sr-users July 2004

sr-users@lists.kamailio.org

178 participants
449 discussions

[Serusers] log
by Blanca Mayayo 29 Jul '04

29 Jul '04

Hi! The log messages are stored by default in /var/log/messages, but I would like to know which way to obtain only the log messages that I wrote in the configuration file, without all the debug messages. On the other hand, is there a way to know exactly how the debugging mode acts (which level corresponds to each 'd' in the debug option), when I make an instance of ser, and what relation it has with the debug option inside a configuration file? Thank you, Blanca _________________________________________________________ Envia tus postales desde Lycos Postales. Envía la tuya desde http://postales.lycos.es

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[Serusers] re : re : redirect using external application.
by Gandhi, Jignesh 29 Jul '04

29 Jul '04

Hello, We have the following setup. A proxy forwarding all the calls to our SER and the SER replies to the proxy with the appropriate location where the call should be routed (statless). So the first call could go to A and the next call could go to C and so on. I want to be able to do some load balancing and smart routing. By smart routing I mean that a program running on the same box as the SER will keep track of the health status of all the destination boxes like A, B, C stated above. So when an invite comes from the proxy, How can I call an external let's say "c" program which will return the destination address the SER should should return to the Proxy to use. I have looked at exec_dst but the admin guide provides only exec_msg examples. I have a feeling that I could use it, but not sure. I hope that I explained the setup properly. If anyone has any questions please ask. thanks in advance for your help. Jignesh Gandhi Software Engineer II Jignesh.Gandhi(a)glenayre.com

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[Serusers] time-of-day-based least cost routing?
by Michael Przybylski 29 Jul '04

29 Jul '04

Has anybody figured out how to do this? Can anybody share any pointers? Best regards, Mike Przybylski

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[Serusers] about fifo
by murat 29 Jul '04

29 Jul '04

Hello, If I run the "serctl moni " as 'roo' , it executes not a problem. But , if I run it as a not 'root'' , for example ,as "murat", I have following errors!. ****************************************** Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config ************************************************** Even I have ' fifo="/tmp/ser_fifo" ' lines on my default 'ser.cfg' file. Question <1>: What is the problem about this? Question <2>: I have some SIP sicripts, and how can I run it?. Thank you.

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[Serusers] t_check_status/t_on_failure does not work with ser v.0.8.12 rpm
by David Lee 29 Jul '04

29 Jul '04

Hello, In the admin documentation of ser, it says that all messages greater than 300 are considered failures when using t_on_failure. In our scenario, we would like the ability to forward to a voice mail on 4xx failure, but not on 302 failure. We are using XTEN Pro softphones and want to be able to allow users the functionality to forward to another SIP url (uses 302). But when ser is seeing 302 (and t_on_failure is active) it considers it a failure and immediately runs failure route block (to voicemail). So our users cannot using forwarding to URL. An example would be ... When I'm out for lunch I would like to forward my line to my secretary first before going to voicemail. We saw some mention of t_check_status ... but ser wont start with that in the cfg. We are using ser v.0.8.12 rpm. Is that in a unstable version only. If some could post how to accomplish this it would be appreciated. Thanks Kurt

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[Serusers] Questions about interconnecting media proxy on SER.
by Kenny Lam 29 Jul '04

29 Jul '04

Hi all, I'm now trying to setup some SER servers with location based routing ability which can handle the traffic according to the source. Before getting to that, I want to know if the media proxy of SER can be interconnected together to form a larger adaptive server? The scenario is as follow : SIP phone <=> Media Proxy A <=> Media Proxy B <=> SIP phone From the Media Server readme, I just found out that the dispatcher.py can load balance the traffic by using the DNS SRV records. Any people got some ideas? Thanks a lot! Kenny Lam SIP Application Engineer Deltapath Commerce & Technology Limited --------------------------------------- SIP By Deltapath! www.deltapath.com

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[Serusers] 483 when registering
by Guillaume Leclanche 28 Jul '04

28 Jul '04

Hello, I don't understand why ser (latest cvs) returns me a 483 "Too Many Hops" when I try to register. I've gone through the list, and I've not seen anything significant. I believe it's because I have not configured the domains for which users can register, but I admit I don't understand where this is to be configured. I'm new to ser, but not sip, and I'm open to any help :) Guillaume Here is the captured network flow : Frame 1 [...] Internet Protocol, Src Addr: 137.194.8.164 (137.194.8.164), Dst Addr: 137.194.14.1 (137.194.14.1) Version: 4 [...] Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x0f48 (correct) Source: 137.194.8.164 (137.194.8.164) Destination: 137.194.14.1 (137.194.14.1) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 350 Checksum: 0x2b99 (incorrect, should be 0xeee3) Session Initiation Protocol Request-Line: REGISTER sip:phoenix.rezel.com SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 137.194.8.164:5060;branch=z9hG4bK2014234759 From: <sip:momo@rezel.com>;tag=1896043603 SIP from address: sip:momo@rezel.com SIP tag: 1896043603 To: <sip:momo@rezel.com> SIP to address: sip:momo@rezel.com Call-ID: 1918743559(a)137.194.8.164 CSeq: 10 REGISTER Contact: <sip:momo@137.194.8.164:5060> Max-Forwards: 5 User-Agent: eXosip/0.1 Expires: 3599 Content-Length: 0 Frame 2 [...] Internet Protocol, Src Addr: 137.194.14.1 (137.194.14.1), Dst Addr: 137.194.8.164 (137.194.8.164) Version: 4 [...] Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x0ead (correct) Source: 137.194.14.1 (137.194.14.1) Destination: 137.194.8.164 (137.194.8.164) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 515 Checksum: 0x8818 (correct) Session Initiation Protocol Status-Line: SIP/2.0 483 Too Many Hops Status-Code: 483 Resent Packet: False Message Header Via: SIP/2.0/UDP 137.194.8.164:5060;branch=z9hG4bK2014234759 From: <sip:momo@rezel.com>;tag=1896043603 SIP from address: sip:momo@rezel.com SIP tag: 1896043603 To: <sip:momo@rezel.com>;tag=93a7c1442d2e114fba8f7e6047741fc5.9068 SIP to address: sip:momo@rezel.com SIP tag: 93a7c1442d2e114fba8f7e6047741fc5.9068 Call-ID: 1918743559(a)137.194.8.164 CSeq: 10 REGISTER Server: Sip EXpress router (0.8.99-dev (i386/linux)) Content-Length: 0 Warning: 392 137.194.14.1:5060 "Noisy feedback tells: pid=8466 req_src_ip=137.194.14.1 req_src_port=5060 in_uri=sip:phoenix.rezel.com out_uri=sip:phoenix.rezel.com via_cnt==6"

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RE: [Serusers] Asterisks to ser to asterisk (voicemail)
by Dave Bath 28 Jul '04

28 Jul '04

As you guys seem to have some experience with this area perhaps I will re-ask my question this thread.. I have SER-->asterisk forwarding working fine now, and I have divert on unavailable working. However, I have two outstanding problems: (1) If a user is called with their alphanumeric ID instead of their numerical alias, * does not pick up the call. This is as expected, as the dial pattern in * is _[1-9] [0-9] [0-9] [0-9]. However, it must be fairly common to call people with their email addresses for example... so how can I make ser pass the alias to * instead of the alpha URI? (2) If a user is offline, I get a 404 immediately, instead of anything else - for example diverting immediately to vm. I don't quite understand this at the moment.. as I have the t_on_failure set up before the location lookups... does the t_on_failure not catch 404 failures? Any thoughts would be very much appreciated.... anything I can provide, please let me know... Thanks again everyone, Dave -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of GR S Sent: 28 July 2004 21:31 To: jon(a)bostontech.com Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Asterisks to ser to asterisk (voicemail) Hello, --- jon(a)bostontech.com wrote: > yes, i know that this will work, but the issue is that not every sip user > who is called has voicemail. I want SER to determine who should be > rerouted or who shouldn't. Still you dont need to worry. Let all un-attended calls come back to Asterisk. It will drop the calls if it can't find a mail box number. May not be the right method, though. > -Jon > > > > > > GR S <gr_sh2003(a)yahoo.com> > 07/28/2004 04:02 PM > > To: jon(a)bostontech.com, serusers(a)lists.iptel.org > cc: oej(a)edvina.net, andres(a)telesip.net > Fax to: > Subject: Re: [Serusers] Asterisks to ser to asterisk > (voicemail) > > > Hello, > > --- "Olle E. Johansson" <oej(a)edvina.net> wrote: > > > Andres wrote: > > > > > > > >> > > >> My question is, is there any way to have ser receive a call from > > >> asterisk and then reroute it back to the same asterisk server without > > > >> getting a "loop detected" error? > > >> > > > Aren't you seeing this "loop detected" on the Asterisk CLI?? If so > > > should post this in the Asterisk list instead. We know this happens > > > anytime you try to loop a call back to Asterisk, but its Asterisk who > > > complains. Not SER. > > > > > Answer from the Asterisk users list :-) > > > > No, there's not a way to do it, but maybe to issue a 302 redirect. > > Haven't tried it, but that may work. > > > > The Loop Detected stuff is annoying, yes. > > > > /O > > > > >From a great fan of Asterisk and SER :-) > > I am not sure about the exact problem, but there is another way to acheive > this. You dont need to > re-route the calls back from SER to Asterisk. Set a timeout in the > Asterisk Dial statement and > forward the call to SER. If the callee attends the call, you can talk, and > if not, make Asterisk > forward the call to voicemail when it hits the timeout. I have this > feature enabled in a local > system running SER on 5060 and Asterisk on 5070. > > Best Regards, > ===== Girish Gopinath <gr_sh2003(a)yahoo.com> __________________________________ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Asterisks to ser to asterisk (voicemail)
by jon＠bostontech.com 28 Jul '04

28 Jul '04

(i'm resending this because I don't think it got posted right...) I'm running into a little problem with forwarding. This is what's happening: I have a PBX connected into an asterisk box, allowing the PBX users to make SIP calls from their regular phones. I'm also running a SER box with my own sip domain. I have the SER box send unavailable calls to voicemail on the Asterisk server. Now, when someone uses the PBX phone to call a sip user from my sip domain and they are unavailable, i'm getting a "loop detected" from ser. PBX -> Asterisk -> Ser -> Asterisk I can see why i am getting this (since from ser's perspective it's doing a loop), even though once Ser forwarded the call to asterisk, it won't receive it again, since the incoming call is in a different context. My question is, is there any way to have ser receive a call from asterisk and then reroute it back to the same asterisk server without getting a "loop detected" error? Thanks -Jon

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RE: [Serusers] 0.8.14 headache....
by Dave Bath 28 Jul '04

28 Jul '04

Its ok.. finally found out what it was! D ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Dave Bath Sent: 28 July 2004 20:47 To: serusers(a)lists.iptel.org Subject: [Serusers] 0.8.14 headache.... Hey guys, I am having a small headache with serweb and the latest version of ser.. tried to upgrade, and things went moderately smoothly... the main problem was forgetting to use the "prefix=..." command when compiling the source! However, whenever I have upgraded, the database always seems to be a problem ... I don't know why this is, and the serweb database fields never seem to be present in the ser_mysql.sh script... perhaps I have done something very strange. However, I now seem to have a functioning database and ser install again, and it has decided to start loading my config again, so all is well. BUT! I seem to have lost admin permissions, particularly for administering ACL's through serweb... I can't figure out which combination of database entries to use. So far I have 'perms' field in 'subscriber' table set to 'admin' And I can see I need to put something in the admin_privileges table, but I'm not sure of the values.. Can anyone who has it working please have a look in their DB and let me know the values? Many thanks Dave

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sr-users July 2004