sr-users July 2006

sr-users@lists.kamailio.org

206 participants
415 discussions

[Users] parsing P-Asserted-Identity
by Klaus Darilion 17 Jul '06

17 Jul '06

Hi! I wonder how I can parse the body of the P-Asserted-Identity: header. This header allows several syntax, e.g: P-Asserted-Identity: "Klaus Darilion" <tel:+43123> P-Asserted-Identity: tel:+43123;context P-Asserted-Identity: tel:201-555-0123;phone-context=+1 P-Asserted-Identity: "nice guy" <sip:ng;abc=def@nicedomain;foo=bar> In my case I want to: 1. check if it is a tel URI or not 2. Get the phone number from the addr-spec 3. Get the parameters (phone-context, ...) Are there generic functions available to parse name-addr, addr-spec, SIP-URIs and tel URIs? Or do I have to use avp_check and avp_subst with some complicated regexps? Thanks Klaus

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[Serusers] Regarding REGISTER message
by Ramachandran Gunasekaran 17 Jul '06

17 Jul '06

Dear All, We are SER as Registrar server and sending REGISTER message using SIPP. Now, after sending REGISTER message it is being received by the SER. The utility /usr/local/sbin/serctl ul show correctly shows the contact details but it is NOT writing in the text file. The configuration file for SER is being attached for kind reference. Any pointer will be highly appreciated. Regards, Ramachandran

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[Serusers] DNS SRV records
by Mike Cassidy 17 Jul '06

17 Jul '06

Hi, I am having trouble with my sip-phone recognising my domain name, and I am wondering if I have set my DNS records correctly. If someone can see something that is missing from my DNS record, I would really appreciate some help. So far, I can register using my IP address, but I can't register using sip.mydomain.com, which is what I would like. Any suggestions? Many thanks, Mike $TTL 14400 @ 86400 IN SOA domain1.com. domain2.com. ( 2006051301 ; serial, todays date+todays 86400 ; refresh, seconds 7200 ; retry, seconds 3600000 ; expire, seconds 86400 ) ; minimum, seconds mydomain.com. 86400 IN NS ns1.mydomain.com. mydomain.com. IN A 1.2.3.4 localhost.mydomain.com. IN A 127.0.0.1 mydomain.com. IN MX 0 mydomain.com. _sip._udp IN SRV 0 0 5060 sip.mydomain.com. mail IN CNAME mydomain.com. www IN CNAME mydomain.com. ftp IN A 1.2.3.4 ns1 IN A 1.2.3.4 sip IN A 1.2.3.4

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[Users] Capacity planning on OpenSER / SER
by Hamid Ali Asgari 17 Jul '06

17 Jul '06

Hi, I was wondering on OpenSER and came to a Question: How can call routing to SIP gateways be done based on their capacity? Lets say we have two SIP gateways each capable of handling 60 cuncurrent calls. How could I tell openser to forward the call number 61 to the next gateway? How would open ser know if the first gateway has free resources (previuos calls ended), and so on. Thanks, Hamid

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[Serusers] Help needed on SER configuration
by Ramachandran Gunasekaran 17 Jul '06

17 Jul '06

Hi All, Can you tell me how to configure SER as location server with dbtext mode? If so, please send me the configuration details for the same ASAP. Thanks for your co-operation. Regards, Ramachandran G

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[Serusers] Re: SER database sharing
by Nick Hoffman 17 Jul '06

17 Jul '06

Hi Greger. On October 25, 2005, you wrote: > At least one thing is for sure: I have now registered a two-digit number > of people who are struggling with and proposing various solutions to load > balancing/failover. We really need to find a solution soon, so all these > bright people can spend their resources on tackling problems that will > bring ser even further!! Such a solution should be a "best practice" > that is *good enough*, and I would vote for simplicity. People who > really need that top-performance/hardware are capable of tuning and > fixing themselves (and maybe improve best-practice along the way), all > the others need a simple, well-described setup. > That is why I have been working with Andreas to try to mirror his > efforts in an onsip.org setup that we will document and make > available as soon as it is ready for prime time. I suggest that anybody > who have opinions or suggestions put them forward so that everything will > be taken into consideration. I was wondering what progress has been made on this front. Obviously the documentation isn't ready since it hasn't been released, but I'd be interested in reading what's been done so far, and contributing my knowledge wherever I can. Cheers, -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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Re: [Serusers] OT: xlite now with video and IM
by zhaomin 17 Jul '06

17 Jul '06

I download the xlite3.0 and testing with Asterisk server ,All the function is all right but video function cann't start to the caller ,Can anyones tell what's the back server needed? Is anyone know why? Regard ZhaoMin ----- Original Message ----- From: "Andrey Kouprianov" <andrey.kouprianov(a)gmail.com> To: <serusers(a)iptel.org> Sent: Friday, July 14, 2006 4:16 PM Subject: Re: [Serusers] OT: xlite now with video and IM > Yes, X-lite ver.3 now even supports presence and looks nicer. However, > as far as i know, it has now only 2 concurrent lines, instead of 3. > > Andrey. > > On 7/14/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: >> Hi! >> >> I just found out that xten offers a new xlite, based on eyebeam. AFAIK >> this is the first free available Windows SIP client with video and IM. >> As IMO eyebeam is the best client for testing and debugging purposes >> (because you can configure everything) I think this is interesting for >> you too. >> >> regards >> Klaus >> _______________________________________________ >> 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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[Users] vhow do i use SIP account to redirect
by ram 16 Jul '06

16 Jul '06

Hi all Till now i have seen examples of rewriting with IP address and port now i have another router from VoIP provider, where is give me sip username and password with IP address and port now i am planning to disconnect IP based authentication provider and go with SIP provider, becoz of Quality and price so how can i change the rewritehostport("provider IP:5060"); to my username/password:ip: port can some one suggest me what is the best method to do ram

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[Serusers] problem with t_relay_to
by Luis Silva 16 Jul '06

16 Jul '06

Hi all!! I'm using the t_relay_to() function from the tm module but I'm having problems sending the message to the port that I want. I have in my cenario a proxy server(port:5062) that relays the message to another ser server that as the listening port 5063. Both ser servers are on the same machine, but has you can see, in different ports. I'm using t_relay_to("5063","10.10.12.4") for relaying the message, but instead of going to port 5063 the message goes to the port 5060 where I have another server. What might be the problem? Can you help? tks a lot Regards, Luis

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[Serusers] how do i use SIP account to redirect
by ram 16 Jul '06

16 Jul '06

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sr-users July 2006