docs are really bad and don't cover exactly how this stuff is supposed to be implemented! if I do a google search on this stuff, I get almost no matches. There's no books either. I'm out of ideas. >> >>I'm just trying to connect to multiple destinations in sequence.... >> >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

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RE: [Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Ok, I'm very confused about this. Bogdan mentioned that the RURI is consumed? When is it consumed? -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:31 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] More Routing.... append_branch will add a new branch. But the t_relay in failure route only uses the new branch, as the previous branch is alreaddy terminated. klaus Douglas Garstang wrote: > Klaus, > > But, doesn't append_branch cause parallel forking? I don't want parallel forking. If I do parallel forking, the same registration may get delivered to the same Asterisk box twice, and that's not what I want. > > Doug. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 9:01 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] More Routing.... > > > Where's the problem? in route[1] or in the failure route? > > You need append branch in failure route > > klaus > > Douglas Garstang wrote: > >>Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! >> >>route(1); >> >>route[1] { >> rewritehostport("192.168.10.7:5060"); >> t_on_failure("2"); >> t_relay(); >>} >> >>failure_route[2] { >> rewritehostport('192.168.10.8:5060"); >> t_relay(); >>} >> >>Doug. >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

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RE: [Users] Routing Examples
by Douglas Garstang 14 Dec '05

14 Dec '05

Ugh... avps....I absolutely can't follow the docs on avp's. I have -no idea- what that's all about. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 8:58 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Routing Examples Douglas Garstang wrote: > I'm having a terrible time trying to get failure routes to work. Can someone point me to some USEFUL examples please? The examples that come with OpenSER are trivial. They all use append_branch("sip:user@domain") which in the real world is barely useful. I need to try sending messages to a specified host with the current user's URI. > > For example, is this the correct usage for trying to connect in sequence to multple destinations? > > rewritehostport("192.168.10.7:5060"); > append_branch(); > t_on_failure("2"); > t_relay(); > > failure_route[2] { > rewritehostport("192.168.10.7:5060"); > append_branch(); > t_relay(); > } > > The second route is never tried. In general, what should I be doing here? > Which is correct? I use: failure_route[1] { # send to voicemail in format sip:<extension>@1.2.3.4:5061 avp_pushto("$ruri/username","$extension"); sethostport("1.2.3.4:5061"); append_branch(); t_relay(); } klaus > > rewritehostport(ip-addr) > append_branch() > t_relay() > > or maybe... > append_branch(ip-addr) > t_relay() > > or maybe... > append_branch(ip-addr1) > append_branch(ip-addr2) > t_relay() > > Do you get my point? The docs are really bad and don't cover exactly how this stuff is supposed to be implemented! if I do a google search on this stuff, I get almost no matches. There's no books either. I'm out of ideas. > > I'm just trying to connect to multiple destinations in sequence.... > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

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RE: [Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Klaus, But, doesn't append_branch cause parallel forking? I don't want parallel forking. If I do parallel forking, the same registration may get delivered to the same Asterisk box twice, and that's not what I want. Doug. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 14, 2005 9:01 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] More Routing.... Where's the problem? in route[1] or in the failure route? You need append branch in failure route klaus Douglas Garstang wrote: > Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! > > route(1); > > route[1] { > rewritehostport("192.168.10.7:5060"); > t_on_failure("2"); > t_relay(); > } > > failure_route[2] { > rewritehostport('192.168.10.8:5060"); > t_relay(); > } > > Doug. > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > >

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Re: [Users] Debug Destionation Set
by Klaus Darilion 14 Dec '05

14 Dec '05

have you tried http://openser.org/docs/pseudo-variables-1.0.x.html#pv_ds klaus Douglas Garstang wrote: > Is there a pseudo variable that will print all the members of the current destination set? A keyword search for 'destination' in the pseudo variables doc didn't find anything, so I'm guessing not. > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Wednesday, December 14, 2005 8:54 AM > To: Douglas Garstang > Cc: users(a)openser.org > Subject: Re: [Users] Debug Destionation Set > > > Douglas Garstang wrote: > >>Is there a way to see what's in the current destionation set? Either by a command or debug output from OpenSER. It doesn't seem to display anything in debug about the destination set at all. > > > use xlog and pseudo variables > > klaus > > >>Doug. >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

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[Users] mysqlso error
by maillisting 14 Dec '05

14 Dec '05

I compile mysql.so with the following command. make modules=modules/mysql modules I have followed the document to configure openuser to use SQL. When I start openser, the error message below is shown. What does it mean the module compile flags? Any clue? openser: ERROR: module compile flags mismatch for /usr/local/lib/openser/modules/mysql.so core: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT <mailto:users@openser.org>

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RE: [Users] Routing Examples
by Douglas Garstang 14 Dec '05

14 Dec '05

Bogdan... Thanks for the info. Ok... well I think I need to do serial forking. I don't want to have a situation where the registration gets delivered to two IP's on the same host because both happen to be available. I don't think Asterisk will have a problem with it, but to be safe I'd like to avoid that. So, back to my original logic of trying first ip on first Asterisk box, if that fails try second ip on first Asterisk box. Then, move onto the next and repeat the process. With pareallel forking, what happens with failures? Does a failure of any single host in destination set cause an error that can be handled with failure_route? Or, do all hosts have to fail before you can detect an error? How can you determine which one failed? I don't think parallel forking is what I need, but thanks for clearing that up for me. Doug. -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wednesday, December 14, 2005 8:55 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] Routing Examples Hi Doug, if you have: rewritehostport("192.168.10.7:5060"); t_on_failure("2"); t_relay(); failure_route[2] { rewritehostport("192.168.10.7:5060"); append_branch(); t_relay(); } you do serial forking - try first destination and if fails, o for the second one. for: rewritehostport("192.168.10.7:5060"); append_branch("192.168.10.7:5060"); t_relay() you do parallel forking - the request will be sent in the same time to all destinations. regards, bogdan Douglas Garstang wrote: >I'm having a terrible time trying to get failure routes to work. Can someone point me to some USEFUL examples please? The examples that come with OpenSER are trivial. They all use append_branch("sip:user@domain") which in the real world is barely useful. I need to try sending messages to a specified host with the current user's URI. > >For example, is this the correct usage for trying to connect in sequence to multple destinations? > >rewritehostport("192.168.10.7:5060"); >append_branch(); >t_on_failure("2"); >t_relay(); > >failure_route[2] { > rewritehostport("192.168.10.7:5060"); > append_branch(); > t_relay(); >} > >The second route is never tried. In general, what should I be doing here? >Which is correct? > >rewritehostport(ip-addr) >append_branch() >t_relay() > >or maybe... >append_branch(ip-addr) >t_relay() > >or maybe... >append_branch(ip-addr1) >append_branch(ip-addr2) >t_relay() > >Do you get my point? The docs are really bad and don't cover exactly how this stuff is supposed to be implemented! if I do a google search on this stuff, I get almost no matches. There's no books either. I'm out of ideas. > >I'm just trying to connect to multiple destinations in sequence.... > > > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

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Re: [Users] More Routing....
by Bogdan-Andrei Iancu 14 Dec '05

14 Dec '05

Doug, in failure route, the RURI was already consumed (you already got reply for it) - so, if you want to relay again, you need to append a new branch. regards, bogdan Douglas Garstang wrote: >Bogdan, > >I don't understand why I need to do this. Doesn't rewritehostport rewrite the current URI so that t_relay can statefully deliver? If so, what's the point of append_branch? It seems redundant, which is why I don't understand the need for it. > >Doug. > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Wednesday, December 14, 2005 8:47 AM >To: Douglas Garstang >Cc: users(a)openser.org >Subject: Re: [Users] More Routing.... > > >Hi Doug, > >in failure route, in order to fork a new serial branch, you need to call >append_branch() before t_relay(). > >regards, >bogdan > >Douglas Garstang wrote: > > > >>Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! >> >>route(1); >> >>route[1] { >> rewritehostport("192.168.10.7:5060"); >> t_on_failure("2"); >> t_relay(); >>} >> >>failure_route[2] { >> rewritehostport('192.168.10.8:5060"); >> t_relay(); >>} >> >>Doug. >> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> > > >

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[Serusers] problems with RTP proxy
by Anthony Chapellier 14 Dec '05

14 Dec '05

Hi, I've got troubles with RTP proxy. When I initiate a call between two UAs, I see RTP packets arriving from both clients to RTP proxy but they don't leave the proxy and of course clients don't receive them... does someone know what the problem could be ? Thanks,

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[Users] RE: failure_route.... I could cry...
by Douglas Garstang 14 Dec '05

14 Dec '05

Oh dear... it's failing because as far as t_relay() is concerned, the packet has been delivered, and the transaction is complete, right? If that's the case, how can I also send the REGISTRATION packet to another destination? Can I use forward() with failure_route? Arrgh!. > -----Original Message----- > From: Douglas Garstang > Sent: Tuesday, December 13, 2005 6:01 PM > To: 'users(a)openser.org' > Subject: failure_route.... I could cry... > > I am trying to have OpenSER forward registration requests to multiple Asterisk boxes with failover. Each Asterisk box has two ip addresses. My logic tries to relay to the first IP on the first Asterisk box (192.168.10.100). If that fails, it tries to relay to the second IP on the first Asterisk box (192.168.10.17). It then goes on and repeats the process with another Asterisk box who's IP's are 192.168.10.10.101 and 192.168.10.8. > > Actually the first IP address on each Asterisk system is fake (100 and 101)... that's how I am testing that it will fail to the second one. > > The openser.cfg below is causing this to be logged to messages: > Dec 13 17:50:08 bil-pdev-3 openser[22318]: route10: Asterisk-1, NIC-1 > Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: Asterisk-2, NIC-1 > Dec 13 17:50:08 bil-pdev-3 openser[22318]: ERROR: t_newtran: transaction already in process 0xb6126560 < --- Why? > Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: t_relay returned error <--- Why? > Dec 13 17:50:09 bil-pdev-3 openser[22328]: failure21: Failed to register with Asterisk-2, NIC-1 > Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: Asterisk-2, NIC-2 > Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: End of routine > Dec 13 17:50:10 bil-pdev-3 openser[22322]: User local found > Dec 13 17:50:10 bil-pdev-3 openser[22320]: User local found > > Why am I getting the t_newtran and t_relay errors in there? What am I doing wrong? I'm just calling rewritehostport(next-ip), followed by append_branch() and t_relay. In fact something really weird happens. When I have the code there for route[20], failure_route[21] and route[22], it causes the code for the first three routing blocks, route[10], failure_route[11] and route[12] to act differently. It seems to make route20 fail to connect eventhough it comes AFTER it in the code. > > I'd really appreciate some help! Am I doing something wrong with rewritehostport and append_branch, because quite frankly the docs are terrible. > > > if ( method == REGISTER ) { > if ( !www_authorize("voip.com", "subscriber") ) { > www_challenge("voip.com", "0"); > }; > save("location"); > route(10); > route(20); > > # > # First Asterisk System. > # > > # Try first IP on this system (fake - will fail) > route[10] { > xlog ("L_INFO","route10: Asterisk-1, NIC-1"); > t_on_failure("11"); > rewritehostport("192.168.10.100:5060"); > append_branch(); > if ( !t_relay() ) { > xlog ("L_INFO","route10: t_relay returned error"); > } > } > > failure_route[11] { > xlog ("L_INFO","failure11: Failed to register with Asterisk-1, NIC-1"); > route(12); > } > > # Try second IP on this system (real - should work) > route[12] { > xlog ("L_INFO","route12: Asterisk-1, NIC-2"); > t_on_failure("13"); > rewritehostport("192.168.10.17:5060"); > append_branch(); > if ( !t_relay() ) { > xlog ("L_INFO","route12: t_relay returned error"); > } > xlog ("L_INFO","route12: End of routine"); > return; > } > > # > # Second Asterisk System > # > > # Try first IP on this system (fake - will fail) > route[20] { > xlog ("L_INFO","route20: Asterisk-2, NIC-1"); > t_on_failure("21"); > rewritehostport("192.168.10.101:5060"); > append_branch(); > if ( !t_relay() ) { > xlog ("L_INFO","route20: t_relay returned error");> > } > } > > failure_route[21] { > xlog ("L_INFO","failure21: Failed to register with Asterisk-2, NIC-1"); > route(22); > } > > # Try second IP on this system (real - should work) > route[22] { > xlog ("L_INFO","route22: Asterisk-2, NIC-2"); > t_on_failure("23"); > rewritehostport("192.168.10.8:5060"); > append_branch(); > if ( !t_relay() ) { > xlog ("L_INFO","route22: t_relay returned error"); > } > xlog ("L_INFO","route22: End of routine"); > return; > } > > Doug > >

3 2

[Users] More Routing....
by Douglas Garstang 14 Dec '05

14 Dec '05

Can someone please tell me why the following extremely simple example doesn't first attempt to relay to 192.168.10.7, and then if that fails, try 192.168.10.8? What am I missing here? The documentation says that t_relay() simple sends statefully to the current URI.... seems to be what I am doing. What am I missing? Please help! route(1); route[1] { rewritehostport("192.168.10.7:5060"); t_on_failure("2"); t_relay(); } failure_route[2] { rewritehostport('192.168.10.8:5060"); t_relay(); } Doug.

3 2

[Users] Routing Examples
by Douglas Garstang 14 Dec '05

14 Dec '05

I'm having a terrible time trying to get failure routes to work. Can someone point me to some USEFUL examples please? The examples that come with OpenSER are trivial. They all use append_branch("sip:user@domain") which in the real world is barely useful. I need to try sending messages to a specified host with the current user's URI. For example, is this the correct usage for trying to connect in sequence to multple destinations? rewritehostport("192.168.10.7:5060"); append_branch(); t_on_failure("2"); t_relay(); failure_route[2] { rewritehostport("192.168.10.7:5060"); append_branch(); t_relay(); } The second route is never tried. In general, what should I be doing here? Which is correct? rewritehostport(ip-addr) append_branch() t_relay() or maybe... append_branch(ip-addr) t_relay() or maybe... append_branch(ip-addr1) append_branch(ip-addr2) t_relay() Do you get my point? The docs are really bad and don't cover exactly how this stuff is supposed to be implemented! if I do a google search on this stuff, I get almost no matches. There's no books either. I'm out of ideas. I'm just trying to connect to multiple destinations in sequence....

3 2

[Users] Debug Destionation Set
by Douglas Garstang 14 Dec '05

14 Dec '05

Is there a way to see what's in the current destionation set? Either by a command or debug output from OpenSER. It doesn't seem to display anything in debug about the destination set at all. Doug.

3 2

[Users] connection failed
by maillisting 14 Dec '05

14 Dec '05

Hi all, Anyone knows the meaning of the following errors? Anyway to solve it? Dec 9 17:16:00 ow01 ./openser[19379]: ERROR: tcp_blocking_connect: timeout (10) Dec 9 17:16:00 ow01 ./openser[19379]: ERROR: tcpconn_connect: tcp_blocking_connect failed Dec 9 17:16:00 ow01 ./openser[19379]: ERROR: tcp_send: connect failed Dec 9 17:16:00 ow01 ./openser[19379]: msg_send: ERROR: tcp_send failed Dec 9 17:16:00 ow01 ./openser[19379]: ERROR:tm:t_forward_nonack: sending request failed Dec 9 17:16:00 ow01 ./openser[19379]: ERROR:tm: _reply_light: can't generate 477 reply when a final 487 was sent out Dec 9 17:16:00 ow01 ./openser[19379]: ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL)

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[Users] Use of append_branch
by Douglas Garstang 14 Dec '05

14 Dec '05

How can I call append branch and use a variable name in the user port (ie the user for which this message is addressed) of the URI? _ALL_ the examples have something like: append_branch("sip:nonsense@192.168.2.16:5064"); which is completely useless. I need to call append_branch and have the user portion of the uri inserted. Something like: append_branch("sip:$to_user@192.168.2.16:5064"); How can I do this? Doug.

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Re: [Devel] [Users] nathelper/rtpproxy when both SIP UA are behind same NAT
by Norman Brandinger 14 Dec '05

14 Dec '05

Hi Rafael, You would use the "example code snippet" when you want to decide whether or not to use rtpproxy or mediaproxy based on the fact that both the caller and the callee are behind the same NAT device. Please review the mailing lists as this topic has been discussed many times. I added some additional comments in regard to the setting of flags 2 and 3 in the posted example at: http://openser.org/dokuwiki/doku.php?id=avp_examples Generally, I have found that if OpenSER fails to start with an error indicating that the function can't be found, it's time to find the correct module to add. In response to your last question "etc." I suggest that you review the three web sites: openser.org, iptel.org and onsip.org. The last one has some really great material that explains alot. Regards, Norm Rafael R. GV wrote: > Hi > I am using nathelper/rtpproxy in ser , please see my ser.cfg attached > and tell me where do I have to use this code snippet? where did you > set flags 2 and 3?, what other modules I need?, etc. > > thank you > Rafael > Lima-Peru > > On 12/7/05, *Norman Brandinger* <norm(a)goes.com <mailto:norm@goes.com>> > wrote: > > Thanks go out to Klaus and Tavis. > > I took the results of this thread and placed it in the docuwiki > for the > rest of the user community to use (at least, the rest of the user > community that reads the docuwiki :) > > http://openser.org/dokuwiki/doku.php?id=avp_examples > > If I made any typos, please feel free to correct them. > > Regards, > Norm > > _______________________________________________ > Users mailing list > Users(a)openser.org <mailto:Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users > <http://openser.org/cgi-bin/mailman/listinfo/users> > > > ------------------------------------------------------------------------ > > # > # If you don't want to enforce RTP proxy for some destinations > # then simply use t_relay() instead of route(1) > # > # Modificado para MYSQL_ACC support > # Handling of Unavailable user and Voicemail redirection (404|408|486) > # > # Log Missed Calls (403|404|408|415|484|486|487). > # > # Group checking and PSTN credentials. > # > # 18/07/2005 Prepaid & B2bua support added > # 11/11/2005 LDI a 192.168.2.132 with a2billing > # > # Pendientes !! - Administrar Multi-Dominio para otros Códigos de Area. > # - Completar Logica para tratamiento de llamadas entre > # un mismo NAT usando avpops. > # > # ----------- global configuration parameters ------------------------ > > #/* Uncomment these lines to enter debugging mode > debug=2 > fork=yes > log_stderror=yes > #*/ > > listen=192.168.2.130 > listen=127.0.0.1 > port=5060 > > # hostname matching an alias will satisfy the condition uri==myself". > alias=mydomain.com.pe > alias=127.0.0.1 > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > children=4 > fifo="/tmp/ser_fifo" > fifo_mode=0666 # Fifo permissions can be changes from here. > > # sip_warning - Should replies include extensive warnings? > # By default yes, it is good for trouble-shooting. > sip_warning=yes > > # ------------------ module loading ---------------------------------- > loadmodule "/usr/local/lib/ser/modules/domain.so" > loadmodule "/usr/local/lib/ser/modules/avpops.so" > 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/group.so" > loadmodule "/usr/local/lib/ser/modules/uri.so" > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > loadmodule "/usr/local/lib/ser/modules/acc.so" > loadmodule "/usr/local/lib/ser/modules/textops.so" > > # digest authentication > loadmodule "/usr/local/lib/ser/modules/auth.so" > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > # !! Nathelper > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > # ----------------- setting module-specific parameters --------------- > > modparam("usrloc", "db_mode", 2) > > # minimize write back window - default is 60 seconds > modparam("usrloc", "timer_interval", 10) > > # database location > modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") > > modparam("usrloc", "use_domain", 1) > modparam("auth_db", "use_domain", 1) > > > modparam("domain", "db_mode", 1) > modparam("domain", "domain_table", "domain") > modparam("domain", "domain_col", "domain") > > > # ------------- Mysql Accounting parameters > modparam("acc", "log_flag", 1) > modparam("acc", "log_level", 2) > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 3) > modparam("acc", "log_missed_flag", 3) > modparam("acc", "db_url", "mysql://seradmin:heslo@localhost/ser") > modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc (para INVITE y ACK) > modparam("acc", "failed_transactions", 1) # *all* non-200 transactions marked for acc will be logged too. > > modparam("acc", "log_fmt", "miocfsputdr") > > modparam("tm", "fr_timer", 20 ) > modparam("tm", "fr_inv_timer", 40 ) # Timer which hits if no final reply for an INVITE > modparam("tm", "wt_timer", 20 ) > > # add value to ;lr param to make some broken UAs happy > modparam("rr", "enable_full_lr", 1) > > modparam("group", "db_url", "mysql://seradmin:heslo@localhost/ser") > modparam("uri_db", "db_url", "mysql://seradmin:heslo@localhost/ser") > > # ------------- registration parameters > modparam("registrar", "nat_flag", 6) > modparam("registrar", "min_expires", 60) > modparam("registrar", "max_expires", 86400) > modparam("registrar", "default_expires", 3600) > modparam("registrar", "desc_time_order", 1) > modparam("registrar", "append_branches", 1) > > modparam("registrar", "use_domain", 1) > > # !! Nathelper > # modparam("registrar", "nat_flag", 6) > modparam("nathelper", "natping_interval", 30) # Ping interval 30 s > modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT > > # -------------------------- request routing logic -------------------------- > > route { > > log(1, "-------------------------------------------\n"); > log(1, "entering main loop\n"); > > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > break; > }; > if ( msg:len >= max_len ) { > sl_send_reply("513", "Message too big"); > break; > }; > > # !! Nathelper > # Special handling for NATed clients; first, NAT test is > # executed: it looks for via!=received and RFC1918 addresses > # in Contact (may fail if line-folding is used); also, > # the received test should, if completed, should check all > # vias for rpesence of received > if (nat_uac_test("19")) { > # Allow RR-ed requests, as these may indicate that > # a NAT-enabled proxy takes care of it; unless it is > # a REGISTER > > if (method == "REGISTER" || ! search("^Record-Route:")) { > log("LOG: Someone trying to register from private IP, rewriting\n"); > > # This will work only for user agents that support symmetric > # communication. We tested quite many of them and majority is > # smart enough to be symmetric. In some phones it takes a configuration > # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is > # called "symmetric media" and "symmetric signalling". > > fix_nated_contact(); # Rewrite contact with source IP of signalling > if (method == "INVITE") { > fix_nated_sdp("1"); # Add direction=active to SDP > }; > force_rport(); # Add rport parameter to topmost Via > setflag(6); # Mark as NATed > }; > }; > > > # set flag for Radius Accounting: > > > if (!method=="OPTIONS") setflag(3); # SET MISSED_CALLS FLAG FOR ACC > > if (method=="INVITE") { > log(1, "INVITE MESSAGE RECEIVED - START ACC\n"); > setflag(1); /* set for accounting (the same value as in log_flag!) */ > }; > > if (method=="BYE") { > log (1, "BYE - STOP ACCOUNTING\n"); > setflag(1); > }; > > if (method=="CANCEL") { > log (1, "CANCEL - STOP ACCOUNTING\n"); > setflag(1); > }; > > > # record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > > if (!method=="REGISTER") record_route(); > > # subsequent messages withing a dialog should take the > # path determined by record-routing > # Excluding packets from b2bua (port 5070) > if (loose_route() ) { > log(1," Mark routing logic in request --> rr-enforced \n"); > append_hf("P-hint: rr-enforced\r\n"); > # t_relay(); > route(1); # Nathelper!! > break; > }; > > if (!uri==myself ) { > log(1," Mark routing logic in request --> outbound\r \n"); > append_hf("P-hint: outbound\r\n"); > # t_relay(); > route(1); # Nathelper!! > break; > }; > > if (uri==myself) { > > if (method == "REGISTER") { > > log(1, "ANALYZING REGISTER REQUEST\n"); > # to use digest authentication > if (is_user_in("Request-URI", "desactivado")) { > sl_send_reply("402", "Su cuenta fue desactivada"); > break; > }; > > if (!www_authorize("mydomain.com.pe", "subscriber")) { > log(1," ----- Fails to Register \n"); > www_challenge("mydomain.com.pe", "0"); > break; > }; > > # only signed users are allowed > if (!check_to()) { > log(1, "LOG: Hijack!!!--> unsigned user registration attempt\n"); > sl_send_reply("403", "hijack attempt!!!! Only signed users are allowed"); > break; > }; > log(1," Registered!!! \n"); > if (!save("location")) { > sl_reply_error(); > }; > break; > }; > > > #For *B2bua: First check the source of the call > #********************************************** > # If the call comes from the gateways, no authentication is required. > if (src_ip==192.168.2.145 || src_ip==192.168.2.132 || src_ip==192.168.2.131) { > log(1,"Call from pstn|*, no authentication is required. \n"); > > # If the call comes from B2BUA, no authentication is required, > # The first leg of the call has already been authenticated. > } else if (src_ip==192.168.2.130 && src_port==5070) { > log(1,"Call from B2BUA, no authentication is required. \n"); > } else { > # We check user credentials > if ((method == "INVITE" || method== "CANCEL" || method== "BYE" || method== "ACK") && (!src_ip==192.168.2.130 && !src_port==5070)){ > log(1, "ANALYZING INVITE||CANCEL REQUESTs\n"); > if (!proxy_authorize("mydomain.com.pe", "subscriber")) { > # log(1," ----- Fails to ...proxy_authorize \n"); > proxy_challenge("mydomain.com.pe", "0"); > break; > } else { > if (method == "INVITE" && !check_from()) { > sl_send_reply("403", "Only registered users are allowed"); > log(1," ----> Only registered users are allowed \n"); > break; > }; > }; > > # Not all the users are PREPAID, so we check the database > # to see if the call will be routed through B2BUA. > # If every call is to be routed through B2BUA the "is_user_in" > # conditional is not required. > # Do not use b2bua for local calls > > if (is_user_in("From", "prepaidb") && uri=~"^sip:00") { > log(1," ----> Usuario PREPAGO!!! enviando a b2bua... \n"); > rewritehostport("192.168.2.130:5070"); > t_relay_to_udp("192.168.2.130", "5070"); > break; > }; > > };# End of if (method == "INVITE" |... > }; > > > /* *********** Dial out to Local and PSTN logic ********* */ > > > # Forward +9n digit requests to gateway Cisco-AS5350 (Celulares): > > if(uri=~"^sip:9" ){ > log(1," digit expression match - Celulares \n"); > if (!is_user_in("from", "movlim")) { > sl_send_reply("403", "No permission for mobile calls"); > acc_db_request("403 Forbidden", "missed_calls"); > break; > }; > rewritehostport("192.168.2.145:5060"); > route(1); > break; > }; > > > /* ******************************************************************** */ > > lookup("aliases"); > > > # does the user wish redirection on no availability? (i.e., is he > # in the voicemail group?) -- determine it now and store it in > # flag 4, before we rewrite the flag using UsrLoc > > if (is_user_in("Request-URI", "voicemail")) { > log(1, "requested user is in voicemail group \n"); > setflag(4); > }; > > > # native SIP destinations are handled using our USRLOC DB: > > # LOOKUP (location)!!!! > > if (!lookup("location")) { > log(1,"unable to locate user X ... sending to route(4)! \n"); > # handle user which was not found > route(4); > break; > }; > > ### Test if UAS are in the same NAT: > > # get the host part of the final uri (IP part) and store it in AVP ID 13 > > avp_write("$ruri/domain", "i:13"); > if (avp_check("i:13","eq/$src_ip/i")) { > log(1, "source IP is the same as destination IP\n"); > route(3); > break; > }; > avp_delete("i:13/g"); > > > }; # End of "if(uri==myself)" > > > append_hf("P-hint: usrloc applied\r\n"); > route(1); > > # if user is on-line and is in Voicemail group, enable redirection > if (method == "INVITE" && isflagset(4)) { > log(1, "invite for voicemail user->initiate failureroute[1]\n"); > t_on_failure("1"); > }; > } > > ### ##### ####### ########## - ROUTES - ############### ################# ################## > > route[1] > { > > # if client or server know to be behind a NAT, enable relay > if (isflagset(6)) { > log(1, "Route1: force rtp proxy!!!\n"); > force_rtp_proxy(); > }; > > # NAT processing of replies; apply to all transactions (for example, > # re-INVITEs from public to private UA are hard to identify as > # NATed at the moment of request processing); look at replies > t_on_reply("1"); > > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > if (!t_relay()) { > sl_reply_error(); > }; > log(1, "Route[1]: Send it out now!!!\n"); > break; > } > > # !! Nathelper > onreply_route[1] { > # NATed transaction ? > # Not all 2xx messages have a content body so here we > # make sure our Content-Length > 0 to avoid a parse error > > if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { > fix_nated_contact(); > if (!search("^Content-Length:\ 0")) { > force_rtp_proxy(); > }; > # otherwise, is it a transaction behind a NAT and we did not > # know at time of request processing ? (RFC1918 contacts) > } else if (nat_uac_test("1")) { > fix_nated_contact(); > }; > } > > # -------------- SIP-to-PSTN call routed --------------------- > > route[2]{ > log(1,"route[2]:SIP-to-GW call routed \n"); > if(!t_relay()){ > sl_reply_error(); > }; > log(1, "Route[2]: Send it out now!!!\n"); > } > > # -------------- Same NAT Call Routing (no force rtpproxy) ---- > > route[3]{ > log(1," route[3]: UAs are in the same nat, NO force_rtp_proxy "); > > # What do I have to do here? > > if(!t_relay()){ > sl_reply_error(); > }; > log(1, "Route[3]: Send it out now!!!\n"); > } > > # --------------- Handling of Unavailable user ---------------- > route[4] { > > # non-Voip -- just send "off-line" > if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" || method=="BYE" || method=="OPTIONS")) { > sl_send_reply("404", "Not Found"); > acc_db_request("404 Not Found", "missed_calls"); > log(1, "acc 404 Not Found 1 \n"); > break; > }; > > if (!isflagset(4) && !method=="OPTIONS" && !method=="ACK" && !method=="BYE") { > sl_send_reply("404", "Not Found and no voicemail turned on !! "); > acc_db_request("404 Not Found", "missed_calls"); > log(1, " acc 404 Not Found and no voicemail \n"); > break; > }; > > # forward to voicemail adding prefix to simplify asterisk "extension.conf" > prefix("vm"); > acc_db_request("404 Not Found -> Vm", "missed_calls"); > rewritehostport("192.168.2.131:5070"); > t_relay_to_udp("192.168.2.131", "5070"); > } > > # if forwarding downstream did not succeed, try voicemail running at Asterisk > > failure_route[1]{ > if (t_check_status("408")){ > # revert_uri (); back to the original URI, makes me loose all lookup/rewrite stuff > prefix("vm"); > rewritehostport ("192.168.2.131:5070"); > acc_db_request("408 Timeout -> Vm", "missed_calls"); > append_branch(); > t_relay(); > break; > } else if (t_check_status("486")){ > # revert_uri (); back to the original URI, makes me loose all lookup/rewrite stuff > prefix("vm"); > rewritehostport ("192.168.2.131:5070"); > acc_db_request("486 Busy -> Vm", "missed_calls"); > append_branch(); > t_relay(); > break; > } > } > > ### The End ###

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Re: [Serusers] Changing Sip Port
by Voipers Portugal 14 Dec '05

14 Dec '05

There are 2 port setting that you can change. Either you change the signalling port to contact SER: when specifying the SIP Server just do: sip_server:port In this case, if you change the port, make sure that your SER Server is listening on that port as well. Or you can change the listen port of your Softphone. In this case, you don't need to alter anything else. Not sure about what you wanted. Can you provide the SIP messages log? Jose Simoes On 12/13/05, Juan <juan(a)uwtcallback.com> wrote: > > > Hi Jose. > I tried your way but I did not get it registred. I used many ports but > couldn't get it registred. > I am not using any authentication and when using port 5060 the softphone > registres OK. > My SER and my softphone (X- lite) are on the same network. > Any ideas? > Thanks in advance, > > Juan Ferrari > United World Telecom > > > ------------------------------ > *From:* serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] *On > Behalf Of *Voipers Portugal > *Sent:* Saturday, December 10, 2005 8:48 AM > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Changing Sip Port > > > Most softphones allow that. When you are specifying a SIP server, for > default (normally) it uses 5060. If you want to alter this port just do the > following: > > ser_IP_address:port_number > > 127.0.0.1:5090 > > Jose Simoes > > p.s. this works with X-Lite > > On 12/5/05, Andrey Kouprianov <andrey.kouprianov(a)gmail.com> wrote: > > > > I guess it depends on your softphone... if it chooses to communicate > > from port 5060, there's nothing you can do about it, unless you can > > change it in the phone's settings... > > Some softphones, for instance, get a random port... > > > > On 12/5/05, sagar <vids_cs(a)yahoo.com> wrote: > > > Hi All, > > > > > > I am using ser 0.8.14 +asterisk B2bua. > > > > > > Currently softphone is registering on standard sip port 5060 . > > > Is it possibe to use any other port like 5090 or 8069 > > > > > > > > > I tried to do so by changing the listining port to 5090 in ser.cfgbut in > > > case the softphone is not getting register. > > > > > > > > > Here is the ser log > > > > > > > > > 0(11348) SIP Request: > > > 0(11348) method: <REGISTER> > > > 0(11348) uri: <sip: 192.168.1.100> > > > 0(11348) version: <SIP/2.0> > > > 0(11348) parse_headers: flags=1 > > > 0(11348) end of header reached, state=5 > > > 0(11348) parse_headers: Via found, flags=1 > > > 0(11348) parse_headers: this is the first via > > > 0(11348) After parse_msg... > > > 0(11348) preparing to run routing scripts... > > > 0(11348) DEBUG : is_maxfwd_present: searchi ng for max_forwards > > header > > > 0(11348) parse_headers: flags=128 > > > 0(11348) DEBUG: is_maxfwd_present: value = 70 > > > 0(11348) parse_headers: flags=8 > > > 0(11348) DEBUG: add_param: > > > tag=fa37b062306344d68023aed26289e820 > > > 0(11348) DEBUG: add_param: epid=f657995ecc > > > 0(11348) end of header reached, state=29 > > > 0(11348) parse_headers: flags=256 > > > 0(11348) end of header reached, state=9 > > > 0(11348) DEBUG: get_hdr_field: <To> [37]; > > > uri=[sip:2851816416@192.168.1.100:5090] > > > 0(11348) DEBUG: to body [<sip:2851816416@192.168.1.100:5090> > > > ] > > > 0(11348) get_hdr_field: cseq <CSeq>: <1> <REGISTER> > > > 0(11348) DEBUG: get_hdr_body : content_length=0 > > > 0(11348) found end of header > > > 0(11348) find_first_route(): No Route headers found > > > 0(11348) loose_route(): There is no Route HF > > > 0(11348) parse_headers: flags=64 > > > 0(11348) parse_headers: flags=64 > > > 0(11348) parse_headers: flags=33554432 > > > 0(11348) check_self - checking if host==us: 13==13 && [ > > 192.168.1.100] == > > > [192.168.1.100 ] > > > 0(11348) check_self - checking if port 5090 matches port 5060 > > > 0(11348) check_self: host != me > > > 0(11348) DEBUG: t_addifnew: msg id=1 , global msg id=0 , T on > > > entrance=0xffffffff > > > 0(11348) parse_headers: flags=-1 > > > 0(11348) parse_headers: flags=60 > > > 0(11348) t_lookup_request: start searching: hash=58712, isACK=0 > > > 0(11348) DEBUG: proceeding to pre-RFC3261 transaction matching > > > 0(11348) DEBUG: t_lookup_request: no transaction found > > > 0(11348) DEBUG: mk_proxy: doing DNS lookup... > > > 0(11348) check_via_address(192.168.1.2, 192.168.1.2, 0) > > > 0(11348) DEBUG: add_to_tail_of_timer[4]: 0xb6165fe8 > > > 0(11348) DEBUG: add_to_tail_of_timer[0]: 0xb6165ffc > > > 0(11348) SER: new transacti on fwd'ed > > > 0(11348) DEBUG:destroy_avp_list: destroing list (nil) > > > 0(11348) receive_msg: cleaning up > > > > > > Please advise. > > > > > > Thanks In Advance. > > > > > > > > > ________________________________ > > > Yahoo! Personals > > > Single? There's someone we'd like you to meet. > > > Lots of someones, actually. Yahoo! Personals > > > > > > > > > _______________________________________________ > > > 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 > > > >

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Re: [Serusers] Please, help with SUBSCRIBE timeouts
by Andrey Kouprianov 14 Dec '05

14 Dec '05

yes.. thanx On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > I know such situations from my work too. If the problems show again let me > know... > Vaclav > > On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > > > Thanx a lot for understanding. :)))) > > > > On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > >> :-) It happens... > >> Vaclav > >> > >> On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > >> > >>> And then again my UA is working OK now... Im just stunned... I have a > >>> deadline for this app. Have to present it on coming friday... Hehe... > >>> If it fails to SUBSCRIBE... I'll be in trouble... > >>> > >>> On 12/14/05, Andrey Kouprianov <andrey.kouprianov(a)gmail.com> wrote: > >>>> Yes, i've noticed that too... And yes my config implies statefull > >>>> forwarding.... I think it's my mistake... I was trying with forward > >>>> instead of t_relay at the time I generated traces... > >>>> > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >>> > >> > > >

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[Serusers] using pa module
by lo ™ 14 Dec '05

14 Dec '05

hi all, i tried to use the presence module "pa.so" (without jabber support or xcap authorization) but i don't understand which functionalities have been fixed and which are currently unstable... i'm using eyebeam and i follow these "Configuring Presence Agent" instructions: http://pals.internet2.edu/trials/pic-ser/eyeBeamInstall.html this is part of my ser.cfg: if (method=="PUBLISH") { if (!t_newtran()) { log(1,"ser.cfg: PUBLISH: t_newtran error\n"); sl_reply_error(); }; log(1,"ser.cfg: PUBLISH: handle_publish\n"); handle_publish("registrar"); break; } else if (method=="SUBSCRIBE") { if (!t_newtran()) { log(1,"ser.cfg: SUBSCRIBE: newtran error\n"); sl_reply_error(); }; log(1,"ser.cfg: SUBSCRIBE: handle_subscription\n"); handle_subscription("registrar"); break; }; is it correct? i would like to see my contacts to eventually change state... is it possible? thank you!

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[Serusers] SIP call hold/unhold
by tcchan 14 Dec '05

14 Dec '05

Dear All, I encountered a UA that behave as follow: UA1 call UA2 UA1 ---- invite with SDP -----> UA2 UA1 <------OK with SDP --------UA2 UA1 hold the call, UA1 ------invite (hold) ------>UA2 UA1 <--------OK but not SDP ---UA2 UA1 unhold the call UA1 ------invite ----------->UA2 UA1 <-------OK but also no SDP---UA2 I found that most ATA(UA) would attached a SDP when reply an OK whether the call is hold or unhold. But there is one ATA that does not (UA2). I noticed that some ATA--Linksys (UA1) would send a BYE to UA2 if it receive a OK with no SDP in the hold case. My Question is : Is UA2 behave according to RFC? Would deeply appreciate if someone could let me know the answer as I always have trouble reading RFCs... Thanks fot any help available. Regards, TC Chan

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Re: [Serusers] end the session if no media-conclusion
by Hakan YASTI 14 Dec '05

14 Dec '05

Dear All, After studied on different scenarios, we can not end the session when there is no rtp on media channels. I have heard that some media proxies can do this (like nextone,acmepacket) but I did not test it. As a brief, if the bye will not come from one of the legs because of network problems, you do not have chance to end the session by checking the rtp packets. How did we solve the problem ? Now we are using a session controller. when a session starts, we start a timer, after 1 hour, if the session is going on we are sending a notify about thr session to the network control center. They can kill the session through the session controller. In any case, after 2 hours later if the session is again going on, we kill it automatically from the session controller and send a notification mail to the sysadmin which contains the session information. In this scenario, my customers do not have sessions more than 2 hours. ----- Original Message ----- From: Greger V. Teigre To: Hakan YASTI Sent: Thursday, November 17, 2005 9:50 AM Subject: Re: [Serusers] end the session if no media Good luck :-) And please copy serusers on your conclusions. g-) ----- Original Message ----- From: Hakan YASTI To: Greger V. Teigre Sent: Wednesday, November 16, 2005 1:46 PM Subject: Re: [Serusers] end the session if no media Dear Greger, sorry for late reply. I was on vacation. The scenario is like that. A ( ATA ) ----------------> send INVITE to SER ------------> SER INVITES to session controller --------> session controller invites to ---------> soft switch --------------call setup to ------PSTN B. After signalling ok, the media is also proxied by SER. A Side ------> ------ media goes to SER --<----------media ----------->Media Gateway--------> B side. Possible scenario ; 1-) The BYE is not coming from A side. At that time I see the session on session controller as going on. But infact it is abondened. You are right when the call is on hold. at that time the rtp will not go also. For this session controller must understand when it is on hold.But in any case, I do not know, how will SER end if there is no media session. As I see, the media is controlled by SER. when the bye has come the media session is ended. In fact, When I study on it, I saw that in some cases I saw that, the BYE is coming to soft switch but soft switch send s reply as 500 INTERNAL Error. This is an also issue is related with my soft switch. I have opened a case for this also. Thanks for your interest. I will inform you about the result. Best Regards, ----- Original Message ----- From: Greger V. Teigre To: Hakan YASTI ; serusers(a)lists.iptel.org ; ritesh.j(a)net4.in Sent: Friday, October 28, 2005 6:53 AM Subject: Re: [Serusers] end the session if no media What I'm saying is that you probably want to differ between a long call that after 59 minutes is put on hold and a terminated call where the BYE never came. Both will show no RTP (dependent on UA). Your session controller may be able to tell you if the call is on hold. With regards to No transaction exist: Do you send the BYE directly to the UA and the UA replies with this? If so, I wouldn't worry too much, it just means that you were right in terminating the call because the UA does know of any call. I'm a bit uncertaint about the error though, looks like you may get it from ser (if you send through ser), then the BYE never reaches the UA. (I would expect the UA to respond Call leg doesn't exist). How do you send the BYE that gives this error and to where? g-) ----- Original Message ----- From: Hakan YASTI To: Greger V. Teigre ; serusers(a)lists.iptel.org ; ritesh.j(a)net4.in Sent: Thursday, October 27, 2005 10:25 AM Subject: RE: [Serusers] end the session if no media Yes, I know about it. I can see the active sessions from the session.py. But do not know how to end the seesion, if there is no rtp session and the bye has not come. My session con troller does not check the rtp session because the media is proxied at SER. As I know, may be I am wrong, the ser is managing the media proxy. Ser says to media proxy to start when a call is proxied, and end when the call is finished. Do you suggest a SBC that can do this function..? Thanks, Best Regards, ------------------------------------------------------------------------ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Thursday, October 27, 2005 9:20 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media session.py will show if the session is active. Also when a call is put on hold, there will be no rtp stream. I don't know if your SBC handles that. Or is your question something else? g-) ----- Original Message ----- From: Hakan YASTI To: Greger V. Teigre ; serusers(a)lists.iptel.org ; ritesh.j(a)net4.in Sent: Wednesday, October 26, 2005 09:18 AM Subject: RE: [Serusers] end the session if no media Hi Teigre, Thanks for your reply and kind suggestions. In fact, I am using a session controller for sip calls. And it alerts me via email when a sessiıon is reached for about 60 minutes. At that time, I check it manually and kil the session if the session is abondened. By sending BYE messages to both legs. From the PSTN side I get the reply what I assume ( 200 OK ) but from the sip side I get No transaction exist. Because the session was ended before and session controller couldn’t get the BYE message for some reasons. ( Network failure or something else ). What I am looking for that, from the media proxy, I see the active rtp session when I use media proxy for the calls. And I assume that, when the call has finished and couldn’t send or get BYE message, there will be no rtp stream going on. May be I am wrong for this. If rtp session is still open when the a side hunp up and couldn’t send bye , there is nothing to do except killing the session with sending bye messages to both sides, because there is no way to understand the call is active or not. Best Regards, Hakan ---------------------------------------------------------------------- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, October 26, 2005 10:00 AM To: Hakan YASTI; serusers(a)lists.iptel.org; ritesh.j(a)net4.in Subject: Re: [Serusers] end the session if no media Hakan and Ritesh, It seems to me that you are both asking about the same thing. This topic has been discussed many times on serusers. Search google (site:mail.iptel.org) for "BYE stateful sipsak", I'm pretty sure you will get some info. Also, this FAQ could help: http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2 g-) ----- Original Message ----- From: Hakan YASTI To: serusers(a)lists.iptel.org Sent: Thursday, October 13, 2005 09:13 AM Subject: [Serusers] end the session if no media Hi, I wonder, is there any way, to end the session when the media has ended. Sometimes, we have problem at BYE message. For some reasons, the bye is not coming and may be it is coming wrong. At that time, the media is ended but the call is seen as active. I am using ser and using media proxy. Any suggestion for this issue ? Best Regards, Hakan YASTI NGN Engineer Teletek Telekomünikasyon Hizmetleri A.Þ. Ayazma Dere Cad. Akþit Plaza No:12 Fulya, Beþiktaþ ISTANBUL - TURKEY Tel : +90 212 310 22 34 Fax : +90 212 227 8700 GSM : +90 533 669 06 84 E-mail: hakan.yasti(a)teletek.net http://www.teletek.net -------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir. Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without our companys' authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Our Company makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Our Company. Yasal Uyarı: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Şirketimizin onayı olmaksızın içeriği kopyalanamaz, üçüncü kişilere açıklanamaz veya iletilemez . Bu mesajın gönderilmek istendiği kişi değilseniz (ya da bu e-posta’yı yanlışlıkla aldıysanız), lütfen yollayan kişiyi haberdar ediniz ve mesajı sisteminizden derhal siliniz. Şirketimiz bu mesajın içerdigi bilgilerin doğruluğu veya eksiksiz olduğu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından sorumlu değildir. Bu mesajın içeriği yazarına ait olup, Şirketimizin görüşlerini içermeyebilir. -------------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] subscription
by hgaillac-sip＠yahoo.fr 14 Dec '05

14 Dec '05

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1 0

[Serusers] Help installation of SER!
by carlos moreno 14 Dec '05

14 Dec '05

Good morning, friends, nice to meet you, My name is Carlos Manuel Moreno, from Venezuela, has anyone installed SER, from a RPM file? Which is the correct process? Thanks in advance, Carlos _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 2

Re: [Serusers] Please, help with SUBSCRIBE timeouts
by Andrey Kouprianov 14 Dec '05

14 Dec '05

Yes, that's correct... The UA on that IP address is registered and running... On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > From the trafic it seems, that the authorized SUBSCRIBE is forwarded to > 203.159.32.38. Isn't it correct? > Vaclav > > On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > > > Hi, > > > > Yes, i have tried w/o authorization, but the result is the same. > > The traces are from the client side... As you suggested I dumped the > > traces on the server side with ngrep on both network interface and > > loopback... Loopback yielded no outcome whatsoever (ngrep -W byline -d > > lo0 port 5060). Here are the traces from the from the network > > iterface.... > > > > interface: fxp0 (203.159.31.0/255.255.255.0) > > filter: (ip or ip6) and ( port 5060 ) > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 1 SUBSCRIBE. > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK6789ffc624cbb73dbc918bda9a71956f. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.39:5060 > > SIP/2.0 401 Unauthorized. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 1 SUBSCRIBE. > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.8ccb. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK6789ffc624cbb73dbc918bda9a71956f. > > WWW-Authenticate: Digest realm="sip.interlab.ait.ac.th", > > nonce="439fcec8acac4e38ab93d42a6c610db886870639". > > Server: Sip EXpress router (0.9.3 (i386/freebsd)). > > Content-Length: 0. > > Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=279 > > req_src_ip=203.159.32.39 req_src_port=5060 > > in_uri=sip:12345@sip.interlab.ait.ac.th > > out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1". > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.32.39:5060 -> 203.159.31.36:5060 > > SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 70. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > # > > U 203.159.31.36:5060 -> 203.159.32.38:5060 > > SUBSCRIBE sip:12345@203.159.32.38:5060 SIP/2.0. > > Call-ID: 71c1bf26356a141742e567a5e9df81e1(a)203.159.32.39. > > CSeq: 2 SUBSCRIBE. > > To: <sip:12345@sip.interlab.ait.ac.th>. > > Via: SIP/2.0/UDP 203.159.31.36;branch=0. > > Via: SIP/2.0/UDP > > 203.159.32.39:5060;branch=z9hG4bK77df80e3df85663fd666b9fcc970b54b. > > Max-Forwards: 16. > > Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE. > > User-Agent: IntERLab User Agent 1.0a. > > Event: presence. > > Expires: 600. > > Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml. > > Contact: <sip:andrey@203.159.32.39:5060>. > > Authorization: Digest > > username="andrey",realm="sip.interlab.ait.ac.th",nonce="439fcec8acac4e38ab93d42a6c610db886870639",uri="sip:12345@sip.interlab.ait.ac.th",response="414329461496b84d3b006a25d11aabab". > > From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=6278953. > > Content-Length: 0. > > . > > > > exit > > 100954 received, 0 dropped > > > > It seems that SER does not even reply with timeout... The timeout is > > probably generated by my JAIN-SIP... Im not sure, anyway... > > > > What is happening here??? > > > > > > > > > > > > On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > >> Hello, > >> have you tried it without authorization? Any changes in behaviour? > >> The network dump is from the machine with SER or from the client? > > (Could > >> you try do such dump on machine with SER on all interfaces including > >> loopback?) > >> Vaclav > >> > >> On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > >> > >>> Hi, > >>> > >>> This is my 3rd email on this issue.. Please, help me with it. > >>> > >>> I am having the problem with SUBSCRIBE requests in ser 0.9.3. I am not > >>> using PA module, > >>> because my PA and Watcher are located on the client side. > >>> > >>> Whenever I try to subscribe to a presence with any entity OTHER than > >>> myself, SER seems to receive the SUBSCRIBE request, however, it does > >>> not forward it. Instead my application keeps on retransmitting > >>> SUBSCRIBEs until SER replies with 408 Timeout... The same things > >>> happen with NOTIFYs. > >>> > >>> I must also mention that it is NOT all the time that SER replies with > >>> timeout... Just most of the times. And only sometimes it lets the > >>> SUBSCRIBE through (for that, I have to restart my app. again and > >>> again, until I get the right result). > >>> > >>> I included ser.cfg + traces. I've tried to use stateless forwarding ( > >>> forward(uri:host, uri:port) ) instead of stateful (as shown in my > >>> ser.cfg), but it didnt work either. > >>> > >>> Please, tell me what is wrong, because I'm out of ideas here... I > >>> would settle even for reinstalling SER, if it's necessary. > >>> > >>> Here's ser.cfg: > >>> > >>> # ----------- global configuration parameters ------------------------ > >>> debug=3 # debug level (cmd line: -dddddddddd) > >>> fork=yes > >>> log_stderror=no # (cmd line: -E) > >>> > >>> /* Uncomment these lines to enter debugging mode > >>> fork=no > >>> log_stderror=yes > >>> */ > >>> > >>> check_via=no # (cmd. line: -v) > >>> dns=no # (cmd. line: -r) > >>> rev_dns=no # (cmd. line: -R) > >>> #port=5060 > >>> #children=4 > >>> fifo="/tmp/ser_fifo" > >>> fifo_mode=0766 > >>> > >>> sip_warning=yes > >>> server_signature=yes > >>> > >>> alias=sip.interlab.ait.ac.th > >>> > >>> # ------------------ module loading ---------------------------------- > >>> # Uncomment this if you want to use SQL database > >>> 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/textops.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.so" > >>> loadmodule "/usr/local/lib/ser/modules/uri_db.so" > >>> loadmodule "/usr/local/lib/ser/modules/exec.so" > >>> loadmodule "/usr/local/lib/ser/modules/pa.so" > >>> > >>> # ----------------- setting module-specific parameters --------------- > >>> modparam("usrloc", "db_mode", 2) > >>> modparam("auth_db", "calculate_ha1", yes) > >>> modparam("auth_db", "password_column", "password") > >>> modparam("rr", "enable_full_lr", 1) > >>> > >>> # ------------------------- request routing logic ------------------- > >>> route{ > >>> > >>> # initial sanity checks -- messages with > >>> # max_forwards==0, or excessively long requests > >>> if (!mf_process_maxfwd_header("10")) { > >>> log(1, "Message has too many hops. Message disgarted"); > >>> sl_send_reply("483","Too Many Hops"); > >>> break; > >>> }; > >>> if (msg:len >= 4096 ) { > >>> log(1, "Message too big. Message disgarted"); > >>> sl_send_reply("513", "Message too big"); > >>> break; > >>> }; > >>> > >>> #if (method!="REGISTER" ) record_route(); > >>> > >>> #if (loose_route()) { > >>> # # mark routing logic in request > >>> # append_hf("P-hint: rr-enforced\r\n"); > >>> # route(1); > >>> # break; > >>> #}; > >>> > >>> if (!uri==myself) { > >>> # mark routing logic in request > >>> append_hf("P-hint: outbound\r\n"); > >>> route(1); > >>> break; > >>> }; > >>> > >>> if (uri==myself) { > >>> if (method=="REGISTER") { > >>> if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { > >>> log(1, "Authentication for REGISTER"); > >>> www_challenge("sip.interlab.ait.ac.th", "0"); > >>> break; > >>> }; > >>> > >>> #if(!is_user("replicator") & !check_to() ){ > >>> # log(1, "Unregistered user registration attempt\n"); > >>> # sl_send_reply("403", "Forbidden - Registered > >>> users only"); > >>> # break; > >>> #}; > >>> > >>> if(!save("location")) { > >>> log(1, "Error saving AOR"); > >>> sl_reply_error(); > >>> }; > >>> break; > >>> }; > >>> > >>> lookup("aliases"); > >>> if (uri!=myself) { > >>> append_hf("P-hint: outbound alias\r\n"); > >>> route(1); > >>> break; > >>> }; > >>> > >>> #Handle local offline or non-existent users > >>> if(method=="INVITE") { > >>> if(!lookup("location") ) { > >>> log(1, "INVITE for an offline user processing"); > >>> route(3); #offline and non-existent users > >>> break; > >>> }; > >>> }; > >>> > >>> if(method=="SUBSCRIBE") { > >>> if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { > >>> log(1, "Authentication for SUBSCRIBE"); > >>> www_challenge("sip.interlab.ait.ac.th", "0"); > >>> break; > >>> }; > >>> > >>> if(!lookup("location") ) { > >>> log(1, "No registered user found for SUBSCRIBE"); > >>> > >>> sl_send_reply("404", "Not Found"); > >>> break; > >>> }; > >>> }; > >>> > >>> # native SIP destinations are handled using our USRLOC DB > >>> if(method=="NOTIFY") { > >>> if (!lookup("location")) { > >>> sl_send_reply("404", "Not Found"); > >>> break; > >>> }; > >>> > >>> log(1, "NOTIFY reveived"); > >>> }; > >>> }; > >>> > >>> append_hf("P-hint: usrloc applied\r\n"); > >>> route(1); > >>> } > >>> > >>> route[1] > >>> { > >>> # send it out now; use stateful forwarding as it works reliably > >>> # even for UDP2TCP > >>> if (!t_relay()) { > >>> sl_reply_error(); > >>> }; > >>> } > >>> > >>> # ------------ Process offline and non-existent users ---------- > >>> route[3] { > >>> # do not continue if it is a retransmission > >>> if(!t_newtran()) { > >>> sl_reply_error(); > >>> break; > >>> }; > >>> > >>> # use external script to send an email notification in case > >>> # user is offline, or just reply back with 404 (Not Found) if > >>> # the user is not a registered user > >>> if(!exec_msg('/home/andrey/sip_email.sh') ) { > >>> t_reply("404", "Not Found - User does not exist"); > >>> } > >>> else { > >>> t_reply("404", "Not Found - User offline"); > >>> }; > >>> break; > >>> } > >>> > >>> . . . . . . . . > >>> > >>> > >>> Here are the traces: > >>> > >>> Message 1 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 1 SUBSCRIBE > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 2 from 203.159.31.36:5060 to 203.159.32.39:5060 > >>> > >>> > >>> SIP/2.0 401 Unauthorized > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 1 SUBSCRIBE > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.5e75 > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 > >>> WWW-Authenticate: Digest > >>> realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0" > >>> Server: Sip EXpress router (0.9.3 (i386/freebsd)) > >>> Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=17754 > >>> req_src_ip=203.159.32.39 req_src_port=5060 > >>> in_uri=sip:12345@sip.interlab.ait.ac.th > >>> out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1" > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 3 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 4 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 5 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> . . . . . . > >>> > >>> And so it goes, until SER replies with Timeout. > >>> > >>> Can you tell me what is wrong??? > >>> > >>> Andrey. > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >>> > >> > > >

2 11

[Serusers] SER with presence release
by Andrey Kouprianov 14 Dec '05

14 Dec '05

Hi, Dont mean to bother much, but when is stable SER with presence support will be made available?

3 5

[Serusers] SUBSCRIBEs and NOTIFYs timeout
by Andrey Kouprianov 14 Dec '05

14 Dec '05

Hi, Im using ser 0.9.3. The PA and Watcher are located on my UA, so basically my SUBSCRIBEs and NOTIFYs just need to be forwarded between clients. Here's a part of ser.cfg that does the SUBSCRIBE and NOTIFY processing (please, note that I disabled the authentication for now): . . . . . . if(method=="SUBSCRIBE") { if(!lookup("location") ) { log(1, "No registered user found for SUBSCRIBE request"); sl_send_reply("404", "Not Found"); break; }; }; if(method=="NOTIFY") { if (!lookup("location")) { log(1, "No registered user found for NOTIFY request"); sl_send_reply("404", "Not Found"); break; }; }; if( !t_relay() ) { sl_reply_error(); }; . . . . . . . . The problem is that only SOMETIMES SUIBSCRIBEs and NOTIFYs get through without any problem. However, MOST OF THE TIMES they get stuck in the server. The message log does not show any errors and tracesviewer tool shows only retransmissions for the issued requests. In the end SER replies with 408 Timeout response. I have tried to use forward(uri:host, uri:port) instead of t_relay(), but result is the same. I have used eyeBeam to do the presence subscriptions, and the result is the same as with my UA... Can you, please, explain to me what might have happened??? I am out of ideas. Andrey.

2 7

Re: [Serusers] Please, help with SUBSCRIBE timeouts
by Andrey Kouprianov 14 Dec '05

14 Dec '05

So, what could be the problem then??? On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > > It seems that SER does not even reply with timeout... The timeout is > > probably generated by my JAIN-SIP... Im not sure, anyway... > > > > What is happening here??? > > > > Yes, you are right. SER forwards the message but the other side doesn't > respond and the client itself generates a timeout (I think, that timeouts > should be generated by SER too, but may be, that the client has lower > timer values). > Vaclav > > P.S. I missed this part of your message... :-) > > > > > > > > > > > On 12/14/05, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > >> Hello, > >> have you tried it without authorization? Any changes in behaviour? > >> The network dump is from the machine with SER or from the client? > > (Could > >> you try do such dump on machine with SER on all interfaces including > >> loopback?) > >> Vaclav > >> > >> On Wed, 14 Dec 2005, Andrey Kouprianov wrote: > >> > >>> Hi, > >>> > >>> This is my 3rd email on this issue.. Please, help me with it. > >>> > >>> I am having the problem with SUBSCRIBE requests in ser 0.9.3. I am not > >>> using PA module, > >>> because my PA and Watcher are located on the client side. > >>> > >>> Whenever I try to subscribe to a presence with any entity OTHER than > >>> myself, SER seems to receive the SUBSCRIBE request, however, it does > >>> not forward it. Instead my application keeps on retransmitting > >>> SUBSCRIBEs until SER replies with 408 Timeout... The same things > >>> happen with NOTIFYs. > >>> > >>> I must also mention that it is NOT all the time that SER replies with > >>> timeout... Just most of the times. And only sometimes it lets the > >>> SUBSCRIBE through (for that, I have to restart my app. again and > >>> again, until I get the right result). > >>> > >>> I included ser.cfg + traces. I've tried to use stateless forwarding ( > >>> forward(uri:host, uri:port) ) instead of stateful (as shown in my > >>> ser.cfg), but it didnt work either. > >>> > >>> Please, tell me what is wrong, because I'm out of ideas here... I > >>> would settle even for reinstalling SER, if it's necessary. > >>> > >>> Here's ser.cfg: > >>> > >>> # ----------- global configuration parameters ------------------------ > >>> debug=3 # debug level (cmd line: -dddddddddd) > >>> fork=yes > >>> log_stderror=no # (cmd line: -E) > >>> > >>> /* Uncomment these lines to enter debugging mode > >>> fork=no > >>> log_stderror=yes > >>> */ > >>> > >>> check_via=no # (cmd. line: -v) > >>> dns=no # (cmd. line: -r) > >>> rev_dns=no # (cmd. line: -R) > >>> #port=5060 > >>> #children=4 > >>> fifo="/tmp/ser_fifo" > >>> fifo_mode=0766 > >>> > >>> sip_warning=yes > >>> server_signature=yes > >>> > >>> alias=sip.interlab.ait.ac.th > >>> > >>> # ------------------ module loading ---------------------------------- > >>> # Uncomment this if you want to use SQL database > >>> 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/textops.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.so" > >>> loadmodule "/usr/local/lib/ser/modules/uri_db.so" > >>> loadmodule "/usr/local/lib/ser/modules/exec.so" > >>> loadmodule "/usr/local/lib/ser/modules/pa.so" > >>> > >>> # ----------------- setting module-specific parameters --------------- > >>> modparam("usrloc", "db_mode", 2) > >>> modparam("auth_db", "calculate_ha1", yes) > >>> modparam("auth_db", "password_column", "password") > >>> modparam("rr", "enable_full_lr", 1) > >>> > >>> # ------------------------- request routing logic ------------------- > >>> route{ > >>> > >>> # initial sanity checks -- messages with > >>> # max_forwards==0, or excessively long requests > >>> if (!mf_process_maxfwd_header("10")) { > >>> log(1, "Message has too many hops. Message disgarted"); > >>> sl_send_reply("483","Too Many Hops"); > >>> break; > >>> }; > >>> if (msg:len >= 4096 ) { > >>> log(1, "Message too big. Message disgarted"); > >>> sl_send_reply("513", "Message too big"); > >>> break; > >>> }; > >>> > >>> #if (method!="REGISTER" ) record_route(); > >>> > >>> #if (loose_route()) { > >>> # # mark routing logic in request > >>> # append_hf("P-hint: rr-enforced\r\n"); > >>> # route(1); > >>> # break; > >>> #}; > >>> > >>> if (!uri==myself) { > >>> # mark routing logic in request > >>> append_hf("P-hint: outbound\r\n"); > >>> route(1); > >>> break; > >>> }; > >>> > >>> if (uri==myself) { > >>> if (method=="REGISTER") { > >>> if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { > >>> log(1, "Authentication for REGISTER"); > >>> www_challenge("sip.interlab.ait.ac.th", "0"); > >>> break; > >>> }; > >>> > >>> #if(!is_user("replicator") & !check_to() ){ > >>> # log(1, "Unregistered user registration attempt\n"); > >>> # sl_send_reply("403", "Forbidden - Registered > >>> users only"); > >>> # break; > >>> #}; > >>> > >>> if(!save("location")) { > >>> log(1, "Error saving AOR"); > >>> sl_reply_error(); > >>> }; > >>> break; > >>> }; > >>> > >>> lookup("aliases"); > >>> if (uri!=myself) { > >>> append_hf("P-hint: outbound alias\r\n"); > >>> route(1); > >>> break; > >>> }; > >>> > >>> #Handle local offline or non-existent users > >>> if(method=="INVITE") { > >>> if(!lookup("location") ) { > >>> log(1, "INVITE for an offline user processing"); > >>> route(3); #offline and non-existent users > >>> break; > >>> }; > >>> }; > >>> > >>> if(method=="SUBSCRIBE") { > >>> if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { > >>> log(1, "Authentication for SUBSCRIBE"); > >>> www_challenge("sip.interlab.ait.ac.th", "0"); > >>> break; > >>> }; > >>> > >>> if(!lookup("location") ) { > >>> log(1, "No registered user found for SUBSCRIBE"); > >>> > >>> sl_send_reply("404", "Not Found"); > >>> break; > >>> }; > >>> }; > >>> > >>> # native SIP destinations are handled using our USRLOC DB > >>> if(method=="NOTIFY") { > >>> if (!lookup("location")) { > >>> sl_send_reply("404", "Not Found"); > >>> break; > >>> }; > >>> > >>> log(1, "NOTIFY reveived"); > >>> }; > >>> }; > >>> > >>> append_hf("P-hint: usrloc applied\r\n"); > >>> route(1); > >>> } > >>> > >>> route[1] > >>> { > >>> # send it out now; use stateful forwarding as it works reliably > >>> # even for UDP2TCP > >>> if (!t_relay()) { > >>> sl_reply_error(); > >>> }; > >>> } > >>> > >>> # ------------ Process offline and non-existent users ---------- > >>> route[3] { > >>> # do not continue if it is a retransmission > >>> if(!t_newtran()) { > >>> sl_reply_error(); > >>> break; > >>> }; > >>> > >>> # use external script to send an email notification in case > >>> # user is offline, or just reply back with 404 (Not Found) if > >>> # the user is not a registered user > >>> if(!exec_msg('/home/andrey/sip_email.sh') ) { > >>> t_reply("404", "Not Found - User does not exist"); > >>> } > >>> else { > >>> t_reply("404", "Not Found - User offline"); > >>> }; > >>> break; > >>> } > >>> > >>> . . . . . . . . > >>> > >>> > >>> Here are the traces: > >>> > >>> Message 1 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 1 SUBSCRIBE > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 2 from 203.159.31.36:5060 to 203.159.32.39:5060 > >>> > >>> > >>> SIP/2.0 401 Unauthorized > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 1 SUBSCRIBE > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.5e75 > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 > >>> WWW-Authenticate: Digest > >>> realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0" > >>> Server: Sip EXpress router (0.9.3 (i386/freebsd)) > >>> Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=17754 > >>> req_src_ip=203.159.32.39 req_src_port=5060 > >>> in_uri=sip:12345@sip.interlab.ait.ac.th > >>> out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1" > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 3 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 4 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> > >>> > >>> Message 5 from 203.159.32.39:5060 to 203.159.31.36:5060 > >>> > >>> > >>> SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 > >>> Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 > >>> CSeq: 2 SUBSCRIBE > >>> To: <sip:12345@sip.interlab.ait.ac.th> > >>> Via: SIP/2.0/UDP > >>> 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c > >>> Max-Forwards: 70 > >>> Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE > >>> User-Agent: IntERLab User Agent 1.0a > >>> Event: presence > >>> Expires: 600 > >>> Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml > >>> Contact: <sip:andrey@203.159.32.39:5060> > >>> Authorization: Digest > >>> username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" > >>> From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 > >>> Content-Length: 0 > >>> > >>> . . . . . . > >>> > >>> And so it goes, until SER replies with Timeout. > >>> > >>> Can you tell me what is wrong??? > >>> > >>> Andrey. > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >>> > >> > > >

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[Serusers] Please, help with SUBSCRIBE timeouts
by Andrey Kouprianov 14 Dec '05

14 Dec '05

Hi, This is my 3rd email on this issue.. Please, help me with it. I am having the problem with SUBSCRIBE requests in ser 0.9.3. I am not using PA module, because my PA and Watcher are located on the client side. Whenever I try to subscribe to a presence with any entity OTHER than myself, SER seems to receive the SUBSCRIBE request, however, it does not forward it. Instead my application keeps on retransmitting SUBSCRIBEs until SER replies with 408 Timeout... The same things happen with NOTIFYs. I must also mention that it is NOT all the time that SER replies with timeout... Just most of the times. And only sometimes it lets the SUBSCRIBE through (for that, I have to restart my app. again and again, until I get the right result). I included ser.cfg + traces. I've tried to use stateless forwarding ( forward(uri:host, uri:port) ) instead of stateful (as shown in my ser.cfg), but it didnt work either. Please, tell me what is wrong, because I'm out of ideas here... I would settle even for reinstalling SER, if it's necessary. Here's ser.cfg: # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_mode=0766 sip_warning=yes server_signature=yes alias=sip.interlab.ait.ac.th # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/textops.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.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/pa.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { log(1, "Message has too many hops. Message disgarted"); sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 4096 ) { log(1, "Message too big. Message disgarted"); sl_send_reply("513", "Message too big"); break; }; #if (method!="REGISTER" ) record_route(); #if (loose_route()) { # # mark routing logic in request # append_hf("P-hint: rr-enforced\r\n"); # route(1); # break; #}; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { log(1, "Authentication for REGISTER"); www_challenge("sip.interlab.ait.ac.th", "0"); break; }; #if(!is_user("replicator") & !check_to() ){ # log(1, "Unregistered user registration attempt\n"); # sl_send_reply("403", "Forbidden - Registered users only"); # break; #}; if(!save("location")) { log(1, "Error saving AOR"); sl_reply_error(); }; break; }; lookup("aliases"); if (uri!=myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; #Handle local offline or non-existent users if(method=="INVITE") { if(!lookup("location") ) { log(1, "INVITE for an offline user processing"); route(3); #offline and non-existent users break; }; }; if(method=="SUBSCRIBE") { if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { log(1, "Authentication for SUBSCRIBE"); www_challenge("sip.interlab.ait.ac.th", "0"); break; }; if(!lookup("location") ) { log(1, "No registered user found for SUBSCRIBE"); sl_send_reply("404", "Not Found"); break; }; }; # native SIP destinations are handled using our USRLOC DB if(method=="NOTIFY") { if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; log(1, "NOTIFY reveived"); }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # ------------ Process offline and non-existent users ---------- route[3] { # do not continue if it is a retransmission if(!t_newtran()) { sl_reply_error(); break; }; # use external script to send an email notification in case # user is offline, or just reply back with 404 (Not Found) if # the user is not a registered user if(!exec_msg('/home/andrey/sip_email.sh') ) { t_reply("404", "Not Found - User does not exist"); } else { t_reply("404", "Not Found - User offline"); }; break; } . . . . . . . . Here are the traces: Message 1 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 1 SUBSCRIBE From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Content-Length: 0 Message 2 from 203.159.31.36:5060 to 203.159.32.39:5060 SIP/2.0 401 Unauthorized Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 1 SUBSCRIBE From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.5e75 Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 WWW-Authenticate: Digest realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0" Server: Sip EXpress router (0.9.3 (i386/freebsd)) Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=17754 req_src_ip=203.159.32.39 req_src_port=5060 in_uri=sip:12345@sip.interlab.ait.ac.th out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1" Content-Length: 0 Message 3 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 Message 4 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 Message 5 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 . . . . . . And so it goes, until SER replies with Timeout. Can you tell me what is wrong??? Andrey.

2 2

[Users] inbound call failed
by unplug 14 Dec '05

14 Dec '05

Below is the common configuration of the network. Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger) I can make call from softphone to Telephone. However, it is failed to make call from Telephone to softphone. I wonder why it happened and any reference to trace the problem. Anyone have such experience?

2 4

[Serusers] Using SER together with Asterisk
by bharat.sarvan＠wipro.com 14 Dec '05

14 Dec '05

Hello all, I am trying to integrate SER and Asterisk. So could anybody please guide me with the changes and the configuration that I need to in either SER or Asterisk for achieving it. Like for suppose I want to achieve a call being placed from a User 1 registered to SER to User 2 who is registered to Asterisk.. So how do I move ahead to achieve the same. And are there any other new things that one can come up with using SER and Asterisk together which can't be achieved using SER or Asterisk alone? Kindly do let me know. Thanks and Regards, Bharat Sarvan The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

Re: [Serusers] simple click-to-call
by Kapil Dhawan 14 Dec '05

14 Dec '05

If SER is running, that means fifo is ON. >From: Ken Rozinsky <xraycharlie(a)gmail.com> >To: Kapil Dhawan <sersavvy(a)hotmail.com> >Subject: Re: [Serusers] simple click-to-call >Date: Tue, 13 Dec 2005 11:24:58 -0700 > >Thanks for the info. > >A question for you if you have the time: > >The script checks if it can write to /tmp/ser_fifo and fails. >The comment mention having to have the fifo server running. How do I >check if this is running or not? > >I see in the documantation a mention of fifo_db but I don;t think I >need this as am not worried about persistance. > >The permissions on /tmp/ser_fifo are currently prw-rw---- > >I tried running the script as root and get the write error also. > >Thanks again. > >Regards > >Ken > > >On 12/12/05, Kapil Dhawan <sersavvy(a)hotmail.com> wrote: > > Ken > > > > Click-to-dial is present in example directoy as ctd.sh. It takes two > > arguments From and To. To run it from Web-interface, you need to run >this > > command thru script. > > > > Regards > > > > > > >From: Ken Rozinsky <xraycharlie(a)gmail.com> > > >To: serusers(a)lists.iptel.org > > >Subject: [Serusers] simple click-to-call > > >Date: Mon, 12 Dec 2005 17:27:18 -0700 > > > > > >Hello, > > > > > >As an exercise in learning how all this stuff works, I'm trying to set > > >up a simple click-to-call service for our office to use. And > > >potentialy to allow clients to use vai the web site to save them on > > >the long distance charges. > > > > > >I set up Asterisk with the TACI script and got it to all work with an > > >account on gafachi but I noticed that it could only handle one call at > > >a time. > > > > > >I was told that openSer would allow more than one call at a time so > > >I've installed it (basic no mySql) and have it up and running. > > > > > >Now, being new to this I have no idea how to connct to gafachi and do > > >a click-to-call type thing. > > > > > >Would anybody be able to help me with this? > > > > > >Regards > > > > > >_______________________________________________ > > >Serusers mailing list > > >serusers(a)lists.iptel.org > > >http://lists.iptel.org/mailman/listinfo/serusers > > > > _________________________________________________________________ > > Shah Rukh fan? Know all about the Baadshah of Bollywood. On MSN Search > > http://server1.msn.co.in/profile/shahrukh.asp > > > > _________________________________________________________________ How good are you in a Formula One car? Play now http://server1.msn.co.in/sp05/tataracing/onlinegame.asp

1 0

Re: [Serusers] using pa module
by lo ™ 14 Dec '05

14 Dec '05

ok, i set up eyebeam with peer-to-peer mode in the presence agent settings but i don't understand yet if it is close to work or not... thank you, again! this is my log: Maxfwd module- initializing db_url=0x8208c78 db_url=mysql://ser:heslo@localhost/ser db_url.len=31 pa_domain=<domain.tld> pa_mod: use_db=1 db_url.s=mysql://ser:heslo@localhost/ser pa_domain=< domain.tld> pa_mod_init done subscribe_fixup: pdomain name is registrar subscribe_fixup: pdomain name is registrar INFO: udp_init: SO_RCVBUF is initially 108544 INFO: udp_init: SO_RCVBUF is finally 217088 INFO: udp_init: SO_RCVBUF is initially 108544 INFO: udp_init: SO_RCVBUF is finally 217088 INFO: fifo process starting: 28887 SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... handle_subscription() entered get_pres_uri: _puri=sip:ennevu@<domain.tld> handle_subscription() entered get_pres_uri: _puri=sip:lo2@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=11@<domain.tld> presid=1 new_presentity_no_wb=0xb618bdb8 for uri=11@<domain.tld> add_presentity _p=0xb618bdb8 p_uri=11@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=ennevu@< domain.tld> presid=2 new_presentity_no_wb=0xb618be08 for uri=ennevu@<domain.tld> add_presentity _p=0xb618be08 p_uri=ennevu@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=lo2@<domain.tld> presid=3 new_presentity_no_wb=0xb618be60 for uri=lo2@<domain.tld> add_presentity _p=0xb618be60 p_uri=lo2@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=10@<domain.tld> presid=4 new_presentity_no_wb=0xb618beb0 for uri=10@<domain.tld> add_presentity _p=0xb618beb0 p_uri=10@<domain.tld> pdomain_load_presentities: pdomain=registrar presentity uri=lo@<domain.tld> presid=5 new_presentity_no_wb=0xb618bf00 for uri=lo@<domain.tld> add_presentity _p=0xb618bf00 p_uri=lo@<domain.tld> db_read_watcherinfo: _p->uri='11@<domain.tld>' db_read_watcherinfo: _p->uri='11@<domain.tld>' done db_read_watcherinfo: _p->uri='ennevu@<domain.tld>' db_read_watcherinfo: _p->uri='ennevu@<domain.tld>' done db_read_watcherinfo: _p->uri='lo2@<domain.tld>' db_read_watcherinfo: _p->uri='lo2@<domain.tld>' done db_read_watcherinfo: _p->uri='10@<domain.tld>' db_read_watcherinfo: _p->uri='10@<domain.tld>' done db_read_watcherinfo: _p->uri='lo@<domain.tld>' db_read_watcherinfo: _p->uri='lo@<domain.tld>' done new_watcher starting new_watcher: _p->uri=ennevu@<domain.tld> db_new_watcher: watcher->uri=11@<domain.tld> new_watcher: watcher->package=presence new_watcher: getting values: res=0x81019b8 res->n=1 new_watcher: status=active handle_subscription about to return 1: w->event_package=1 w->accept=2 p->flags=4 w->flags=1 w=0xb618c490 new_watcher starting new_watcher: _p->uri=lo2@<domain.tld> db_new_watcher: watcher->uri=11@<domain.tld> new_watcher: watcher->package=presence new_watcher: getting values: res=0x8101998 res->n=1 new_watcher: status=active handle_subscription about to return 1: w->event_package=1 w->accept=2 p->flags=4 w->flags=1 w=0xb618c928 handle_subscription() entered get_pres_uri: _puri=sip:lo@<domain.tld> new_watcher starting new_watcher: _p->uri=lo@<domain.tld> db_new_watcher: watcher->uri=10@<domain.tld> new_watcher: watcher->package=presence new_watcher: getting values: res=0x810a640 res->n=1 new_watcher: status=active handle_subscription about to return 1: w->event_package=1 w->accept=2 p->flags=4 w->flags=1 w=0xb618fb38 handle_subscription() entered get_pres_uri: _puri=sip:11@<domain.tld> new_watcher starting new_watcher: _p->uri=11@<domain.tld> db_new_watcher: watcher->uri=10@<domain.tld> new_watcher: watcher->package=presence new_watcher: getting values: res=0x810a500 res->n=1 new_watcher: status=active handle_subscription about to return 1: w->event_package=1 w->accept=2 p->flags=4 w->flags=1 w=0xb6191980 timer_presentity: _p=0xb618bdb8 11@<domain.tld> flags=4 watchers=0xb6191980 w=0xb6191980 10@<domain.tld>ennevu<sip:10@<domain.tld>> flags=1 notifying 10@<domain.tld> _p->flags=4 _w->event_package=1 _w->accept=2 _w->status=1 send_pidf_notify timer_presentity: _p=0xb618be08 ennevu@<domain.tld> flags=4 watchers=0xb618c490 w=0xb618c490 11@<domain.tld>lo<sip:11@<domain.tld>> flags=1 notifying 11@<domain.tld> _p->flags=4 _w->event_package=1 _w->accept=2 _w->status=1 send_pidf_notify timer_presentity: _p=0xb618be60 lo2@<domain.tld> flags=4 watchers=0xb618c928 w=0xb618c928 11@<domain.tld>lo<sip:11@<domain.tld>> flags=1 notifying 11@<domain.tld> _p->flags=4 _w->event_package=1 _w->accept=2 _w->status=1 send_pidf_notify timer_presentity: _p=0xb618bf00 lo@<domain.tld> flags=4 watchers=0xb618fb38 w=0xb618fb38 10@<domain.tld>ennevu<sip:10@<domain.tld>> flags=1 notifying 10@<domain.tld> _p->flags=4 _w->event_package=1 _w->accept=2 _w->status=1 send_pidf_notify Warning: run_failure_handlers: no UAC support (0, 0) Warning: run_failure_handlers: no UAC support (0, 0) Warning: run_failure_handlers: no UAC support (0, 0) Warning: run_failure_handlers: no UAC support (0, 0) Removing watcher 10@<domain.tld> notifying 10@<domain.tld> _p->flags=0 _w->event_package=1 _w->accept=2 _w->status=1 Removing watcher 11@<domain.tld> notifying 11@<domain.tld> _p->flags=0 _w->event_package=1 _w->accept=2 _w->status=1 Removing watcher 11@<domain.tld> notifying 11@<domain.tld> _p->flags=0 _w->event_package=1 _w->accept=2 _w->status=1 Removing watcher 10@<domain.tld> notifying 10@<domain.tld> _p->flags=0 _w->event_package=1 _w->accept=2 _w->status=1

2 1

[Serusers] Mediaproxy Call hold/unhold
by tcchan 14 Dec '05

14 Dec '05

Dear All, I encountered a problem like this. Case 1: ------- If UA1 calls UA2 using the following: Normal call: UA1 ------invite -------port 30000 -----> UA2 UA1 <--------OK---port 26000--------------UA2 The connection and media is established. Hold the call: UA1------invite ---port 30000 (hold)--> UA2 UA1 <----------OK port 26000----------UA2 The media is in hold state. Unhold the call: UA1------invite ----port 30000--------->UA2 UA1 <---------OK port 260000----------UA2 Media resume. The above mentioned work fine with mediaproxy. Case 2: ------- If UA1 calls UA2 using the following: Normal call: UA1 ------invite -------port 30000 -----> UA2 UA1 <--------OK---port 26000--------------UA2 The connection and media is established. Hold the call: UA1------invite ---port 30000 (hold)--> UA2 UA1 <----------OK port 26000----------UA2 The media is in hold state. Unhold the call: UA1------invite ----port 30001--------->UA2 UA1 <---------OK port 260000----------UA2 *****Noticed that the reinvite has a different port! In this situation, media does not resume and mediaproxy has a "stray hold" session. I tested case2 without flowing through the mediaproxy and found that the call can be hold/unhold successfully. I am not familiar with RFC at all, please help! (1) I would like to know whether it is valid to reinvite with a different port as in case 2, i.e., is the UA1 ill-behave in case 2 when unhold the call with different port? (2) Is the mediaproxy designed to work using the some port when hold/unhold. Thanks you very much for any help available. Regards, TC Chan

1 0

[Users] failure_route.... I could cry...
by Douglas Garstang 14 Dec '05

14 Dec '05

I am trying to have OpenSER forward registration requests to multiple Asterisk boxes with failover. Each Asterisk box has two ip addresses. My logic tries to relay to the first IP on the first Asterisk box (192.168.10.100). If that fails, it tries to relay to the second IP on the first Asterisk box (192.168.10.17). It then goes on and repeats the process with another Asterisk box who's IP's are 192.168.10.10.101 and 192.168.10.8. Actually the first IP address on each Asterisk system is fake (100 and 101)... that's how I am testing that it will fail to the second one. The openser.cfg below is causing this to be logged to messages: Dec 13 17:50:08 bil-pdev-3 openser[22318]: route10: Asterisk-1, NIC-1 Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: Asterisk-2, NIC-1 Dec 13 17:50:08 bil-pdev-3 openser[22318]: ERROR: t_newtran: transaction already in process 0xb6126560 < --- Why? Dec 13 17:50:08 bil-pdev-3 openser[22318]: route20: t_relay returned error <--- Why? Dec 13 17:50:09 bil-pdev-3 openser[22328]: failure21: Failed to register with Asterisk-2, NIC-1 Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: Asterisk-2, NIC-2 Dec 13 17:50:09 bil-pdev-3 openser[22328]: route22: End of routine Dec 13 17:50:10 bil-pdev-3 openser[22322]: User local found Dec 13 17:50:10 bil-pdev-3 openser[22320]: User local found Why am I getting the t_newtran and t_relay errors in there? What am I doing wrong? I'm just calling rewritehostport(next-ip), followed by append_branch() and t_relay. In fact something really weird happens. When I have the code there for route[20], failure_route[21] and route[22], it causes the code for the first three routing blocks, route[10], failure_route[11] and route[12] to act differently. It seems to make route20 fail to connect eventhough it comes AFTER it in the code. I'd really appreciate some help! Am I doing something wrong with rewritehostport and append_branch, because quite frankly the docs are terrible. if ( method == REGISTER ) { if ( !www_authorize("voip.com", "subscriber") ) { www_challenge("voip.com", "0"); }; save("location"); route(10); route(20); # # First Asterisk System. # # Try first IP on this system (fake - will fail) route[10] { xlog ("L_INFO","route10: Asterisk-1, NIC-1"); t_on_failure("11"); rewritehostport("192.168.10.100:5060"); append_branch(); if ( !t_relay() ) { xlog ("L_INFO","route10: t_relay returned error"); } } failure_route[11] { xlog ("L_INFO","failure11: Failed to register with Asterisk-1, NIC-1"); route(12); } # Try second IP on this system (real - should work) route[12] { xlog ("L_INFO","route12: Asterisk-1, NIC-2"); t_on_failure("13"); rewritehostport("192.168.10.17:5060"); append_branch(); if ( !t_relay() ) { xlog ("L_INFO","route12: t_relay returned error"); } xlog ("L_INFO","route12: End of routine"); return; } # # Second Asterisk System # # Try first IP on this system (fake - will fail) route[20] { xlog ("L_INFO","route20: Asterisk-2, NIC-1"); t_on_failure("21"); rewritehostport("192.168.10.101:5060"); append_branch(); if ( !t_relay() ) { xlog ("L_INFO","route20: t_relay returned error"); } } failure_route[21] { xlog ("L_INFO","failure21: Failed to register with Asterisk-2, NIC-1"); route(22); } # Try second IP on this system (real - should work) route[22] { xlog ("L_INFO","route22: Asterisk-2, NIC-2"); t_on_failure("23"); rewritehostport("192.168.10.8:5060"); append_branch(); if ( !t_relay() ) { xlog ("L_INFO","route22: t_relay returned error"); } xlog ("L_INFO","route22: End of routine"); return; } Doug

1 0

[Serusers] simple click-to-call
by Ken Rozinsky 14 Dec '05

14 Dec '05

Hello, As an exercise in learning how all this stuff works, I'm trying to set up a simple click-to-call service for our office to use. And potentialy to allow clients to use vai the web site to save them on the long distance charges. I set up Asterisk with the TACI script and got it to all work with an account on gafachi but I noticed that it could only handle one call at a time. I was told that openSer would allow more than one call at a time so I've installed it (basic no mySql) and have it up and running. Now, being new to this I have no idea how to connct to gafachi and do a click-to-call type thing. Would anybody be able to help me with this? Regards

2 2

[Serusers] Changing Sip Port
by sagar 13 Dec '05

13 Dec '05

Hi All, I am using ser 0.8.14 +asterisk B2bua. Currently softphone is registering on standard sip port 5060 . Is it possibe to use any other port like 5090 or 8069 I tried to do so by changing the listining port to 5090 in ser.cfg but in case the softphone is not getting register. Here is the ser log 0(11348) SIP Request: 0(11348) method: <REGISTER> 0(11348) uri: <sip:192.168.1.100> 0(11348) version: <SIP/2.0> 0(11348) parse_headers: flags=1 0(11348) end of header reached, state=5 0(11348) parse_headers: Via found, flags=1 0(11348) parse_headers: this is the first via 0(11348) After parse_msg... 0(11348) preparing to run routing scripts... 0(11348) DEBUG : is_maxfwd_present: searching for max_forwards header 0(11348) parse_headers: flags=128 0(11348) DEBUG: is_maxfwd_present: value = 70 0(11348) parse_headers: flags=8 0(11348) DEBUG: add_param: tag=fa37b062306344d68023aed26289e820 0(11348) DEBUG: add_param: epid=f657995ecc 0(11348) end of header reached, state=29 0(11348) parse_headers: flags=256 0(11348) end of header reached, state=9 0(11348) DEBUG: get_hdr_field: <To> [37]; uri=[sip:2851816416@192.168.1.100:5090] 0(11348) DEBUG: to body [<sip:2851816416@192.168.1.100:5090> ] 0(11348) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 0(11348) DEBUG: get_hdr_body : content_length=0 0(11348) found end of header 0(11348) find_first_route(): No Route headers found 0(11348) loose_route(): There is no Route HF 0(11348) parse_headers: flags=64 0(11348) parse_headers: flags=64 0(11348) parse_headers: flags=33554432 0(11348) check_self - checking if host==us: 13==13 && [192.168.1.100] == [192.168.1.100] 0(11348) check_self - checking if port 5090 matches port 5060 0(11348) check_self: host != me 0(11348) DEBUG: t_addifnew: msg id=1 , global msg id=0 , T on entrance=0xffffffff 0(11348) parse_headers: flags=-1 0(11348) parse_headers: flags=60 0(11348) t_lookup_request: start searching: hash=58712, isACK=0 0(11348) DEBUG: proceeding to pre-RFC3261 transaction matching 0(11348) DEBUG: t_lookup_request: no transaction found 0(11348) DEBUG: mk_proxy: doing DNS lookup... 0(11348) check_via_address(192.168.1.2, 192.168.1.2, 0) 0(11348) DEBUG: add_to_tail_of_timer[4]: 0xb6165fe8 0(11348) DEBUG: add_to_tail_of_timer[0]: 0xb6165ffc 0(11348) SER: new transaction fwd'ed 0(11348) DEBUG:destroy_avp_list: destroing list (nil) 0(11348) receive_msg: cleaning up Please advise. Thanks In Advance. --------------------------------- Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Yahoo! Personals

4 3

[Serusers] Segmentation Fault (SER Not Starting)
by sagar 13 Dec '05

13 Dec '05

Hi All, I am having trouble when trying to start SER. I am using Ser 0.8.14 on Redhat ES 3. When I am trying to start it gives Segmentation Fault. Can anybody suggess how to overcome this problem. Thanks And Regards Vidhya Sagar Dixit --------------------------------- Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping

4 3

[Serusers] Getting Nokia 6680 PTT to work
by Pieter De Wit 13 Dec '05

13 Dec '05

Hello Guys, Please can someone give me some pointers as to how I can get this to work - I have looked at the guide and it's a bit big for a "browse" - Isn't there a mini-how-to ? I need the server to access reg's etc. Thanks, Pieter �This e-mail is sent on the Terms and Conditions that can be accessed by Clicking on this link http://www.vodacom.net/legal/email.aspx "

1 0

[Serusers] SUBSCRIBE timeout again
by Andrey Kouprianov 13 Dec '05

13 Dec '05

Hi, I am having the same problem with SUBSCRIBEs again.... SER just keeps on sending me back timeouts for them... The problem persists only for any other presentity except myself. SUBSCRIBE timeouts happen most of the times, however, it's only sometimes, that they get through. Here's a part of ser.cfg responsible for SUBSCRIBE and NOTIFY: . . . . . . if(method=="SUBSCRIBE") { if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { log(1, "Authentication for SUBSCRIBE"); www_challenge("sip.interlab.ait.ac.th", "0"); break; }; if(!lookup("location") ) { log(1, "No registered user found for SUBSCRIBE request"); sl_send_reply("404", "Not Found"); break; }; }; if(method=="NOTIFY") { if (!www_authorize("sip.interlab.ait.ac.th", "subscriber")) { log(1, "Authentication for SUBSCRIBE"); www_challenge("sip.interlab.ait.ac.th", "0"); break; }; if (!lookup("location")) { log(1, "No registered user found for NOTIFY request"); sl_send_reply("404", "Not Found"); break; }; }; if( !t_relay() ) { sl_reply_error(); }; . . . . . . . . Here are the traces: Message 1 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 1 SUBSCRIBE From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Content-Length: 0 Message 2 from 203.159.31.36:5060 to 203.159.32.39:5060 SIP/2.0 401 Unauthorized Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 1 SUBSCRIBE From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 To: <sip:12345@sip.interlab.ait.ac.th>;tag=1a5a85ff3e08e000f4c2527642cf14ac.5e75 Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK3ad443c5b49de4b050bd2b028597a868 WWW-Authenticate: Digest realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0" Server: Sip EXpress router (0.9.3 (i386/freebsd)) Warning: 392 203.159.31.36:5060 "Noisy feedback tells: pid=17754 req_src_ip=203.159.32.39 req_src_port=5060 in_uri=sip:12345@sip.interlab.ait.ac.th out_uri=sip:12345@sip.interlab.ait.ac.th via_cnt==1" Content-Length: 0 Message 3 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 Message 4 from 203.159.32.39:5060 to 203.159.31.36:5060 SUBSCRIBE sip:12345@sip.interlab.ait.ac.th SIP/2.0 Call-ID: 7762edc6f3954c87c8b95a0270653250(a)203.159.32.39 CSeq: 2 SUBSCRIBE To: <sip:12345@sip.interlab.ait.ac.th> Via: SIP/2.0/UDP 203.159.32.39:5060;branch=z9hG4bK991c906a283ca5d57e8856c98097e81c Max-Forwards: 70 Allow: REGISTER,INVITE,BYE,ACK,CANCEL,SUBSCRIBE,NOTIFY,MESSAGE User-Agent: IntERLab User Agent 1.0a Event: presence Expires: 600 Accept: application/pidf+xml,application/xpidf+xml,application/xvcresource+xml Contact: <sip:andrey@203.159.32.39:5060> Authorization: Digest username="andrey",realm="sip.interlab.ait.ac.th",nonce="439ee222efefefbb8cbb78e75d6d15b723a1e1d0",uri="sip:12345@sip.interlab.ait.ac.th",response="1c7fa7ceb1f654d0e2ea91c54823bb8d" From: "Andrey Kuprianov" <sip:andrey@sip.interlab.ait.ac.th>;tag=15493888 Content-Length: 0 . . . . . . And so it goes, until SER replies with Timeout. There are no error messages in the log. Can you tell me what is wrong???

1 0

[Serusers] include
by FaberK 13 Dec '05

13 Dec '05

Hi folks, I've a question. Is it possible to use "include" files into ser.cfg? I would like to keep the ser,cfg locked and use the include to make changes into routing. Any ideas? Thanks -- .:FaberK:.

2 1

RE: [Users] Can't create table './openser/pending.frm'
by Maria Yndefors 13 Dec '05

13 Dec '05

Yes, It seems the error was caused by too low number defined in MaxNoOfOrderedIndexes in the NDB cluster configuration. Thanks for the help. /Maria > -----Original Message----- > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org]On > Behalf Of Bogdan-Andrei Iancu > Sent: den 13 december 2005 12:25 > To: Maria Yndefors > Cc: users(a)openser.org > Subject: Re: [Users] Can't create table './openser/pending.frm' > > > Hi Maria , > > I would say it;s more a problem related to mysql setup - try > to google > to find the meaning of errno 4242 > > regards, > bogdan > > Maria Yndefors wrote: > > >Hallo, > >I have tried to install openser and run into this problem > when doing openser_mysql.sh create > > > >creating database openser ... > >ERROR 1005 (HY000) at line 215: Can't create table > './openser/pending.frm' (errno: 4242) > > > >I run Mysql server on another server and have set the > variables in the .opensermyslqrc file > > > >sbc2:/etc/openser# cat .opensermysqlrc > ># File .opensermysqlrc included by openser_mysql.sh > ># Set static variables to be used by the script > > > >export SQL_HOST=195.54.xxx.xxx > >export DBHOST=195.54.xxx.xxx > >export TABLE_TYPE="ENGINE=NDB" > >export SIP_DOMAIN=BREDBAND > > > >When I do show tables in mysql I can see: > > > >mysql> show tables; > >+-------------------+ > >| Tables_in_openser | > >+-------------------+ > >| acc | > >| active_sessions | > >| aliases | > >| dbaliases | > >| grp | > >| location | > >| missed_calls | > >| version | > >+-------------------+ > >8 rows in set (0.02 sec) > > > >some of the tables have been created. > >Why could not the pending table be created? > > > >Regards > >/Maria > > > > > > > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[Serusers] Problems building RTPproxy on FC4
by sip 13 Dec '05

13 Dec '05

I'm having an issue building RTPproxy on FC4. I checked out the source yesterday, and the configure script went well, but the build is failing on some weird things. make all-am make[1]: Entering directory `/home/eneref/new/rtpproxy' gcc -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -c main.c main.c: In function 'handle_command': main.c:363: warning: pointer targets in passing argument 6 of 'recvfrom' differ in signedness main.c: In function 'main': main.c:997: error: 'useconds_t' undeclared (first use in this function) main.c:997: error: (Each undeclared identifier is reported only once main.c:997: error: for each function it appears in.) main.c:997: error: syntax error before 'delay' main.c:1230: error: 'delay' undeclared (first use in this function) make[1]: *** [main.o] Error 1 make[1]: Leaving directory `/home/eneref/new/rtpproxy' make: *** [all] Error 2 First off, useconds_t is in sys/types.h which is one of the includes in the main. But looking at the GCC line, it looks almost as though something's missing. I checked the Makefile, and the DEFAULT_INCLUDES is set to -I. -I$(srcdir) -I. and then there's that space... where I assume $(INCLUDES) would go... although that's not set anywhere in the Makefile. Anyone like to smack me with the bat of stupidity and tell me what I'm doing wrong? N.

1 0

[Serusers] Postgres instead of Mysql
by Felipe Louback 13 Dec '05

13 Dec '05

I am trying to use Postgres instead of Mysql, but I dont know why, then I initialize ser, I get the following error: 0(0) find_mod_export: <db_use_table> in module sql not found 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) ERROR: mod_init(): Can't bind database module I cant figure out the reason for this message "<db_use_table> in module sql not found". I have no idea why this is happening. When I was working with Mysql, I didn't have this problem, but now, I just cant put Postgres to work. Any ideas? Here is part of my ser.cfg: ============================================================ loadmodule "/usr/local/lib/ser/modules/postgres.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/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc", "user_column", "username") modparam("usrloc", "db_url", "sql://ser:my_pass@localhost:5432/ser") =================================================================== Here is part of the log when I initialize SER: ============================================================= 0(0) DEBUG: init_mod: usrloc 0(0) usrloc - initializing 0(0) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(0) unixsock_register_cmd: New command (ul_stats) registered 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) find_mod_export: <db_use_table> in module sql not found 0(0) bind_dbmod: Module sql does not export db_use_table function 0(0) ERROR: mod_init(): Can't bind database module 0(0) init_mod(): Error while initializing module usrloc ERROR: error while initializing modules 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(27948) shm_mem_destroy 0(27948) destroying the shared memory lock ================================================================ Any clue is more than welcome... thanks. Felipe Louback -- Mestrando - Departamento de Engenharia Elétrica Grupo de Pesquisa em Engenharia da Computação e Telecomunicações Universidade Federal de Minas Gerais - Brasil "Todas as coisas cooperam para o bem daqueles que amam a Deus" Romanos 8:28

2 4

[Users] How to start ser on boot? how to monitor it?
by Cesc 13 Dec '05

13 Dec '05

Hi, An easy one ... or two. 1 - I need to start up ser automatically on the boot of the machine ... i am sure there are scripts and instructions on how to do it. Any pointer? 2 - I need a way to monitor the status of ser (and sems), so that in case of a crash they are automatically killed and restarted. I thought about writing some kind of cron thingy using serctl ... Again, pointers? Regards, Cesc

6 8

[Users] OpenSER 1.1
by Douglas Garstang 13 Dec '05

13 Dec '05

I see documentation available for OpenSER 1.1 at www.openser.org, but I can't seem to find anywhere where it's available for download. It is available yet? Thanks

3 2

[Users] Can't create table './openser/pending.frm'
by Maria Yndefors 13 Dec '05

13 Dec '05

Hallo, I have tried to install openser and run into this problem when doing openser_mysql.sh create creating database openser ... ERROR 1005 (HY000) at line 215: Can't create table './openser/pending.frm' (errno: 4242) I run Mysql server on another server and have set the variables in the .opensermyslqrc file sbc2:/etc/openser# cat .opensermysqlrc # File .opensermysqlrc included by openser_mysql.sh # Set static variables to be used by the script export SQL_HOST=195.54.xxx.xxx export DBHOST=195.54.xxx.xxx export TABLE_TYPE="ENGINE=NDB" export SIP_DOMAIN=BREDBAND When I do show tables in mysql I can see: mysql> show tables; +-------------------+ | Tables_in_openser | +-------------------+ | acc | | active_sessions | | aliases | | dbaliases | | grp | | location | | missed_calls | | version | +-------------------+ 8 rows in set (0.02 sec) some of the tables have been created. Why could not the pending table be created? Regards /Maria

2 1

[Serusers] Problems with STOP accounting
by Vesna Levkova Damjanovski On.net 13 Dec '05

13 Dec '05

We have a problem registering the stop accounting record in our database. We have freeradius server. When bye method occurs radius don't receive stop accounting record so we don't get the end time of call. The strange thing is that in some cases we received in some not.

1 0

[Serusers] Where does SER get the public IP of a NATed uac from for sending as STATUS message?
by Frank Fischer 13 Dec '05

13 Dec '05

Hi all related to 0.9.1pre. I have a problem with NATed UACs in my setup. Therefor i would like to know where SER takes the public IP of a NATed UAC from, a request is forwarded to (using the reply handler) if i call fix_nated_contact() to rewrite the NATed UACs contact header. Is it always taken form the received field in the location record? Just for explanation: It seems that my SER does not rewrite the contact header field in the reply route when i call fix_nated_contact(). The contact header value stays the same, just before and after call to fix_nated_contact(). So i think, it could be SER does not get the correct "NAT"-information for the related enpoint and the reads a value from a request field? Anyone ever encountered a similar situation? Is there a way to get closer information on what's happening when calling fix:nated_contact() during runtime? Thanks for you help. Kind regards Frank

4 4

[Serusers] post
by Vesna Levkova Damjanovski On.net 13 Dec '05

13 Dec '05

1 0

[Serusers] SER presence snapshot
by Vaclav Kubart 13 Dec '05

13 Dec '05

Hi all, there is new presence snapshot available at (simple-compilable sources with demo config): ftp://ftp.iptel.org/pub/ser/presence/ [it is now second version: ser-0.10.99-dev25-timers-pa-2_src.tar.gz] Actual version of "presence handbook" is there too: ftp://ftp.iptel.org/pub/ser/presence/presence_book.xhtml Vaclav

1 0

[Serusers] SER, presentity and mySQL
by Ángel Gómez 13 Dec '05

13 Dec '05

Hi, I'm working wich SER and making it to suport presence. Now I have SER instaled with the mySQL and pa modules loaded, the thing is than I make the pa module to use db to store the presence data, but when I send a Notify to the server its don't store any data in the presentity_contact table, and I want to know why it dosen't store anything in there. If some one knows why isn't stored any data, of if I make the notify bad, or if I doing this in the wrong way... please tell me, my objetive is to store the presentity status in teh data base. here is the notify than I use: NOTIFY sip:10.22.1.5:5060 SIP/2.0 To: <sip:angel@10.22.1.249> From: <sip:angel@10.22.1.249>;tag=edf6b60e8b292eeca4ee51d20e01ffa7-e36a CSeq: 1 NOTIFY Call-ID: af2756608d177822 Content-Length: 244 User-Agent: Sip EXpress router(0.9.3 (i386/linux)) Event: presence Content-Type: application/pidf+xml Subscription-State: active;expires=594 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "//IETF//DTD RFCxxxx PIDF 1.0//EN" "pidf.dtd"> <presence entity="sip:ninguna@10.22.1.249"> <tuple id="9r28r49">ser"> <status> <basic>closed</basic> </status> </tuple> </presence>

2 1

[Serusers] How can you check if From: SIP URI is Registered.
by Leo Papadopoulos 12 Dec '05

12 Dec '05

Dear SerUsers: I am running SER 0.9.4 and have just finished reading all the README's of all the modules, but cannot find what I am trying to do. I want to check all incoming messages to see if the SIP URI in the "From:" field is presently registered. I am using MySQL for registration and authentication. If the SIP URI is the "From:" field is NOT registerd I want to return a "403 Forbidden" message. I know that From: fields can be spoofed, but I want to do this anyway. Does anyone know of an existing module that can help with this? Or of anyway to do this? Leo P.

3 2

[Serusers] ERROR: cannot forward to af 2, proto 2 no corresponding listening socket
by Matt morris 12 Dec '05

12 Dec '05

Hello List, I got this error message when I try to do a forward_tcp: forward_req: ERROR: cannot forward to af 2, proto 2 no corresponding listening socket Do I need to configure SER to use tcp sockets at the top, or is it something else? Thank you all in advance. _________________________________________________________________ Take advantage of powerful junk e-mail filters built on patented Microsoft® SmartScreen Technology. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

1 0

[Serusers] SER don't start
by Guido 12 Dec '05

12 Dec '05

Hello, I'm using Fedora C4, ser-0.9.4 with radiusclient-ng-0.5.2. When I start SER I view this error message and SER don't start 0(14959) WARNING: fix_socket_list: could not rev. resolve 66.167.141.22 0(14959) WARNING: fix_socket_list: could not rev. resolve 66.167.141.22 Listening on udp: 66.165.171.22 [66.167.141.22]:5060 tcp: 66.165.171.22 [66.167.141.22]:5060 Aliases: WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) info: mediaproxy: loaded SIP asymmetric clients file containing 0 entries. 0(0) info: mediaproxy: loaded RTP asymmetric clients file containing 0 entries. 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 14961 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(14959) new_connection(): Access denied for user: 'serro@localhost' (Using password: YES) 0(14959) db_init(): Could not create a connection 0(14959) ERROR:domain_db_init: cannot initialize database connection 0(14959) ERROR: domain:child_init(): Unable to connect to the database 0(14959) init_mod_child(): Error while initializing module domain 0(14959) main_dontfork: init_child failed 1(14960) INFO: signal 15 received 2(14961) 0(14959) child process 14960 exited normally, status=0 0(14959) INFO: dont_fork turned on, living on INFO: signal 15 received 0(14959) child process 14961 exited normally, status=0 0(14959) INFO: dont_fork turned on, living on Can anybody help me ? Regards, Guido

2 1

[Serusers] problems to install ser with presence
by victoria beltran 12 Dec '05

12 Dec '05

Hi,I am trying to install the last unstable SER version with support for SIP-based presence. When I compile and install SER with presence modules (make install include_modules="pa rls mysql), it does this error: action.c: no such onsend.h route.c: no such onsend.h forward.c: no such onsend.h I have installed all libraries required (libxml2 y libcurl3) and the libxml2-dev and libcurl3-dev devel packages. The file "onsend.h" does not be in /sip_router directory. Could anybody help me? Thanks!! ______________________________________________ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es

3 4

[Serusers] ser_fifo and access denied issue (Maybee Context Problem) can you help?
by Douglas Mahy 12 Dec '05

12 Dec '05

I have had considerable success in getting ser working under selinux particularly as I am new to linux having just recently moved over from working in a windows environment. However I have had a lot of problems trying to resolve a permissions problem that is resulting in serweb kicking out the error. FIFO not running or bad path to it I have tried all the usual stuff such as including the lines I have also turned off selinux. fifo="/tmp/ser_fifo" fifo_mode=0777 or sock_mode=0666 I now believe that this may be something to do with context, is there anybody that has experienced this problem. Below is the error I am getting using dmseg, it increments each time I refresh the admin page. audit(1134382735.156:78): avc: denied { getattr } for pid=2682 comm="httpd" name="ser_fifo" dev=sda2 ino=1887848 scontext=root:system_r:httpd_t tcontext=root:object_r:tmp_t tclass=fifo_file Below are the permissions on the ser_fifo and its context. /dev/ brw-rw---- root disk system_u:object_r:fixed_disk_device_t sda1 brw-rw-rw- root disk system_u:object_r:fixed_disk_device_t sda2 brw-rw---- root disk system_u:object_r:fixed_disk_device_t sda3 /tmp/ -rw------- root root root:object_r:tmp_t php-install.GQ3113 prwxrwxrwx root root root:object_r:tmp_t ser_fifo prw-rw-rw- root root root:object_r:tmp_t ser_receiver_3011 Any help would be very much appreciated. Doug Mahy Telewest Broadband Head of Research & Development Network Engineering and Strategy M +44(0) 7985 807 534 Email: <mailto:douglas.mahy@telewest.co.uk> douglas.mahy(a)telewest.co.uk ------------------------------------------------------------------------------ Altogether Better www.telewest.co.uk The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. ==============================================================================

1 0

[Users] location table refresh
by Matteo Piazza 12 Dec '05

12 Dec '05

Hi all, I have a question. When I change the location tables is it necessary to restart openser? I see that if I don`t restart the openser the program don`t see the modifications on the location table. Is it possible to refresh without restart the program, which command can I use? Matteo -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 1

[Serusers] Problem configuring presence module
by Javi Lavandeira 12 Dec '05

12 Dec '05

Hello, I work for a Japanese company and I'm writing in behalf of a coworker who can read some English, but cannot write it. He tells me that he's trying to use authentication in SER's presence module, but gets the following error when trying to run it: Dec 2 16:55:22 gorilla3 ser: set_mod_param_regex: parameter <auth> not found in module <pa> Dec 2 16:55:22 gorilla3 ser: parse error (93,20-21): Can't set module parameter He thinks that the problem might be the following line in ser.cfg: modparam("pa", "auth", "xcap") The version he's using is ser-0.10.99 from CVS. Any idea on what could be the problem? Thanks in advance, -- Javi Lavandeira Technical manager International Systems Research http://www.isr.co.jp

2 1

Re: [Serusers] SER Presence
by Vaclav Kubart 12 Dec '05

12 Dec '05

These messages in log mean, that the SUBSCRIBE request didn't contain any known document types in Accept headers. There was a bug in PA module - it processed only the first Accept header, it was corrected on Friday, thus with actual CVS version it may work; but only in the case, that your client sends more Accept headers and one of them is known to PA module. If it will NOT work with actual version, please send me the message flow. thanks, Vaclav

1 0

[Serusers] problems to install SER with presence
by victoria beltran 12 Dec '05

12 Dec '05

Hi, I am trying to install the last unstable SER version with support for SIP-based presence. When I compile and install SER with presence modules (make install include_modules="pa rls mysql), it does several errors: receive.o: In the function 'receive_msg': /root/sip_router/receive.c:231:undefined reference to 'reset_user_avps' receive.o:/root/sip_router/receive.c:232:undefined reference to 'reset_user_avps' receive.o:/root/sip_router/receive.c:251:undefined reference to 'reset_user_avps' receive.o:/root/sip_router/receive.c:252:undefined reference to 'reset_domain_avps' I have downloaded current ser sources by "cvs -d :pserver:anonymous@cvs.berlios.de:/cvsroot/ser checkout sip_router" and I have seen the usr_avp.h and user_avp.c files. In this files, the reset_user_avps and reset_domain_avps functions hasnt been defined. Could anybody help me? Thanks!! --------------------------------- Correo Yahoo! Comprueba qué es nuevo, aquí http://correo.yahoo.es

3 2

[Users] Routing
by Douglas Garstang 12 Dec '05

12 Dec '05

We have a configuration where OpenSER sit's between Polycom phones and Asterisk. Off-net/PSTN calls are also sent from Asterisk to OpenSER to be routed to a Mediacodes Gateway. I'm having a hell of a time with the routing in general! What is the best way to route calls? Should I be looking at the URI? The From:? The To:? The method? Also, the convention seems to be that the URI contains where the message is going TO rather than where it came FROM. However, routing is much easier if the URI has where it came from. For example, messages from the phones have a URI of: uac.voip.com and messages from Asterisk have the IP adress in the URI, as Asterisk doesn't seem to let you set the URI. Calls from the Mediacodes have a URI of: pstn.voip.com Is adding sub-domains like 'uac' and 'pstn' a good way to route? I've looked at all the examples I could find but unfortunately they're ridiculously simple. Every time I look at openser.cfg I get VERY STRESSED OUT! Thanks.

2 1

[Users] 302 and contact address new invites an request header..
by Helge Waastad 12 Dec '05

12 Dec '05

Hi, I was just wondering if there has been any changes to the pseudo variables? I receive a 302 with a new contact address, however if I print out in the failure router $ct I get the $from..... br hw PS, another thing I've realized now, is that I have a problem solving the new INVITE after a redirect. Since the new INVITE after redirect has the same call-id, does that mean that all flags etc is kept and still are valid? For instance; INVITE: xxx@<cisco-gw> -> yyy@<domain> REDIRECT: <- 302 Contact: zzz@<domain> NEW INVITE xxx@<cisco-gw> -> zzz@<domain> I have a : if (uri=~"^sip:yyy@.*") in the beginning of my script and it return true on the new INVITE.. Is this right? br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 3

Fwd: [Serusers] Adding users in the loaded db
by samuel 12 Dec '05

12 Dec '05

Forgot to send it to the list.....I'm still on holiday mood..... Sam. ---------- Forwarded message ---------- From: samuel <samu60(a)gmail.com> Date: 12/12/2005 09:59 Subject: Re: [Serusers] Adding users in the loaded db To: Xavier TRENTIN <xtrentin(a)softiphone.net> Hi all!!! If you use serctl, it will do both load the user in the database and in the cache automatically. what serctl does: Update the database: standard [mysql] query. Update SER's cache: it makes use of the exported FIFO command ul_add: serctl fifo ul_add <table> <aor> <contact> <expires> <q> <replicat> <flags> example: serctl fifo ul_add location sam(a)example.com sam(a)1.2.3.4 3600 1 0 0 about the dbtext...let my guess...I've been out for a while and things may have changed but I think serctl in the SER CVS does not handle dbtext...there was a post from Cesc with some patche adding dbtext support to serctl and I think that in the openser repository there's some dbtext serctl.... Hope it helps, Sam. 2005/12/11, Xavier TRENTIN <xtrentin(a)softiphone.net>: > I agree with you except that if SER is running and I add a user in my > dbtext subscriber table, > he can't register if I don't restart SER. > > sip a écrit : > > >*********************** > >Pas de virus detecté dans l'attachement no filename > > > >Votre Mail a été vérifié par InterScan. > >***********-*********** > > > > > >SER doesn't really load users on startup into memory except from the location > >table (i.e. user/contact information). When you create a user in the > >subscriber table, that user is added. Then, when his client sends a register, > >he gets added to the location table in memory/in the db. There's nothing you > >really need to to to manually add someone to the location table... and really, > >there's nothing you SHOULD be doing. Users add themselves on login with all > >the correct information about their contact info. > > > >N. > > > > > >On Fri, 09 Dec 2005 18:58:56 +0100, Xavier TRENTIN wrote > > > > > >>Hi, > >> > >>I'd like to know if there is a way to add users in the database > >>loaded in memory, using a fifo command for example? > >> > >>If, yes, could you give an exemple. > >> > >>Cause I don't want to restart ser anytime I add a user ;p > >> > >>B.R. > >>Xavier. > >> > >>_______________________________________________ > >>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 >

1 0

[Serusers] SER Presence
by Tina Suneli 12 Dec '05

12 Dec '05

Hi, I have just joined that list and want to know why the following error is occuring when i used to subscribe with SER with an OPAL 2.0 client. 0(2876) handle_subscription() entered 0(2876) check_message(): Accepts application/simple-message-summary not valid for event package et=message-summary 0(2876) handle_subscription(): Error while checking message 0(2876) handle_subscription about to send_reply and return -2 Tina

1 0

[Serusers] SER and REGISTRATIONs UAC
by Federico Giannici 11 Dec '05

11 Dec '05

We use SER as a proxy to a PSTN gateway that is "transparent" to the final users: the users register on the proxy and it authenticate to the gateway (by means of the UAC module). Currently we use Asterisk for the registration on the PSTN gateway, but we'd like to get rid of it. What is the best way to use SER to register on the gateway? We have to write a little program that sends REGISTER messages to the gateway, or somebody already developed something (maybe a module)? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 2

[Serusers] Adding users in the loaded db
by Xavier TRENTIN 11 Dec '05

11 Dec '05

Hi, I'd like to know if there is a way to add users in the database loaded in memory, using a fifo command for example? If, yes, could you give an exemple. Cause I don't want to restart ser anytime I add a user ;p B.R. Xavier.

2 3

[Serusers] http tunneling
by Imran Akbar 10 Dec '05

10 Dec '05

Hi, I've tried to use a softphone from behind a corporate firewall which only allows http requests through its proxy - and it doesn't work, even with STUN. I've heard that tunneling through HTTP would allow VoIP to work in this situation - does anyone know how to set this up, with SER or other hardware/software? Thanks, Imran

3 2

[Serusers] SER + SEMS & Music on Hold
by Waldo Rubinstein 09 Dec '05

09 Dec '05

Does anyone have any configuration sample on setting up SER + SEMS in order to "put a call on hold and listen to MoH" while SER reaches the destination peer(s)? Or at least point to some documentation referencing this? Thanks, Waldo

1 0

[Serusers] problems with ser + radius
by Guido 09 Dec '05

09 Dec '05

Hello, I'm using fedora core 4 and radiusclient-0.4.3. installing SER 0.9.4 with support for radius I can't install radius modules and when run my SER I view this mesasge error 0(441) ERROR: load_module: could not open module </usr/local/lib/ser/modules/auth_radius.so>: /usr/local/lib/ser/modules/auth_radius.so: cannot open shared object file: No such file or directory 0(441) parse error (41,13-14): failed to load module 0(441) set_mod_param_regex: No module matching auth_radius found | 0(441) parse error (71,20-21): Can't set module parameter 0(441) set_mod_param_regex: No module matching auth_radius found | 0(441) parse error (72,20-21): Can't set module parameter 0(441) parse error (326,34-35): unknown command, missing loadmodule? Anybody can help me ? Thanks Guido

2 2

[Serusers] announcement: student project finished: partiSIPation
by Juha Heinanen 09 Dec '05

09 Dec '05

matthias, i looked at partiSIPpation documentation and didn't quite understand why microphone and speaker volumes are part of the api. i would think that those are internal matters of the gui component. when gui places a call, i would imagine it gives to core an ip address and port that it wants to use for the media plus a list of codecs it supports. i don't understand what volume issue have to do with communication between gui and core. -- juha

2 3

[Users] multi-leg accounting
by Helge Waastad 09 Dec '05

09 Dec '05

Hi, to get multi-leg accounting to work, is it necessary to avp_write "something" to the src_leg_avp_id/dst_leg_avp_id avp's? Right now I only get "n/a" in those columns. I've tried to use avp_write without any success. br, hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

3 5

[Users] RE: [Devel] unixODBC Database Connection Problem
by Lenir 09 Dec '05

09 Dec '05

Ok, I tried using the FreeTDS driver for unixodbc and I got a more detailed error message that might shine a light: Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect IM007:1:0:[unixODBC][FreeTDS][SQL Server]Could not find UID parameter HYC00:2:0:[unixODBC][FreeTDS][SQL Server]Driver not capable 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration These are my entries in /etc/odbc.ini [SYBASE] Driver = /usr/local/lib/libtdsodbc.so Description = Sybase Server Trace = Yes Trafcefile = /tmp/odbc.log #Servername = JDBC Server = xx.xx.xx.xx Port = 5000 DSN = database Database = database #UserID = username # I tried using both UID and UserID, had same results UID = username Password = passwd TDS_Version = 5 These are my entries in /etc/freetds.conf: [SYBASE] host = xx.xx.xx.xx port = 5000 tds version = 5 For whatever reason the module is not picking up the UID and PWD from the db_url parameter i.e. modparam("usrloc|auth_db", "db_url", "unixodbc://username:passwd@xx.xx.xx.xx/SYBASE") The way I fixed the problem was by changing the file my_con.c, line 69: From: sprintf( stringDNS, "%s%s%s", "DSN=", id->database, ";"); to: sprintf( stringDNS, "%s%s%s", "DSN=", id->database, ";UID=username;PWD=passwd;"); And now works fine. Also, to add debug to figure out which SQL queries fail, I changed the file dbase.c line 75: From: LOG(L_ERR, "Return value: %d\n", ret); To: LOG(L_ERR, "Return value: %d, Performing query: %s\n", ret, _s); Now everytime a query fails it'll say the result msg, as well as the actual query that failed. Lenir _____ From: devel-bounces(a)openser.org [mailto:devel-bounces@openser.org] On Behalf Of lenirsantiago(a)yahoo.com Sent: Thursday, December 08, 2005 12:59 PM To: devel(a)openser.org; users(a)openser.org Subject: [Devel] unixODBC Database Connection Problem Guys, Im trying to use unixodbc.so to connect to a Sybase database using Sybase native ODBC drivers. This is the entry that I have on: /etc/odbc.ini: ============= [SYBASE] Description = Sybase ODBC Data Source UserID = username Password = passwd Driver = ASE Server = xx.xx.xx.xx Port = 5000 Database = database UseCursor = 1 /etc/odbcinst.ini ============ [ASE] Description = Sybase ODBC Driver Driver = /opt/sybase/ODBC-12_5/lib/libsybdrvodb.so FileUsage = -1 UsageCount = 1 The database is on a remote host. I can connect from the openser server using unixodbc's isql tool, as such: [root@ser openser]# isql SYBASE username passwd +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> This proves that unixodbc is installed correctly and that the Sybase odbc driver is installed and functioning properly. However when I try to run openser (debug=4) I get the following: 0(0) fixing /usr/local/lib/openser/modules/auth_db.so www_authorize 0(0) db_init: Connection 'unixodbc://username:passwd@xx.xx.xx.xx/SYBASE' not found in pool Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration 0(0) XLOG: destroy module ... 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(22871) shm_mem_destroy 0(22871) destroying the shared memory lock [root@ser openser]# This is a snippet of my openser.cfg ------------------------------START-------------------------------- debug=4 fork=no log_stderror=yes listen=yy.yy.yy.yy port=5060 children=4 server_signature=no dns=no rev_dns=no check_via=no syn_branch=yes memlog=1000 sip_warning=no fifo_mode=0666 fifo="/tmp/openser_fifo" fifo_db_url="unixodbc://username:passwd@xx.xx.xx.xx/SYBASE" reply_to_via=no loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/postgres.so" loadmodule "/usr/local/lib/openser/modules/unixodbc.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_radius.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/options.so" #loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" modparam("usrloc", "db_mode", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("usrloc|auth_db", "db_url", "unixodbc://username:passwd@xx.xx.xx.xx/SYBASE") ---------------------------------END---------------------------------------- - Any suggestions? Lenir

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[Users] Relaying and failure route - misrouted ACK
by Stefan Prelle 09 Dec '05

09 Dec '05

Hi folks, I need an advice regarding failure routes. I want the (Open)SER to route all incoming calls to a PSTN gateway. If that route fails (e.g. because the gateway is down) it shall route the calls to another gateway. I managed to archieve that in the failure case the second gateway receives the call. But the ACK request that the caller sends when receiving the 200 OK is delivered to the primary target again, not to the fallback gateway. My configuration (excerpt): route { ... record_route(); if (loose_route()) { t_relay(); exit; }; t_on_failure("1"); t_relay_udp("192.168.0.37","5060"); } failure_route[1] { append_branch(); t_relay_to_udp("192.168.0.39", "5060"); } Thanks in advance. Regards, Stefan

2 6

[Serusers] Radius - LDAP
by Paco Orozco 09 Dec '05

09 Dec '05

Hiya, I'm trying to use RADIUS server to authorize (authenticate) SIP users. I'm using auth_radius module. My radius server is the same machine, FreeRadius wich is configured to use LDAP as DB. Whe I try to authenticate a SIP user I've got a problem, SIP SER sends a radius request with digest method, but my Radius needs a User-Password suplied in order to bind in LDAP. How can intruct SIP SER in order to pass auth password as User-Password althout Digest??? Thanks in advance. -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida.

2 2

[Users] New Feature: Supported methods and lookup
by Bogdan-Andrei Iancu 09 Dec '05

09 Dec '05

Hi, based on Juha's patch - saving in usrloc the supported methods advertised during REGISTER - I extended lookup() function to perform method based filtering. What this means: lookup() may return only contacts which supports the method of the processed request. Ex: there are two clients registering as same user: 1) a SIP phone which support INVITE & family, but no presence and messaging. 2) a soft messaging client : supports only MESSAGE When a call hits the server, the lookup() will send it only to contact (1), to the phone, since the contact (2) does not support INVITE. When a MESSAGE hits the server, the lookup() will send it only to contact (2), to the soft client, since the contact (1) does not support MESSAGE. the advantage is that you avoid un-necessary parallel forking and extra signalling. Also there are SIP implementation which does not even reply to methods they are not supporting, forcing the proxy to generate timeout..:( The feature may be turned on or off via the module parameter "method_filtering": http://www.openser.org/docs/modules/1.1.x/registrar.html#AEN247 For more, see also the script snapshot from : http://www.openser.org/docs/modules/1.1.x/registrar.html#AEN288 regards, bogdan

2 4

RE: [Serusers] What does SER/FIFO error usually mean?
by Yong, Chee Yeew 09 Dec '05

09 Dec '05

That worked. Thanks. I am not sure why this (fifo_mode=0777) wasn't inserted in either the ser.cfg or ser.cfg.sample files. Chee -----Original Message----- From: Steve Blair [mailto:blairs@isc.upenn.edu] Sent: 08 December 2005 18:39 To: Yong, Chee Yeew Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] What does SER/FIFO error usually mean? Permissions problem I'm guessing. Do you have the following in your config file? fifo="/tmp/ser_fifo" # named pipe needed by serctl fifo_mode = 0777 Yong, Chee Yeew wrote: > Hello all, > > I am trying to add a user to my MySql database with the command: > > serctl add joeuser passwd joeuser(a)here.com <mailto:joeuser@here.com> > > but I am getting an error: > error: SER/FIFO not accessible: 2 > > Any pointers to where to start debuging this error? > > Thanks. > > chee > >------------------------------------------------------------------------ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

2 1

Re: [Serusers] Set up aliases
by Anthony Chapellier 09 Dec '05

09 Dec '05

It doesn't work. It seems 'mhomed = 1' binds ser to localhost. So when I try to forward the request to my SIP server after I did the local forward to save the user alias, it says 'can't access to the net'. Anthony Michal Matyska a écrit : >Try to add following line into your config file: >mhomed=1 > >Michal > >On Fri, Dec 09, 2005 at 01:46:26PM +0100, Anthony Chapellier wrote: > > >>I'm using SER 0.9.4. >> >>Michal Matyska a écrit : >> >> >> >>>Which version did you use? >>>I'm getting the same error... looking into the code, but nothing found yet. >>> >>>Michal >>> >>>On Fri, Dec 09, 2005 at 12:19:20PM +0100, Anthony Chapellier wrote: >>> >>> >>> >>> >>>>I've got a problem with forwarding in loopback. I tried to forward the >>>>request with the following code (tested both) : >>>> >>>>t_relay_to_udp("127.0.0.1", "5060"); >>>> >>>>or >>>> >>>>forward(127.0.0.1, 5060); >>>> >>>>In both of them, I got an error 477 : Can't process to the next hop - >>>>127.0.0.1:5060 Noisy Feefback... Do you know where it could come from ? >>>> >>>>Thanks, >>>> >>>>Anthony >>>> >>>>Jan Janak a écrit : >>>> >>>> >>>> >>>> >>>> >>>>>Forward the request to the same proxy again over loopback. In the 2nd >>>>>run you can detect that the request is coming from 127.0.0.1 and save it >>>>>in aliases table instead of location. >>>>> >>>>>jan. >>>>> >>>>>On 08-12-2005 11:15, Anthony Chapellier wrote: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>>Hi, >>>>>> >>>>>>When a register request is incoming, I save user location and rewrite >>>>>>contact header after that. I'd like to save this new rewriten user >>>>>>location as an alias, is there a way to set up an alias in the fly ? >>>>>> >>>>>>I saw you need to use serctl command to add aliases but can you do it >>>>>>in SER in real time ? >>>>>> >>>>>>Thanks, >>>>>> >>>>>>_______________________________________________ >>>>>>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 >>> >>> >>> >>> >>> >>> >>> >>> > > > > > >

1 0

[Serusers] Jain SIMPLE
by Andrey Kouprianov 09 Dec '05

09 Dec '05

I know this is not the proper place to ask this kind of question, but does anybody have any idea, when JAIN SIMPLE package is coming out?

2 1

[Serusers] Set up aliases
by Anthony Chapellier 09 Dec '05

09 Dec '05

Hi, When a register request is incoming, I save user location and rewrite contact header after that. I'd like to save this new rewriten user location as an alias, is there a way to set up an alias in the fly ? I saw you need to use serctl command to add aliases but can you do it in SER in real time ? Thanks,

3 4

[Serusers] create user problem with serweb
by Ashutosh kumar 09 Dec '05

09 Dec '05

Hey, We have worked up our own installation of serweb without Confirmation mail by tweaking the sql. What *ROUGHLY* you have to do is to 1. remove the "sql insert" into pending table from reg/index.php. 2. remove the "sql table copy" (select * into from...) from pending to subscribers table in reg/confirmation.php. 3. Add the equivalent of above two codes as a single insert into subscribers in the file reg/index.php. (I may be missing the file names correctly though, since I did that a long ago.) This will cause the registration records to directly go into the subscribers table without going to the pending table. You might want to change/edit the related code and messages accordingly. But remember, it is always good to follow the prescribed path. Regards, Ashutosh Kumar -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Karel Kozlik Sent: Friday, December 09, 2005 1:47 PM To: ciprian niculescu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] create user problem with serweb Hi, unfotunately current version of serweb doesn't allow to register users without sending confirmation mail. It is feature of planed next release. I hope the beta version will be avaiable shortly after New Year. You may use serctl for now. regards, Karel ciprian niculescu napsal(a): > hi, > > i have installed ser and serweb, i use it for teaching sip signaling, > and i don't want to use confirmation mail for creating user accounts, > i've looked around the docs/code but nothing pop up. > > i'm using CentOS, mysql, last ser and last serweb > > ciprian > > _______________________________________________ > 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

1 0

Re: [Users] transfer not possible!!
by Jayesh Nambiar 09 Dec '05

09 Dec '05

hi klaus, Thanx for your response. I have hard phones connected to Audio codes 4-port gateway. The gateway is configured to send the calls to openser for processing. suppose there are three users connected to it. user1 calls user2. The call goes through. Now user 2 presses flash and tries to call user3, but both the phone gets disconnected. I did an ngrep and what i noticed was this: When user2 tries to transfer the call to user 3, the "INVITE" goes from user2 to user1 again and the call gets disconnected due to request timeout. I am not able to see any REFER method in the ngrep. Any advice will be appreciated. Thanx again. Jayesh Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: Yes, REFER method is for call transfer. Which clients do you use? Does the call orginate also from a SIP phone or from a gateway? What exactly is the problem? Do you see the REFER request? Including an "ngrep port 5060" would be useful. klaus Jayesh Nambiar wrote: > hi everyone, > > I was just trying to transfer the call to another user. All users are > registered at the same openser server. Now when I receive a call, is it > not possible to transfer the call to another IP user. > I thought REFER mehtod does this, but i think I am wrong. Is this > possible, can somebody pls clear my understanding regarding this concept. > > thanks in advance!! > jayesh > > ------------------------------------------------------------------------ > Enjoy this Diwali with Y! India Click here > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users --------------------------------- Yahoo! India Matrimony: Find your partner now.

3 5

[Serusers] mysql.so
by rajesh.rajguru＠wipro.com 09 Dec '05

09 Dec '05

Hi, I have downloaded ser-0.9.4_src.tar . After compiling we found that mysql.so is not created. When I try to compile this module separately it gives error that mysql.h file is not present and when I check in modules/mysql, it is not present. How to build mysql.so module then ? Thanks in advance for the help Regards, Rajesh Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

3 3

[Users] Re: unixODBC Database Connection Problem
by Marco Lorrai 09 Dec '05

09 Dec '05

If you cannot connect to database using the command isql (provided by unixodbc) the module cannot work. You should first verify the command isql work. A problem could be a wrong connection socket, if is this restart Sysbase with the correct socket file. -- Marco Lorrai - marco.lorrai(a)abbeynet.it R&D Software Engineering Abbeynet S.p.A. - http://www.abbeynet.it VoIP (Office): +39-070-2339262 Fax: +39-070-229016

1 0

[Serusers] create user problem with serweb
by ciprian niculescu 09 Dec '05

09 Dec '05

hi, i have installed ser and serweb, i use it for teaching sip signaling, and i don't want to use confirmation mail for creating user accounts, i've looked around the docs/code but nothing pop up. i'm using CentOS, mysql, last ser and last serweb ciprian

2 1

[Serusers] Rtpproxy & Nathelper config
by Joel Solanki 09 Dec '05

09 Dec '05

Hello all, Can anybody provide me working ser.cfg with rtpproxy + nathelper. I have already installed rtpproxy. I have gone through Onsip.org download but config didnt work. Can any body provide me the config. Regards, joel

2 3

[Serusers] Development of Berkeley DB module for SER
by Jiri Kuthan 09 Dec '05

09 Dec '05

I am seeking SER programmers for development of Berkeley DB driver for SER under iptel's sponsorship. I'm inviting all folks interested in this activity to contact me privately. I will prefer applicants who send me samples of their SER code in attachement. Sanity and performance of the resulting code is important to me. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] Help!!
by George Lambson 09 Dec '05

09 Dec '05

Does anyone know what I would put in my ser.cfg to authenticate by IP address? and have mediaproxy still handle that interaction? Thanks, George

1 0

[Serusers] announcement: student project finished: partiSIPation
by Matthias Liebig 09 Dec '05

09 Dec '05

Hello, recently we, a group of students, finished our project here at iptel. It is a SIP user agent framework. Below you'll find the introduction of our project report which is published at BerliOS [1]. "PartiSIPation is the result of a student project at the Institute for Open Communication Systems (OKS) at the Technical University Berlin. When we started with PartiSIPation in April 2005 there already were some SIP clients available, but none of them was actually usable. So our main goal was to create a SIP soft client that does not have to hide itself from hard clients - SIP phones. Therefore it has to be able to handle several parallel calls and accounts. But we also thought about future applications like online games. So PartiSIPation should be easy integrateable in other applications on different operating systems. To reach all these goals we designed PartiSIPation as follows: PartiSIPation is a SIP user agent which consists of two parts: a Core written in portable C and an easily replaceable GUI. So it can be used as a softphone as well as it can be integrated in any kind of application. It uses external libraries for signaling and speech transfer which are called over interfaces and adapters. An adapter encapsulates the interface of a library and provides a simpler interface expected by the user agent. This makes an easy library replacement possible. Only the adapter has to be edited, no Core changes are necessary. Because PartiSIPation comes with its own SIP state machine a GUI or application can easily place calls by using its simple XML-RPC API. XML-RPC is language independent which means that the GUI can base on Java, GTK, Qt or whatever depending on the preferences of the GUI developer or its purpose. Every call has its own state machine, so parallel calls are no problem. And together with the account management of PartSIPation which allows the registration with several accounts it is possible to use this user agent for complex situations like they are occurring in gaming. Recently, PartiSIPation became Open Source Software and was published at BerliOS: http://developer.berlios.de/projects/partisipation/ ." Best regards, Matthias Liebig [1] http://developer.berlios.de/docman/?group_id=5063

1 0

[Users] RE: [Devel] unixODBC Database Connection Problem
by Lenir 08 Dec '05

08 Dec '05

Ok, I tried using the FreeTDS driver for unixodbc and I got a more detailed error message that might shine a light: Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect IM007:1:0:[unixODBC][FreeTDS][SQL Server]Could not find UID parameter HYC00:2:0:[unixODBC][FreeTDS][SQL Server]Driver not capable 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration These are my entries in /etc/odbc.ini [SYBASE] Driver = /usr/local/lib/libtdsodbc.so Description = Sybase Server Trace = Yes Trafcefile = /tmp/odbc.log #Servername = JDBC Server = xx.xx.xx.xx Port = 2000 DSN = database Database = database #UserID = username # I tried using both UID and UserID, had same results UID = username Password = passwd TDS_Version = 4.2 These are my entries in /etc/freetds.conf: [SYBASE] host = xx.xx.xx.xx port = 2000 tds version = 4.2 _____ From: devel-bounces(a)openser.org [mailto:devel-bounces@openser.org] On Behalf Of lenirsantiago(a)yahoo.com Sent: Thursday, December 08, 2005 12:59 PM To: devel(a)openser.org; users(a)openser.org Subject: [Devel] unixODBC Database Connection Problem Guys, Im trying to use unixodbc.so to connect to a Sybase database using Sybase native ODBC drivers. This is the entry that I have on: /etc/odbc.ini: ============= [SYBASE] Description = Sybase ODBC Data Source UserID = username Password = passwd Driver = ASE Server = xx.xx.xx.xx Port = 5000 Database = database UseCursor = 1 /etc/odbcinst.ini ============ [ASE] Description = Sybase ODBC Driver Driver = /opt/sybase/ODBC-12_5/lib/libsybdrvodb.so FileUsage = -1 UsageCount = 1 The database is on a remote host. I can connect from the openser server using unixodbc's isql tool, as such: [root@ser openser]# isql SYBASE username passwd +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> This proves that unixodbc is installed correctly and that the Sybase odbc driver is installed and functioning properly. However when I try to run openser (debug=4) I get the following: 0(0) fixing /usr/local/lib/openser/modules/auth_db.so www_authorize 0(0) db_init: Connection 'unixodbc://username:passwd@xx.xx.xx.xx/SYBASE' not found in pool Failed to connect The driver reported the following diagnostics whilst running SQLDriverConnect 0(0) auth_db:str_fixup: Unable to open database connection 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration 0(0) XLOG: destroy module ... 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(22871) shm_mem_destroy 0(22871) destroying the shared memory lock [root@ser openser]# This is a snippet of my openser.cfg ------------------------------START-------------------------------- debug=4 fork=no log_stderror=yes listen=yy.yy.yy.yy port=5060 children=4 server_signature=no dns=no rev_dns=no check_via=no syn_branch=yes memlog=1000 sip_warning=no fifo_mode=0666 fifo="/tmp/openser_fifo" fifo_db_url="unixodbc://username:passwd@xx.xx.xx.xx/SYBASE" reply_to_via=no loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/postgres.so" loadmodule "/usr/local/lib/openser/modules/unixodbc.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_radius.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/options.so" #loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" modparam("usrloc", "db_mode", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("usrloc|auth_db", "db_url", "unixodbc://username:passwd@xx.xx.xx.xx/SYBASE") ---------------------------------END---------------------------------------- - Any suggestions? Lenir

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