similar to: asterisk-users Digest, Vol 51, Issue 51

Hi, 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, Dial(SIP/88888) exten => 88888,n,Hangup the C program under /usr/local/src/parallel/fire will wait for the input, if it's 1

I've encountered an odd situation with Asterisk 1.4.19 that I can't figure out. If I dial an extension via a Cisco AS5400 with the "g" option to come back, when I then Dial another extension after that, we don't get audio from the caller. There are no firewalls, no routers, no anything but a network switch between. The calls come in as SIP from the Cisco and

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

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.) > >

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 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

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

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

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

Hi All, I need help with integrating AVAYA 8300, the avaya can do outbound calls but cannot do inbound calls, im sending calls from sip to avaya using E1 ISDN line. My config was based on aspect dialer it's working with aspect but not with avaya. My config and error is below. zaptel.conf span=1,1,1,ccs,hdb3 bchan=1-15,17-31 dchan=16 zapata.conf group=0 context=avaya switchtype=euroisdn

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

On Tue, Jun 25, 2013 at 7:07 PM, Daniel P. Berrange <berrange@redhat.com>wrote: > On Tue, Jun 25, 2013 at 07:02:40PM +0200, Roland Giesler wrote: > > System: > > $ cat /etc/issue > > Ubuntu 12.10 \n \l > > > > $ uname -a > > Linux Matt-HP 3.5.0-34-generic #55-Ubuntu SMP Thu Jun 6 20:20:19 UTC 2013 > > i686 i686 i686 GNU/Linux > > > >

Hi all, we observed a strange effect when copying an file within a samba share: Both atime an mtime of the target file are set to the mtime of the original file. The atime of the original file is updated to the current time. 1. Status of the original file: # stat test.txt File: `test.txt' Size: 3 Blocks: 8 IO Block: 4096 regular file Device: 811h/2065d

I took one of the examples and tried to run against my database ls -l /data1/mail/db/cur.1 total 1129624 -rw-r--r-- 1 jwl jwl 0 2011-06-09 02:27 flintlock -rw-r--r-- 1 jwl jwl 28 2011-06-09 02:27 iamchert -rwxrwxrwx 1 jwl jwl 7258 2011-06-09 02:27 position.baseA -rwxrwxrwx 1 jwl jwl 7046 2011-06-09 02:27 position.baseB -rwxrwxrwx 1 jwl jwl 474226688 2011-06-09 02:28

Greetings! We have a somewhat noisy background in our call center, and I'd like to reduce this. Obviously, we could plaster the walls with sound absorbing material, but is there anything we can do in software either using any algorithms for our open source-based SIP library or inside Asterisk itself? Related to this, anyone have a good source for good panels? We are using