similar to: multiple pbxes, multiple domains, same user ids?

Hello ppl, Any way to specify alphabetical exten patterns in the dialplans on Asterisk? All my users would have alpha/numerical ids. I don't want to add a line for every user in my dialplans. I searched around, but couldn't get anything useful. Any way to get around this? Thanks in advance - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are

Hello All, Is CALLERID() setting broken in 1.4.4? My small dialplan : [testclid] exten => _0.,1,Set(CALLERID(all)=Ben Jacob <988077>) exten => _0.,n,Dial(SIP/${EXTEN}) Correct me if I am wrong, Set(CALLERID(all) above supposed to change the display name as above(Ben Jacob) and change the From URI to 988077 at myip?? As of now, only the _display name_ is being replaced, but not the

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

Hello All, I am using ODBC storage for voicemail on my asterisk box. I want to have a common voicemail box for different extensions. I know how to do that, but the question troubling me is how and where do I store the the extension name for which a particular voicemail was left. e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 55555. Now, when someone calls 1000, and leaves a

Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console=> notice,error ;messages => notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential

Hello ppl, Have implemented a really nice Billing engine using AGI scripts. So far it works fine, tho haven't yet put it in the torture cell. The AGI scripts have been written in PHP, using MySQL for the billing and profile information. The major disadvantages I see using AGI scripts : 1. A new process(invocation of PHP scripts) on every new call. 2. MySQL connections on every instance of

if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand -------------- next part -------------- An HTML attachment was scrubbed... URL:

Test ------Mensaje original------ De: tracinet Remitente:asterisk-users-bounces at lists.digium.com Para:Asterisk Users Mailing List - Non-Commercial Discussion Responder a:Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP Enviado: 6 Mar, 2009 5:55 PM Basically, Server 1 is the main customer PBX where we have multiple

Is this getting through?? EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions

Hello All, I've got the following warning messages a couple of days back: /chan_sip.c: Maximum retries exceeded on transmission <SIPcallId> for seqno 1 (Critical Response). /Have got the warnings repeatedly for one Callid. If maximum retries have exceeded why should it give me those warnings again n again for the same callid, with a gap 4 seconds between each warning. The callids

Hello ppl, Sorry to re-post it, but kinda these issues are getting on my nerves. I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on 1.4.4. The problem : 1. I receive call from caller 'AAA' on my number, 'BBB' which is on my Asterisk box. 2. I have to redirect the call to some other number, say, my cell num - 'CCC'. 3. My PSTN provider wants the

Hello ppl, Searched all over, but couldn't find anything conclusive. Does an off-the-shelf version of Asterisk run without any issues on a 64-bit machine? Does anyone have any 'conclusive' figures? Apologies if this is a repeat question. Would appreciate if I could be redirected to the appropriate link. cheerz - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it

Hi, I'm working on a 500 seats Asterisk project. I'm wondering whether or not I should consider using Asterisk Realtime and a database to manage phones registrations. Stories in Dev mailing list say Realtime is mis-used or should be improved. So, what's the bottom line ? Can I consider anything I can do with .conf files can be done with a combination of .conf files and Realtime.

Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in

I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like

i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two

Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK,

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hi is there any ritual sacrifice a newbie has to perform to be welcome on this list? I am new to this whole PBX thing in general and Asterisk in particular. I had hoped that the community on this list would welcome a newbie like myself and help me with some answers to my stupid questions, but somehow it seems to me that nobody likes to respond to somebody who appears to be a complete

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the