similar to: Two Lines, Two Businesses

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

I am setting up Xen for the first time and have run into a problem. My system was initially setup with Fedora Core 4. I was able to successfully create a File-backed VBD. When I boot the domain,I get an error message that is listed at the end of the below boot messages. I suspect the problem is casued by my fstab. However, I am not sure how it should be configured. Fstab listed also. Any pointers

I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp 0 788 0.0.0.0:5060 0.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A

Consider the following: > 20:28[1] 20 21 22 23 24 25 26 27 28> 20:29 [1] 20 21 22 23 24 25 26 27 28 29 It seems that when a vector has 10 elements, it prints out differently than one with 9 (extra space before the opening bracket). I can't see why this is happening. I am writing a manual containing many examples and the misalignment of the [1]'s is visually problematic. For

Appreciate any help - I am working through Agile web development (4.0) and am stuck on the deployment using phusion passenger. I get this: "Forbidden You don''t have permission to access / on this server." when I put depot.yourhost.com in the browser. I have been through various web searches and for file permissions and suggestions to change the <Files> container.

I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great !

Or you can use authenticate() and have it take its 'passwords' form a text file on your machine. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > trixter aka Bret McDanel > Sent: Sunday, January 29, 2006 5:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion

I have fixed this before, but I cannot for the life of me remember how I did it. I have a TDM400P with one fxo module and one fxs module. I setup Asterisk @Home and everything works fine, except when I try and call out. If I call out with a SIP phone it calls the zap extension and not the pstn line? If I take the zap extension offhook and call with the SIP phone it dials out the pstn line

Hi, I have a scenario that I am failing to implement using the Queue app, but which I had thought would be commonplace... 1) (this bit works fine) I want a queue caller to have access to the basic set of agents initially, with an overflow to additional agents if they are busy - This is done using penalty: queues.conf: member => SIP/dev1,0,Agent1 member => SIP/dev2,0,Agent2 member =>

I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code

>From the queues.com file. ; An announcement may be specified which is played for the member as ; soon as they answer a call, typically to indicate to them which queue ; this call should be answered as, so that agents or members who are ; listening to more than one queue can differentiated how they should ; engage the customer ; ;announce = queue-markq This allows you to have one

Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:

Are phone numbers portable in other countries? Are the same rules and conditions that exist here in the States mirrored elsewhere? How does a person in Europe go fully VoIP and still keep the main number? Do they use call forwarding? Is their another way to use an origination carrier without loosing your number? Alex -------------- next part -------------- An HTML

OK here it goes.. Caller ID is two parts or actually three: Part 1 Number only Part 2 Number + Name Part 3 Whole lotta stuff (also known as ADSI) Here is the US, I cannot speak for other countries. When party A places a call to Party B. Party A's Telco picks up the number, either from a table on the switch or passed from the PRI from Party A. Then on the far side (Party B's Telco)

Is there anyway to "spoof" a BLF signal to a phone from within a dialplan? That being so I can turn a BLF light on or off on a phone when a certain thing happens in a dialplan?

I have a couple of Samba 2.2 servers, in different locations, configured as NT Domain Controllers, but I'm experiencing some problems with logging on to the domains as Administrator in order to perform some administration tasks, such as configuring antivirus software on workstations. I won't go into the details here; I think the basic problem is my lack of understanding of how the

First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't

This is a very KGB / NSA / InterPOL / CIA type question, but if I have a recorded file (G.711, no compression) can I feed it into standard in of an application and have it recreate the fax that was send? I don't know enough about the Fax handshaking to understand this. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi how can i retrieve the call unique id of asterisk in the dialplan? I have enabled the cdr logging on a postgres database. In the table cdr each record has a field that assumes an unique id (for example: 1141628669.51) Can i retrieve this from the dialplan? For example: exten => 203,1,Answer exten => 203,2,Set(CALLERID(name)=UNIQUE_ID - ${var_name_unique_id}) exten =>