sr-users February 2005

sr-users@lists.kamailio.org

215 participants
467 discussions

[Serusers] Aastra 480i
by Jain, Hitesh 07 Feb '05

07 Feb '05

Hi, I am trying to make the new Aastra 480i register with the SER, but its not happening. I was told that the manual configuration on Aastra doesnt woek.So I made the changes in the appropiate files in the TFTP server (aastra.cfg and MAC add.cfg), but no luck. Can anyone please send me their config files just for reference? Thanks, HItesh.

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RE: [Serusers] ser features
by peter.3.edwards＠bt.com 07 Feb '05

07 Feb '05

Ah. Thanks very much for the replies Dragos and Jan - I'll take a closer look at the developer documentation (I can't believe I missed it!). Hmm .. I think I have got the wrong end of the stick somewhere, though - I thought ser was a variant of a SIP Application Server. If it is just a SIP proxy that probably doesn't fit the purpose we're looking at (not that it'll stop me installing it anyway .. ;). Is there anything in the open source world in the SIP Application Server space, or is ser as close as it gets? Many thanks for the very fast responses!! Peter. > -----Original Message----- > From: Jan Janak [mailto:jan@iptel.org] > Sent: 07 February 2005 11:37 > To: Edwards,PR,Peter,XKD44 R > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] ser features > > On 07-02 00:09, peter.3.edwards(a)bt.com wrote: > > Hi, > > > > Being one of those who tend towards open source, I'm trying > to put forward ser for use within my current project as, from > my reading, it seems to have most of the features I think > we're going to need. Problem is, before I've had a chance to > get it in and play with it, there's a paper sift going on > which may see it being shelved before I get a chance to even > propose it. :( > > > > I realise it's entirely unforgivable, netiquette-wise, but > I was hoping if I posted the list of criteria I was looking > at whether someone could help confirm or deny what I've > cobbled together wrt ser. Any links to more information on > the web would be ideal. > > > > Any help at all would be gratefully received! > > > > Many thanks, > > > > Peter. > > > > 1) Support for SIP - (preferably 3GPP ISC interface) > > - Obviously SIP is supported but I can't see any > explicit mention in the docs wrt to 3GPP ISC .. ? Has ser > been developed with this in mind? > > SIP yes, 3GPP ISC no. > > > 2) Provide flexible application run time support in terms > of standard / well-defined API sets such as SIP servlets > > - As ser is written in C, it's obviously not exposing > SIP servlets internally, but I can't seem to find a specific > API specificiation. I think it sounds like applications are > created as C modules which plug into ser. Is that right (I'm > not a C developer, so any clarification appreciated)? Is it > a ser-properietary interface or something that follows a > particular standard? > > There is nothing comparable to SIP Servlets, SER is not a servlet > container, but a SIP message mangler with the possibility to keep > transaction state. > > SER modules are written in C and can access SER internals directly. > Each module can export function that the administrator can then call > in the configuration file. > > There is no particular standard for this, it is very > similar to the Apache > module API. > > > 3) support carrier grade non functional requirements e.g. > %age availability, multi-site installation, latency, throughputs etc. > > - I can't see any specific claims for reliability, or > any info on how to deal with redundancy etc. Has this been > looked at before? > > Yes, high availability extensions are available under comercial > license from iptel.org. > > > 4) Any interfaces that can be exposed to application logic > hosted on a remote platform in an untrusted environment - > e.g. a Java RMI, Web Services etc. > > No. > > > - Does ser expose anything else, other than SIP? How > would a third party application running on, let's say for > argument's sake, a J2EE application running on a separate > JBoss server? Would a C module need to be written and > plugged into ser to expose, say web services? Has anything > like that been done already? > > No, that is not possible. SER is not an application server, it is a > sip proxy. > > > 5) Application Developer support / tools > > - Is there anything like a forum or tools to aid a > module developer? > > The C sources and SER developers guide describing the API. > > http://iptel.org/ser/devel.html > > > > > 6) OSS integration > > - Is there any? > > No. > > Jan. >

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TR: [Serusers] rtpproxy general questions
by Christian de Balorre 07 Feb '05

07 Feb '05

Thanks very much, very valuable info indeed. I have a few questions that must have been asked a thousand times but I did not find any clear answer in archive : Or can you point me to relevant doc ? 1) fix_nated_sdp : - I guess you use this fonction if you want a nated ua to talk directly to another ua w/o rtproxy ? - what is exactly the sdp field modified ? c ? o ? - is it relevant if you use force_rtp_proxy ? (my guess is no...) 2) force_rport() - Could not figure out the usefulness of this command... Deprecated by fix_nated_contact ? 3)force_rtp_proxy 'f' flag -I really cannot figure out what it does (or does not!) on a sip packet. I would just blindly use it but I would like to understand nonetheless :-) Also I would need an advice : You often see this portion of code at the beginning of a "main routing logic" config file : record_route(); # loose-route processing if (loose_route()) { t_relay(); break; If I am correct it says "if this packet as already been processed by a sip proxy, route it accordingly". But suppose a sip INVITE packet went through a first Proxy, then it will be routed back instead of being correctly processed with lookup("location") for exemple. Would if ( !(method=="INVITE")) { if (loose_route()) { t_relay(); break; }; be more appropriate in this case ? Thanks for your patience - Christian -----Message d'origine----- De : Marian Dumitru [mailto:marian.dumitru@voice-sistem.ro] Envoyé : jeudi 3 février 2005 21:34 À : Christian de Balorre Cc : serusers(a)lists.iptel.org Objet : Re: [Serusers] rtpproxy mess Hi Christian, To configure rtpproxy to do bridging, just set the listen parameter as "-l addr1/addr2". In your configuration I guess addr1 is the private address and addr2 is the public one. But be careful about using a chain of rtpproxy - you can end having a dead-lock for the media relaying. More information about the force_rtp_proxy flags can be found in modules/nathelper/nathelper.c at the beginning; also take a look at the example modules/nathelper/examples/alg.cfg. Best regards, Marian Christian de Balorre wrote: > Thanks for your previous rtpproxy workaround. > > Here is actually what I want to do (not really extraordinary, inter-sites > design) : > > > __ priv net __ pub net __ priv net __ > |__|<------->|__|<------------>|__|<------------>|__| > ua1 rtp rtp ua2 > site1 proxy1 proxy2 site 2 > > rtp stream should always flow between ua1 et ua2 through rtpproxy1 and > rtpproxy2 (or vice versa of course) > each rtp server is also a ser server with 2 interface, one in priv net and > the other in pub net > there is no nat involved > > I guess we can call this bridging... Can rtpproxy do that ? >

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[Serusers] error: mediaproxy/getContactURI(): error parsing Contact body
by Mohamed Omar 07 Feb '05

07 Feb '05

I'm getting the error below on my SER. I'm running SER.8.14. does any one know what causes this. Everything is working but I get this error. thanks 2(12599) error: mediaproxy/getContactURI(): error parsing Contact body 2(12599) error: mediaproxy/getContactURI(): error parsing Contact body 2(12599) Trying to register 4(12605) Trying to register 2(12599) Trying to register 2(12599) Registered 3(12602) 2(12599) error: mediaproxy/getContactURI(): error parsing Contact body 2(12599) error: mediaproxy/getContactURI(): error parsing Contact body 2(12599) Trying to register 4(12605) Trying to register 2(12599) Trying to register 2(12599) Registered 3(12602) error: mediaproxy/getContactURI(): error parsing Contact body 3(12602) error: mediaproxy/getContactURI(): error parsing Contact body 3(12602) Trying to register 1(12596) Trying to register 3(12602) Trying to register 3(12602) Registered 4(12605) error: mediaproxy/getContactURI(): error parsing Contact body 4(12605) error: mediaproxy/getContactURI(): error parsing Contact body 4(12605) Trying to register 2(12599) Trying to register 3(12602) error: mediaproxy/getContactURI(): error parsing Contact body 3(12602) Trying to register 1(12596) Trying to register 3(12602) Trying to register 3(12602) Registered 4(12605) error: mediaproxy/getContactURI(): error parsing Contact body 4(12605) error: mediaproxy/getContactURI(): error parsing Contact body 4(12605) Trying to register 2(12599) Trying to register --------------------------------- Post your free ad now! Yahoo! Canada Personals

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[Serusers] 't_relay_to_tcp' function not found in TM module
by scm-j＠nuntius.com 07 Feb '05

07 Feb '05

I am trying to use t_telay_to_tcp instead of t_relay to send the SIP messages over TCP, instead of UDP. However, when I run SER, it errors out saying that it is unable to locate 't_relay_to_tcp' I looked at the TM source and verified that the USE_TCP compile flag is defined in the main Makefile. I am using SER version 0.8.14 Thanks. SCM

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[Serusers] ser features
by peter.3.edwards＠bt.com 07 Feb '05

07 Feb '05

Hi, Being one of those who tend towards open source, I'm trying to put forward ser for use within my current project as, from my reading, it seems to have most of the features I think we're going to need. Problem is, before I've had a chance to get it in and play with it, there's a paper sift going on which may see it being shelved before I get a chance to even propose it. :( I realise it's entirely unforgivable, netiquette-wise, but I was hoping if I posted the list of criteria I was looking at whether someone could help confirm or deny what I've cobbled together wrt ser. Any links to more information on the web would be ideal. Any help at all would be gratefully received! Many thanks, Peter. 1) Support for SIP - (preferably 3GPP ISC interface) - Obviously SIP is supported but I can't see any explicit mention in the docs wrt to 3GPP ISC .. ? Has ser been developed with this in mind? 2) Provide flexible application run time support in terms of standard / well-defined API sets such as SIP servlets - As ser is written in C, it's obviously not exposing SIP servlets internally, but I can't seem to find a specific API specificiation. I think it sounds like applications are created as C modules which plug into ser. Is that right (I'm not a C developer, so any clarification appreciated)? Is it a ser-properietary interface or something that follows a particular standard? 3) support carrier grade non functional requirements e.g. %age availability, multi-site installation, latency, throughputs etc. - I can't see any specific claims for reliability, or any info on how to deal with redundancy etc. Has this been looked at before? 4) Any interfaces that can be exposed to application logic hosted on a remote platform in an untrusted environment - e.g. a Java RMI, Web Services etc. - Does ser expose anything else, other than SIP? How would a third party application running on, let's say for argument's sake, a J2EE application running on a separate JBoss server? Would a C module need to be written and plugged into ser to expose, say web services? Has anything like that been done already? 5) Application Developer support / tools - Is there anything like a forum or tools to aid a module developer? 6) OSS integration - Is there any?

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[Serusers] general sip problem
by arda balkanay 07 Feb '05

07 Feb '05

hi i have a sip network and when trying to make a call, caller can receive called party's voice but called guy cannot hear any voice. When called guy tries to make a call he can hear called party's voice and similarly new called guy cannot hear any voice. i wanna ask that what can be the reason about my problem Thanks best regards Arda Balkanay

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Re: [Serusers] Modifying "From" header field of an INVITE message
by Andr�s Parra L. 07 Feb '05

07 Feb '05

Don't hesitate anymore, use asterisk to do so with the SetCallerID command in extensions.conf. You should forward the call through Asterisk from SER. If you don't want to have all the media pass through you, and i beleive you don't, use canreinvite=yes in sip.conf, you should fight a little bit with the codecs with allow and disallow but it should be fine. The next links could help you with the reinvitation issue: http://www.voip-info.org/wiki-Asterisk+SIP+media+path http://www.voip-info.org/wiki-Asterisk+sip+canreinvite Hope it helps. Andres __________________________________ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com

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[Serusers] Trusted IP and security.
by Tom Lowe 07 Feb '05

07 Feb '05

Hi all. I have a "security" question regarding "trusted IP's". Is it possible for someone to SUCCESSFULLY spoof an IP and actually make working calls? For example, '10.10.10.10' sends calls to SER (or any other proxy server) at 20.20.20.20, but actually spoofs the IP by sending an IP address of 30.30.30.30, which happens to be trusted by the SER at 20.20.20.20. I ask because I'm having a discussion with a vendor who is trying to tell me that using trusted IP's for SIP validation is insecure and easily hacked. I don't think it is because when SER gets an INVITE from 30.30.30.30, it is going to send it's progress messages to 30.30.30.30, regardless of the contents of the SIP messages....so the spoofer at 10.10.10.10 won't get any of the progress messages, and more importantly won't be able to establish a talk path. I suspect he may still cause SER to initiate some brief outbound calls, but they should fail when the SIP protocol falls apart. Does anyone have any thoughts on this? Tom

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[Serusers] SER : no voice between two local sip phones
by fauzi＠aplio.com.my 06 Feb '05

06 Feb '05

Dear all, I had installed the SER Server at our own LAN. I had two sip phones connected locally via LAN. Both units can dialed each other without any problems but there are no voice on these two phones. Can you advise me on this. Dialing using Ip without going to the server had no problem at all but not when it use the server. Regards, fauzi

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sr-users February 2005