[asterisk-users] realtime static config include contexts

Hello ppl, Any idea how do I write in include lines(for contexts or include files) in the database, in the ARA static config? thanks in advance. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCR

Re: [asterisk-users] Asterisk Realtime Arch - static or realtime?

Rushowr wrote: Benjamin Jacob wrote: Hello ppl, Wanted to know your experiences, if you've worked with Asterisk Realtime Architecture. Which one do you prefer, static or realtime? I personaly think, the static architecture is a better solution, cuz, in the realtime config, to check the dial

Re: [asterisk-users] TE411P or TE412P?

Tony, the VPM450 is far better than the TE411P's VPM400. One main thing is that is has full 128ms tails on all spans whereas the VPM400 shared 128ms as you used more spans, and second is the Octasic chip makes the sound real crisp and clear. For the moment, if you need FAX tone detection, you will

[asterisk-users] MS LCS 2005 / SER / Asterisk Integration

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

[asterisk-users] Handling incoming calls from VoIPbuster

Hi all,Currently i've made some tests with VoIP buster and everything is running ok for outbound calls.Now i've created new VoIPbuster account, and my goal is to allow VoIPbuster partners to dial into my dialplan IVRs for free. But I always get my VoIPbuster account (currently registred with my ast

Re: [asterisk-users] modifying the INVITE headers

On 09/11/06 18:36 Paco Brufal said the following: Hello, Here in Spain there is a VoIP provider (Telefonica) that only works if when you make an outgoing call, the SIP headers are like this: INVITE sip:@telefonica.net SIP/2.0 But Asterisk is sending this: INVITE sip:@sbc.ngn

[asterisk-users] SIP trunk

hello If I want to use asterisk to hookup to a SIP account I just use the "register" line in sip.conf with the extension number at the end... But how about if I want to use a SIP trunk from a provider which gives me 10 DID numbers with the same account? thanx in advance rick ___

Re: [asterisk-users] Problems with outgoing calls

On 11 Sep 2006, at 10:29, Roy Gardner wrote: Hi, Our setup is: Asterisk 1.0.7 running on Debian 2.4.27-2-386 TE110P card ISDN 30 (UK E1 PRI) When making outgoing calls to the PSTN using call files I get the following problems: 1. No hangup detection - have to wait for time-out 2. No pick

[asterisk-users] Ringtones

Hi All I use Grandsteam GXP2000 phones. Is there anyway within the dialplan/indications etc to have a custom ringtone based on who is calling the phone. i.e if i have a call from an internal user i get one ringtone if its an external call i get a different ringtone?? Many Thanks in Advance

[asterisk-users] Can Asterisk bind on multiple ports?

Can Asterisk bind on multiple ports? I wish I could in my sip.conf make Asterisk bind different ports per different context, so that calls coming in udp port 5060 would fall in one context and calls coming in port 5061 fall in other different context. Is that possible? How can I edit my sip.con

Re: [asterisk-users] Max Size of Conf Files

Think of BerkeleyDB as a barebones embedded RDBMS. It carries the much the  same functionality (such as ACID) as most Relational DBs but without overhead (such as SQL translation) - for many years it formed (possibly still forms - not sure of present) the core of MySQL.Steve Totaro <[EMAIL PROTECTE

[asterisk-users] modifying the INVITE headers

Hello, Here in Spain there is a VoIP provider (Telefonica) that only works if when you make an outgoing call, the SIP headers are like this: INVITE sip:@telefonica.net SIP/2.0 But Asterisk is sending this: INVITE sip:@sbc.ngn.rima-tde.net SIP/2.0 because "sbc.ngn.rima-t

[asterisk-users] I am not getting 302 redirects...

Hi, How do the 302 redirects work in asterisk, and what is the "promiscredir" directive doing? I am not getting the documentation on this. I have following happening on my asterisk box: -- Executing Dial("mISDN/1-1", "SIP/[EMAIL PROTECTED]||Tt") in ne

Re: [asterisk-users] Asterisk Realtime Arch - static or realtime?

Benjamin Jacob wrote: > Hello ppl, > Wanted to know your experiences, if you've worked with Asterisk Realtime > Architecture. > > Which one do you prefer, static or realtime? > I personaly think, the static architecture is a better solution, cuz, in > the realtime config, to check the dialplan(n h

Re: [asterisk-users] QUINTUM TENOR ASM200 Configuration

FRANCISCO PEREZ-LANDAETA wrote: Hi, this message is for Steve. Sorry for replying to the digest. It wasn't my intention. I would appreciate if you can guide as to how make the tenor asm200 work with asterisk. I am using asterisk at home. I guess my problem is configuring the tenor so that it is

Re: [asterisk-users] Call Forward Problem

use:exten => s,2,Read(fwdnum|audiofile-to-play|10)then you'll have the number entered in variable ${fwdnum}.Type "show application read" in asterisk console for further details... Hope this helps...2006/9/9, James Williams <[EMAIL PROTECTED]>: I'm currently trying to write a section into my dialpla

Re: [asterisk-users] Max Size of Conf Files

I guess I will give it a try. The numbers are pretty much static in the way they are routed. I just did not know if Asterisk would choke on a conf file with a couple thousand lines. Thanks, Steve picciuX wrote: don't know for conf size limitation (but i guess it won't be a problem with a w

[asterisk-users] TE411P or TE412P?

I believe the TE412P is intended to supersede the TE411P. What, if any, are the advantages of the TE412P over the TE411P? Apart from the disadvantage that the hardware DTMF detection doesn't work on the TE412P (will that be fixed in the future?) My UK supplier can supply both cards, but the TE412P

[Asterisk-Users] SIP hardphones and BLF monitoring keys

Hi,For a small call center, we would like to change default behaviour.Current setup and behaviour are :- a bristuffed 1.2.10 Asterisk server with 4 BRI ports- 5 SIP hardphones (Snom 320) with BLF for line or extension monitoring - incoming calls ring all phones and light BLF on- when a call is comi

[asterisk-users] iax2 warning!

I always got this warning after I'm using IAX2 channels . Sep 11 21:44:09 WARNING[30229]: chan_iax2.c:6536 socket_read: Received trunked frame before first full voice frame What's it mean?___ --Bandwidth and Colocation provided by Easynews.com -- as

[asterisk-users] Outgoing callerid in AMI

Hello I have a problem with callerid in the manager interface. I think that Asterisk has a strange way to handle callerid, until I found out to set the o-switch in the DIAL statement, it did not work the way I wanted, it still doesnt, but now it works ok in one direction. My extension is 311, i

Re: [asterisk-users] How to integrate freepbx with a2billing?

Sharon Lim wrote: Hi all, I have tried to install freepbx and a2billing application. Now see both application is not integrated special on cdr part. Any idea how to integrated it?Confuse! -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *

Re: [asterisk-users] QUINTUM TENOR ASM200 Configuration

You missed my point completely. His "original" post was a reply was hijacking a very long thread (the digest thread) that he did not trim. Just trying to teach some netiquette so he will get more help. I have found that top vs bottom posting is not a major issue to most, but few make a big

Re: [asterisk-users] Max Size of Conf Files

don't know for conf size limitation (but i guess it won't be a problem with a well-sized machine).About asking on the fly vs writing on change: if your "routing information" varies very often, on the fly should make more sense. Otherwise, it's not useful to retrieve continually same data: better to

[asterisk-users] Problems with outgoing calls

Hi,Our setup is:Asterisk 1.0.7 running on Debian 2.4.27-2-386TE110P cardISDN 30 (UK E1 PRI)When making outgoing calls to the PSTN using call files I get the following problems:1. No hangup detection - have to wait for time-out2. No pickup detection - the dial-plan starts as soon as the line ringsAl

[Asterisk-Users] SIP parameter to prevent a call from being added in missed calls logs

Hi,If you set Asterisk to ring several extensions for an incoming call, it appears that the call will be added in every phone's missed calls logs though the call was picked by one extension.In the long run, this prevent users from using missed calls features as these logs would filled with many cal

[asterisk-users] Asterisk Realtime Arch - static or realtime?

Hello ppl, Wanted to know your experiences, if you've worked with Asterisk Realtime Architecture. Which one do you prefer, static or realtime? I personaly think, the static architecture is a better solution, cuz, in the realtime config, to check the dialplan(n hence the sql database) for each

Re: [asterisk-users] Zaptel-1.2.9 compile error

Rich Adamson wrote: That's strange; how many people just responded with "that worked"? None that I've seen! If they did, then they started with a subversion directory, or they were responding to a different situation. I suggest you start with a clean tarball and try it yourself. Or look in the

Re: [asterisk-users] Context

I have two contexts how could I isolate context A from context B ,in other words I want to ban context A from calling context B In sip.conf, define phones/extensions something like this: [1000] type=friend context=cust-a [1001] type=friend context=cust-a [2000] type=friend context=cust-b [

Re: [asterisk-users] Call Processing Slow 11 seconds

You could disable dialing altogether unless they press hash - that way they would learn about the hash key feature pretty quickly :-)   Unfortunately I dont see an easy solution since a dialplan covering all possibilities may be too complicated.   Cheers   Gerry     - Original Message

[asterisk-users] How to integrate freepbx with a2billing?

Hi all, I have tried to install freepbx and a2billing application. Now see both application is not integrated special on cdr part. Any idea how to integrated it?Confuse!-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *

[asterisk-users] Context

Dear   I have two contexts how could I isolate context A from context B ,in other words I want to ban  context A from calling context B   Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplor

Re: [asterisk-users] beginners question....

http://www.voip-info.org/wiki/ here got alots of example but you need to find it. You can start with http://www.trixbox.org/ that install everything. Good luck! On 9/11/06, Panagiotis Zikos <[EMAIL PROTECTED]> wrote: Hi all,   I am new in the asterisk company. I need to set up a small voip syste

[asterisk-users] beginners question....

Hi all,   I am new in the asterisk company. I need to set up a small voip system for about 60 phones ( a small enterprise organization). The system must support voip calls (calls inside the enterprise) but must be able to send calls over isdn (24 channels).   Thus the asterisk server must oper

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