similar to: Call Pick Up

I''m having troubles with my outbound VOIP connection. I''m convinced that I don''t have QOS/traffic shaping configured properly in my shorewall linux firewall, which serves as my Asterisk VOIP server and Internet router/gateway. I don''t have a separate router box. I''ve been using VOIP for about a year now, but just recently realized that I need to

Many times the "speed" of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A "speed" change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. Thanks, Jim

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

I released app_changrab.c lastnight really late... It includes a way to hijack a channel and originate calls from the CLI. /b --- Keep Your Friends Close, But Your Enemies Even Closer...

I've upgraded to asterisk-1.6.0-beta2. I'm trying to get the new Telco MWI detection function working. It doesn't appear to be working. I have this in zapata.conf ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mwimonitor=yes ;mwilevel=512 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signalling=fxs_ks context=incoming

How do I get a context to continue to execute commands after the caller hangs up after a Dial command? I'm using the "e" option to the Dial application. I though the "e" option would allow the context to continue. This doesn't want to work for me. I'm using asterisk-1.6.beta5 I never get to "3" below. I get a message saying the "2" ended

I'm struggling to get my VOIP installation to be acceptable. I'm looking for advice on what else I can look for. My system: o Teliax VOIP service, voip-ny1 proxy o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with

Formerly on a thread called [asterisk-dev] Where to find the code of application Bridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: >> Can someone please tell me in which file the code for the application to >> be found? I was not able to find a file named app_bridge.c in the folder >> apps. > > app_bridge.c ? app_confbridge.c ?

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

I configured my sla.conf to use with a Polycom phone. I have no idea if I did it right, however, none of the console "sla" commands exist. Do I have to something special to compile in this support, or should it just work out of the box? ~jay

Il 02/08/20 18:54, Stephen John Smoogen ha scritto: > On a side note, you keep emphasizing you aren't expecting an SLA.. but > all your questions are what someone asks to have in a defined SLA. I > have done the same thing in the past when things have gone badly, but > couching it in 'I am not asking' just makes the people being asked > grumpy. Better to be open and say

Hello! I have developed an IVR using AGI and so far it works great. I'm using Cepstral voices, but now want to use the voices from AT & T that are on a Windows server to be heard best. With cepstral what I do is to generate audio files from shipping and this text I reproduce this method it has worked very well. Now, try to do the same by creating the audio file in windows with the

I would be incredibly grateful to anyone who'll help me translate some SAS code into R code. Say for example that I have a dataset named "dat1" that includes five variables: wshed, site, species, bda, and sla. I can calculate with the following SAS code the mean, CV, se, and number of observations of "bda" and "sla" for each combination of

Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly

Hello, I am looking for a "real life" example of using SLA lines under Asterisk. I'll describe my environment and would like to know how I define it in Asterisk (version 1.4 final). Suppose I have two multi lines phones. The first phone has extension 1 assigned to it, and the second phone has extension 2 assigned to it. Now, I want extension 3 to be available on both phones as

I understand that the Asterisk SLA implementation is somewhat different from most key systems and PBX systems. I also understand that in Asterisk, one does not put an SLA line on hold since it is just a MeetMe conference. However, is there any way to make the BLF flash when the answering party on the Asterisk system presses the hold key on their set and leaves the calling party alone in the

If SLA supports IP trunks, can shared extension appearance be achieved using a local SIP trunk in place of an extension? Basically, I'm trying to allow some stations (Polycom IP 650) to have a shared extension amongst all of them. Ideally, I'd like for the LED to show if that extension is in use, and I'd like for the extension to ring all stations on that extension when a call comes

Can I get some recommendations: We are looking for a hosting provider (other than godaddy) with 1: SLA 2: SSH access 3: subversion/rsync or the ability to install binaries / compile source. Would like them to include http/https and email. Any suggestions? -Jason -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- -

With kernel 2.6.30-rc2-git6 and prior I am having problems mounting ext4 partitions after reboot. A successful mount looks like this: /dev/cciss/c0d0p8 on /squid-cache0 type ext4 (rw,noexec,nodev,noatime,data=writeback,errors=panic) /dev/cciss/c0d0p9 on /squid-cache1 type ext4 (rw,noexec,nodev,noatime,data=writeback,errors=panic) /dev/cciss/c0d0p10 on /squid-data type ext4

Has anyone done SLA with Polycom phones? I've got a large project coming up where the customer is keen on SLA for trunks and extensions. Trunks will be on a PRI. We may do this with Cisco phones if they work better. Mark Willis