similar to: Fwd: [Serusers] CDR mediation for VoIP

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

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 >

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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

I think that the range of this question is too large. You should tell us what your scenario is. And tell us more about your configurations. 2006/2/2, Jack Wei <cowlemon@yahoo.com>: > hi, > > I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do > load-balancing. I'm using Asterisk as a voicemail application only and have > successfully

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

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

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

This post cannot be left without comment. People who don't know you or Adrian might get a wrong impression. I know Adrian quite well and know that he is one of the real experts in this industry and he and his stuff does not deserve such a treatment. I would recommend that you change your attitude. It seems like you did not get what you want (for free) and you complain like a small child. An

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.

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

Hi everyone, I use the boot package within the MBESS package to automate much of the hard part for folks interested in performing a (simple) mediation model. The function mediation() works well, except for a (probably) unrealistic artificial data set. When I apply the bootstrap I sometimes get: Error in t.star[r, ] <- res[[r]] which is (I think) due to Inf and -Inf and NaN values produced for

Hello all, I apologize for the simplistic question, but I have been having some trouble learning how to do mediation analysis in R. Ideally, I would like to use Preacher's Bootstrapping test for mediation (Preacher & Hayes, 2004). I have attempted to use the mediate package to set this up, using code that looks basically like this: model.m <- lm(data$outcome ~ data$mediator +

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

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