similar to: SIMPLE support in Asterisk?

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

Hi! Does * do DNS-lookups when outgoing calls are placed to a different SIP domain? Can I call from <sip:1000@mydomain.com> to <sip:2000@remote_domain.com>? Can * work as a regular SIP proxy in that aspect? Can * handle SIP URI:s that are complete SIP URI:s (sip:user@domain) instead of numbers only? Or should I run a SIP proxy on a different machine to handle pure SIP requests and let

I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes.

Hi! Are the GIPS codecs now implemented with the Asterisk? If I need more analog lines, say around 30, what's the easiest way doing it? I checked the Mediatrix box with 24 connections, maybe that would be a good (and rel. cheap) way to go? Any other suggestions? The ports has to support fax machines. rgds, /staffan --------------------------------------

I have found some references to the "hint" (or HINT?) variable and method in the source code, but quite a bit of Google-ing has not turned up any extensive answers as to some real-life examples of how to use this perhaps very useful tool. I understand the point of the tool, but I need to get some actual configs to look at before I think I'll figure it out. Even if my

Hi I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router with two interfaces. My local phones are situated behind the NAT and connects to the outer interface of the */FW/NAT/Router. * is then connected to my SIP providers (since I'm only using the SIP-part of *, PSTN connection through my SIP-provider). Works fine! rgds, /staffan kerker sweden -----Ursprungligt

Hi I know this is a bit off topic, but still pretty interesting. I'm running Asterisk on my Linux router/NAT/FW connected via cable (1mbit/200kbit) to the internet. Now, I wanna do local QoS implementation. Just very simple to give RTP (UDP) highest priority on my outbound interface. So, whenever I got an ongoing call, the RTP traffic should be handled first and other data (file transfers

Dear package mgcv users, I am using package mgcv to describe presence of a migratory bird species as a function of several variables, including year, day number (i.e. day-of-the-year), duration of survey, latitude and longitude. Thus, the "global model" is: global_model<-gam(present ~ as.factor(year) + s(dayno, k=5) + s(duration, k=5) + s(x, k=5) + s(y, k=5), family =

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

Hello List, I have been working on a PHP application in order to build a BLF style script. Until now everything is going Ok but something a little (in my oppinion) strange is going on with the 'ExtensionState' command; The problem is that it does not returns the 'Status' as it's suposed to, mentioned in the A.T.F.O.T book - version 2., where it sais something like:

Dear Sir, Would you be able to advise OpenSSH can support PKCS #11 & Microsoft Crypto API (MSCAPI). Rdgs Jennifer

Hi, wondering if anyone can help me with my problem. I can't get the verify.agi script to work in Asterisk This script is available for download at http://www.sineapps.com/downloads.php The agi script works for recording and playback when accessing it directly at it's extension, but will not record anything when doing the flashhook procedure during a call. Recording is cut off after

Can you then inform me on what structures this information is stored in, in the asterisk code? Then ill try to do a quick dirty version of the replication. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Kevin P. Fleming Sendt: 9. juni 2006 16:25 Til: Asterisk Users Mailing List - Non-Commercial

Hi, I am new to Rails and we are creating a web 2.0 app where we are integrating Rails and Flash. I dont know how to pass the values from flash to Rails to save in database. We have integrated the above specified with PHP and Rails. How to get the data from database and pass on the Flash thru Rails. Wht i exactly need is, We have 3 list box where when i select the first list box i need to

Hello, I''ve "emerged" Shorewall 2.0.7 onto my Gentoo pc. Going through the 2 interface quickstart guide I download the 2.0.1 interface sample and untar it. "tar -zxvf two-interfaces.tgz" Maybe a dumb question but I can''t find anything on Google or the Shorewall mail archives that say anything about this. So I''m assuming its me. :P But the

I have a problem with my Asterisk system. When I use my phone to call my office mailbox I have to end my password with #. (The office do not use Asterisk) " # " is also used as a transfer button on my asterisk, so when I press it I hear my Asterisk trying to transfer the call. Is there any way to change the transfer button or remove it ? Fredrik

Hi I am new here and am wondering if I have the correct list to subscibe to. I am looking for a user forum; technical mutual help/tutorial type list; would this be that type of thing? So far the messages I am seeing are mainly intercommunications between what appear to be developers working on assigned sub-projects of various flavors of samba. I don't want to spam a list with

I have the programable button led's working properly on my snom 200 except they don't flash during a ring event. I found a post by Andre Bierwirth saying he had a patch that he submitted but didn't make it into CVS. I would like to get a copy of that as a starting point to implement button flashing on ring. I have read through all the code and it looks like it should be pretty

Hello, I am wondering if there has been a solution to the following issue with R under Windows (see also thread to PR#6662 in this mailing list: http://tolstoy.newcastle.edu.au/R/devel/04a/0550.html ) I am using R for teaching students Statistics, so they are working with a university-wide installation of R. I have compiled an R-library which contains all the instructions and customised

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,