similar to: One Way Audio After Dial

I've been searching to a solution to this for a while and can't figure it out, perhaps someone has done something similar. I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low on my lightly loaded switched gigabit ethernet network. One Asterisk uses Zaptel and a Digium card, and DTMF recognition

Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to 1111 when I go

On Oct 16, 2008, at 2:36 AM, asterisk-users-request at lists.digium.com wrote: > I want to call an extension like 88888 and invoke an external C > program upon > calling, pass an constant integer like 1 to the C program. > > What I have done is: > > /etc/extensions.conf: > exten => 88888,1,system(/usr/local/src/parallel/fire 1) > exten => 88888,n,

On Mar 19, 2008, at 1:00 PM, asterisk-users-request at lists.digium.com wrote: > Am I expecting too much? Perhaps. I think the hardware on which we run Asterisk can be much more reliable than the software, which is often the case. We have a bunch of HP servers with RAID and have never lost anything. A HD may fail, but the RAID keeps it going until we pop a new drive in there. A

All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy "fixed" the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) After doing this, I recompiled ztdummy and it worked. Note

>> All too common and largely undocumented. I had this same problem. >> >> Installing ztdummy changes Asterisk to use it for timing of playback, >> apparently. Removing ztdummy "fixed" the problem. To get it all to >> work, I had to upgrade to to at least kernel 2.6.23.11 (previous >> versions are either missing options are just broken.) > >

On Mar 5, 2008, at 5:46 PM, asterisk-users-request at lists.digium.com wrote: > If you are running a call centre (large or small) using Asterisk, > I'd be > interested to know how you log your agents in & out: > > E.g. > > - Do you use AgentLogin (to force calls onto the agents, perhaps)? > - Do you still use AgentCallbackLogin? > - If you use

Note that the newish SPA962 has 6 appearances and a color screen. I've noticed that the bright color screen does impress people when they first see it. PoE is also very nice and web provisioning was quite easy. I've yet to try a more automated provisioning method on it. I know that getting the polycom's to auto provision wasn't very straight forward. I do provision some

What does it take to get ztdummy to work correctly? I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 1.4.9.2 Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs. Problem is playback() does not work. So then I stop zaptel, asterisk runs and playback() now works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for that. I am

I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. We'll be using queues (modified), which precludes some of the standard redundancy solutions, since the queue needs to know all the agents

Is anyone aware of a service where we can lookup phone numbers to determine a name and/or name + address available in bulk? We want to look up every number called to our call center, so it will be tens of thousands per day. Services that charge 3 to 5 cents per lookup will get way too expensive very quickly. Thus, I'm looking for a service that can either license a database or

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

I'm trying to get Asterisk to pass thru calls using the G729 codec. I've got a 7960 phone and my gateway is an AS5400. I got the following messages when debugging SIP (7778881000 is the 7960): WARNING[1872]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/7778881000-2874(4) to SIP/as5400-35c1(256) WARNING[1872]: app_dial.c:1002 dial_exec: Had to drop call

Hello All, We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a few Grandstream GXP2000 and a handful of Handytone 486 units. The

I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping R&D costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of

We've got this configuration : Cisco as5400 --- asterisk main server ---- asterisk for cells ---- gsm gateway cisco and the gsm gateway are connected to asterisk via sip, the two asterisk servers are connected via iax. On a succesful call the cisco (not always, 60% of the times) will keep sending a ringtone to the connected phone, even if the call is answered, actually if the user behind

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

How does one get asterisk to timeout realtime request via res_odbc to unixODBC? I've set timeouts as appropriate for freetds (which unixODBC is using.) However, it doesn't seem to work. It takes over 3 minutes to timeout a connection and queries never seem to timeout, so a channel waiting on a query never terminates. I did notice that res_odbc.c never sets a timeout on the query

i want to use Cisco AS5400 media Gateway as my PSTN Gateway instead of using the E1 PCI cards in asterisk box ,is this practically possible? can i use SIP in the connection between Asterisk and Cisco AS 5400 Gateway? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE!