sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

[Serusers] ser masquard ?
by Рудаков Евгений Валерьевич 03 Jan '04

03 Jan '04

hi, i have some accounts to sip provider, but i dont know how to setup ser 8.12 to masquarde local accounts and domain to external account with domain. like this oper:test@local.net -> 12331223:123@deltafour.com test:test@local.net -> 12331223:123@deltafour.com vase:test@local.net -> 454331223:123@deltafour.com it is possible ? thnks

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Re: [Serusers] SER outbound routing?
by Jiri Kuthan 03 Jan '04

03 Jan '04

At 11:35 AM 12/31/2003, Matthew Moyle-Croft wrote: >> I'm not sure, but maybe when a b2bua module is made ser could act as a >> UAC and register with other proxies. > >I think this is quite important - running a SIP gateway for residential use >is going to be more and more common. I'd really like to see this bit happen >so that a Linux Gateway can be setup (maybe a modified version of one of the >firewall/mail server distribution ones) so that anyone can get a … [View More]

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[Serusers] load balancing and route priority
by Bo 03 Jan '04

03 Jan '04

Hello, Does anyone have any idea on how to config ser.cfg to do load balancing and priority routing? 1. load balancing Two PSTN Gateways (A and B) could terminate the same destination numbers. How to do load balancing (just route calls to them randomly)? 2. priority routing Two PSTN Gateways (A and B) could terminate the same destination numbers. SER always try A first. If A couldn't terminate the call (offline, no circuit ...), then it will try B. In other words, Gateway A always has a … [View More]

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[Serusers] Do these SIP dumps look right ("group" locations entry) ?
by Jim Burwell 03 Jan '04

03 Jan '04

Hi, I was having some problems with calls to what I call "group" locations entries. These are entries where users enter their own SIP address for a particular location entry. The desired behavior is that when someone calls this "group" user, multiple other users phones will ring, and if not answered, voice mail will be called for the original "group" callee user. This doesn't seem difficult, but turns out to be a little tricky if the users enter the more logical, flexible, and user … [View More]friendly "sip:user@samedomain" style addresses instead of "sip:user@ip:port". Obviously, requiring possibly non technical end users to enter the IP:port of all their phones for these group users is unacceptable. Using these domain style "indirect" entries is far nicer, since SER ultimately simply looks up all the users' registered phones in the locations table transparently. Using "sip:user@samedomain" style locations destinations causes these calls to fork back through SER a second time (spiraling back through SER) for each destination. This is where I ran into a problem as far as voice mail was concerned. When each branch of this call (each separate destination in the destination set) was forked back through SER, if the particular user in any given branch was a member of the voice mail group, a voice mail timeout would be set for each, ultimately resulting in a separate call to the voice mail system for each user, for the same call! This caused mass confusion in SER and the voice mail system, resulting in a failed voice mail call, for obvious reasons. The solution was to make sure that in this special case, that only the original "group" user gets queued for a voice mail timeout, and any other branches spiraling back through SER did not cause voice mail to be set up. Unfortunately, there's no easy way that I could figure out to test for this condition in ser.cfg, since these forked branches look just like normal calls to these users (which would be proper to set up a voice mail timeout for). The solution I came up with was to require users to use a special prefix, "g+", when entering their SIP URIs into the group locations entries, marking them for special treatment. The ser.cfg then simply looks for URIs that have the prefix "g+", and if present, makes sure that no voice mail failure_route is set up for these branches. Right before the locations lookup is done, the "g+" prefix is stripped off, and the branch proceeds to call the users phone(s) normally. Pretty simple, user friendly, and requires no special external script, etc, to be called. Viola! It worked. But with some misgivings. This solution actually worked in all my testing. However, I'm troubled by some of the SIP dumps of these calls I've examined. Particularly, I noted how for whatever reason, there seems to be multiple repeated/redundant "408 time out" messages sent back and forth between SER and the phones. Also, for whatever reason, the caller's phone sends repeated ACKs, which get routed to the voice mail system, even after it has answered and is already talking to the VM system. Do these traces look right ? Am I just being paranoid ? Please advise! (ser.cfg and traces attached as ZIP file. Sorry! It would be too big for the list otherwise!) - Jim -- +---------------------------------------------------------------------------+ | Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC | +---------------------------------------------------------------------------+ | "I never let my schooling get in the way of my education." - Mark Twain | | "UNIX was never designed to keep people from doing stupid things, because | | that policy would also keep them from doing clever things." - Doug Gwyn | | "Cool is only three letters away from Fool" - Mike Muir, Suicyco | | "..Government in its best state is but a necessary evil; in its worst | | state an intolerable one.." - Thomas Paine, "Common Sense" (1776) | +---------------------------------------------------------------------------+ | Email: jimb(a)jsbc.cc ICQ UIN: 1695089 | +---------------------------------------------------------------------------+ | Reply problems ? Turn off the "sign" function in email prog. Blame MS. | +---------------------------------------------------------------------------+ [View Less]

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[Serusers] Problem with Record-Route Header
by Franz Edler 03 Jan '04

03 Jan '04

Hi all, I am testing with SER 0.8.11 and I see the following problem: I have record-route enabled. The mysterious behaviour is the following: SER adds a Record-Route header to an INVITE-Request which does not only consist of the host-portion of the proxy but also of a user-name. As a consequence (I assume) loose_route does not work on such a Route-header. Below I have documented the received and forwarded Request of SER. Does anybody have an idea what might be wrong? Help very much … [View More]

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Re: [Serusers] NAT types and rtpproxy
by Jiri Kuthan 02 Jan '04

02 Jan '04

you can, the penalty of permanent relay use is bandwidth consumption and voice latency. -jiri At 12:12 AM 12/31/2003, radan wrote: >Hi !! >Can I use rtpproxy with all types of NAT >(full cone, restricted cone, port restricted cone >and symmetric) ? > >Regards >Andrzej > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] Test SER and commands for diagnostics
by Shirley Toh 02 Jan '04

02 Jan '04

Hi, Like to check does anyone know how to specifically know that SER is being configure properly, is there anyway to test. I am running on Linux 9.0, cannot use the command ngrep and sipsak to do any diagnostics. Please kindly advice. Regards, Shirley

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RE: [Serusers] Messenger 5.0 problem
by dhiraj.2.bhuyan＠bt.com 01 Jan '04

01 Jan '04

That's a bug in the Messenger 5.0 - only Microsoft can fix it. Dhiraj -----Original Message----- From: serusers-bounces(a)iptel.org on behalf of Andy Singh Sent: Wed 31/12/2003 22:42 To: 'serusers(a)iptel.org' Cc: Subject: [Serusers] Messenger 5.0 problem

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sr-users January 2004