similar to: Re: [Serusers] high-availibility setup using f5 bigip

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

Sorry it's asterisk-users@lists.digium.com --- harry gaillac <gaillacharry@yahoo.fr> a ?crit : > Luca, > > you may find help here: > > http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/ > http://www.asteriskdocs.org/ http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large > > ask for help to asterisk-users@lists.digium.org > > Regards >

FYI, for those of you who aren't on the serusers list. I'd like to hear how others can get this working in small Asterisk settings; I don't really have the time to implement it, but it looks very interesting. JT >To: serusers@iptel.org >From: Adrian Georgescu <ag@ag-projects.com> >Date: Mon, 31 May 2004 23:05:47 +0200 >Subject: [Serusers] CDR mediation for VoIP

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

Hello Robert, Ser, Asterisk, Mysql and Freeradius are working fine , I'm feeling tired with CDRRTool . I will use an other billing system. Thanks for your answer. Regards Harry --- Robert Zorop <rzorop@gmail.com> a ?crit : > HI, i've got a working config of ser 0.9.6, > freeradius, MySQL, and CDRTool > 4.5.3. I can't get the quotaCheck script working, i > think

Hi I tried Xten, its very good, because it can stay in the taskbar (next to the clock) and start when windows starts, and is allways ready to receive calls. Maybe it s the best way to introduce VoIP to my company workers.... But theres a feature that s missing (or I couldnt find), there s no way to connect this softphone with the adress book. I think this feature is very important, because

You will also need your SIP clients that are behind the same NAT to support ICE (Interactive Connectivty Establishment) if you want calls between them. Xten Eyebeam and Snom phones are the only ones I'm aware of that support it. On 7/5/05, Ricardo Martinez <rmartinez@redvoiss.net> wrote: > And even worst. > There are some kind of NAT that STUN does not work. > You can check

I don't want to offend you but you should have a look to sems . You won't find docs to help you at asterisk.org --- "lucape@inwind.it" <lucape@inwind.it> a ?crit : > hello > > help me to configure ser + asterisk > how to do the configuration? > > Luca > > > > ____________________________________________________________ > 6X

Hi to all, I read http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Integration Is it possible to use ser as a presence server instead of LCS 2005 ? Harry ___________________________________________________________________________ D?couvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/R?ponses pour partager vos

lqbal, I do plan on having alot of users. Two markets I'm trying to get some volume users from are: residential consumers and business users. Residential consumers should get basic line services such as their own DID, voicemail, caller-id, call-waiting, three-way calling, and basically, all the standard features you get from companies like Vonage, etc. This particular market base

Waldo, How do you let your customers manage 'their' PBX. I too have a setup like you. However, I installed a * server for each customer, via vserver. I'd like to now what kind of software/webbased package you use for this. I also have SER installed as a front-end server for the * servers. But, as I'm still not very into SER, don't know exactly how this fits in. Should I use

What's wrong with ARA (asterisk realtime architecture) from voip-info: Asterisk, SER and MWI http://mail.iptel.org/pipermail/serusers/2004-December/013727.html Actually I wrote a patch for this and it supports ast_data too. What you do is tell asterisk that all of your phones IP addresses are your SER machine. Then when a message gets left Asterisk sends the NOTIFY to username at

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If this has already been covered, please point me to that (I didn''t find anything in my searches)... We are using F5 BigIP LTM load balancers. They have many pools of Mongrels they load balance across, and I of course want the F5 to know when a Mongrel goes down or is unavailable, etc. To do that, I need to have an F5 health monitor for HTTP make a request to the Mongrel. We do this

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail.

Has anyone successfully puppetized BigIP (F5)? I''m specifically trying to figure out a path in making our BigIP instances be under puppet so all the VIPs, pools, profiles, etc. are all under puppet control. My requirements are probably going to be fulfilled even with just uploading a rules file if a delta''s detected. The main issue I have now is figuring out when to reload the

Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI> -- Executing Dial("SIP/16008-3d17", "SIP/16007&SIP/16006|20|tr") in new stack -- Called 16007 -- Called 16006

Hello, I try set Ua---SER----Asterisk (voicemail/ARA) | Ua ser stable asterisk cvs head I read http://mail.iptel.org/pipermail/serusers/2005-February/015997.html to forward unavailable or busy sip agents to asterisk voicemail in failure route. How may I configure extensions.conf and ser.cfg ? I have been trying without success! Regards Harry

Beautiful. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith Sent: Monday, June 27, 2005 3:48 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] RE: [Serusers] *** SER - Asterisk On Monday 27 June 2005 14:50, harry gaillac wrote: > I don't want to offend you but you

We are using Centos behind an F5 Bigip load balancer. The linux box is using bonding and tagged VLAN's Everything works fine except that when traffic is forwarded from the BigIP to the linux box on the VLAN where the web server is running the linux box returns the traffic on the wrong VLAN, It returns traffic on the lowest ordered VLAN. ie. here is a tcpdump on my load balancer showing