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[Serusers] Problem with MTU size in SER with MIPv6.
by jose angel calvo 23 Nov '06

23 Nov '06

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[Serusers] NAT and ACK
by Benito Caracuel 23 Nov '06

23 Nov '06

Hello all , This is my scenario: I have SER with Private address behind a Cisco router which Has NAT implemented. I have two UAs with public address each one. UA1 ----------------------> Router Cisco with NAT (public IP) ------------------------> SER with private IP UA2 <----------------------Router Cisco with NAT (public IP) <------------------------ SER with private IP The problem is that when UA1 sends an INVITE with SDP, UA2 reply with 200 ok with SDP but after that, SER doesn't see the ACK. I know that UA1 sends the ACK but this ACK doesn't reach SER. I think that maybe the problem is NAT. Is possible to run SER with private address? How can i solve this? Help me please. Thank you very much. Ben. _________________________________________________________________ Consigue el nuevo Windows Live Messenger http://get.live.com/messenger/overview

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Re: [Serusers] Re: where is the vm.so for voice mail feature ?:
by lawrence k.y. lin 23 Nov '06

23 Nov '06

Hello list, I searched for the voice mail feature and found an example "Example 5-3. Example ser Config File" at the Admin's Guide in the following url: http://old.iptel.org/ser/doc/seruser/seruser.html#AEN1364 Inside that example, there is a statement to load vm.so as follows: loadmodule "modules/vm/vm.so" However, I cannot find vm.so under the ser-0.9.6. Please advise where I can find vm.so for voice mail feature. Thanks and Regards, Larry _________________________________________________________________ View Athletes Collections with Live Search http://sportmaps.live.com/index.html?source=hmemailtaglinenov06&FORM=MGAC01

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[Users] polycom phone proxy failover question
by Zahid Mehmood 23 Nov '06

23 Nov '06

Hi, I am having difficulty configuring polycom phone to failover when one proxy server is not available. I've configured the phone to use sip naptr records that provides two proxy server names. I've configured the "retry timeout" and "retry max count" however, when proxy1 is not available, polycom phone does not send requests to proxy2. Is there any other option I need to configure? I've tried this on polycom 430 and 601 models. Thanks is advance for your help. -- Zahid Mehmood CUIT Network Systems

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[Users] draft-anil-sipping-bla-03.txt (bridged line appearance) support
by Alan Crosswell 22 Nov '06

22 Nov '06

Has anybody implemented BLA support using OpenSER and/or other code? It appears a State Agent is needed to SUBSCRIBE to the UAs and do all kinds of fun stuff. We use Polycom phones which support all this stuff and are migrating cranky customers from a highly functional legacy PBX that heavily use all these "business" features. /a

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[Users] incorrect documentation
by Mark Price 22 Nov '06

22 Nov '06

Hi, I see that the siptrace documentation for openser 1.1.0 says that the function siptrace() will log a packet. However, the correct function name is sip_trace(). Is there a way to go about submitting improvements/fixes to the documentation? Mark Price

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[Users] Loop?
by Kelvin Williams 22 Nov '06

22 Nov '06

Yesterday, something happened-the exact I do not know just yet. However, when looking at the logging that I have my OpenSER performing I seen a ton of INVITES, CANCELs, OpenSER became unusable for a time until I was able to block all SIP traffic at the firewall and let it calm down enough to restart it. During this process calls could not be made (in or out). Is there a way to prevent this type of behavior? It looked as though there were only a handful of calls that caused a continuous loop. What logic is recommended to test for a call that has been handled too many times by the proxy, and to immediately drop it? Thanks in advance.

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[Serusers] Open IMS Core released last Thursday
by Dragos Vingarzan 22 Nov '06

22 Nov '06

Hi *, as you did not had enough of me last week, here I go again: FOKUS recently launched the Open IMS Core. Keeping it short: - project at http://www.openimscore.org (SVN, mail-lists, etc on berlios) - general information at http://www.fokus.fraunhofer.de/ims/OSIMS.php?lang=en or mailto:info@open-ims.org - you get P/I/S-CSCF, HSS, SIP2IMS Gateway, CDiameterPeer (as module for SER, also standalone usable), JavaDiameterPeer (100% Java and decent performance). All in development, so if you look for stability, security or compatibility, you won't find it there yet. -Dragos -- ----------------------------------------- Dipl. Eng. Dragos Vingarzan FOKUS/NGNI Kaiserin-Augusta-Allee 31 10589 Berlin,Germany Phone +49 (0)30 - 3463 - 7385 Mobile +49 (0)163 - 159 - 5221 eMail vingarzan(a)fokus.fraunhofer.de Web www.fokus.fraunhofer.de We could change the world if God would give us the source code... -----------------------------------------------------------------

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[Users] merging string and pseudo-variable in avp_write
by Benko 22 Nov '06

22 Nov '06

Hello! I'd like to implement an avpops serial forking solution as described in http://www.voice-sistem.ro/docs/avpops/ar01s08.html#ex_serial_forking However, i would like to define the adresses as "$rU(a)mydomain.org". What is the correct syntax to merge the string and the variable in avp_write? I would like to do something like this: avp_write("sip:$rU@mydomain1.org", "$serial_fork"); avp_write("sip:$rU@mydomain2.org", "$serial_fork"); where $rU should be parsed. $serial_fork would have a value of e.g. "12632421(a)mydomain1.org" then. Or does this not work at all? How else could i define it? Can someone point me to a fundamental doc where the openser.cfg-syntax is described? My well-tried trial-and-error method doesn't seem to work here... thx Christian

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[Fwd: Re: [Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines?]
by nick 22 Nov '06

22 Nov '06

-------- Original Message -------- Subject: Re: [Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines? Date: Tue, 21 Nov 2006 16:46:42 +0100 From: nick <nick(a)mobilia.it> To: Ovidiu Sas <sip.nslu(a)gmail.com> References: <4562E2D7.7010403(a)mobilia.it> <4562E82E.4070503(a)pernau.at> <45631216.3000800(a)mobilia.it> <6f497e130611210650i409e6cfblfb39b0d8a0b9f2e2(a)mail.gmail.com> <45631873.8010400(a)mobilia.it> <6f497e130611210724p4b3ead9bha95cb848c29d276c(a)mail.gmail.com> Ovidiu Sas wrote: > It seems that you are trying to do a double processing of the SDP > (that's why you have 2 IPs inside SDP). Check you script if you use > force_rtp_proxy and fix_nated_sdp at the same time. > > > Regards, > Ovidiu Sas > > On 11/21/06, nick <nick(a)mobilia.it> wrote: >> Ovidiu Sas wrote: >> > check the '-d' argument. If you don't specify '-d', ngrep will only >> > listen to one interface. '-d any' will force ngrep to listen to all >> > interfaces, including the loopback. >> > >> > >> > Hope this helps, >> > Ovidiu Sas >> >> BTW, I changed my fix_nated_sdp() command to fix_nated_sdp("10") only, >> which is adding the origin IP instead of the mediaproxy one.. I don't >> mind this, but it still seems to be adding the 127.0.0.1 >> >> > Actually, I'm using mediaproxy and fix_nated_sdp at the same time, I am using fix_nated_sdp because for some reason, mediaproxy isn't changing the SDP information in the SIP requests (or at least, it wasn't beforehand). I'll have a look at where it might be overlapping. whoops forgot to cc

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