similar to: Early audio(progress) and MOH

Hi, I have read that DAHDI automagically turns of echo cansellation when it sees that it is a FAX. So I checked this out. I have a fax call into asterisk which is immediately called out to an external fax machine via DAHDI again.. For example, the result is: DAHDI/1-1 = incoming call, DAHDI/2-1 outgoing call. Now, with the help of dahdi show channel, if I check channel 2: echo cancellation is

Only a warning, and doesn't seem to do anything bad. But I can't seem to figure out what these warnings mean? -- Requested transfer capability: 0x00 - SPEECH [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: k [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: o [Feb 17

Hi, I'm now thinking of always dialing out to Local/xxx at outbound/n on all my queue members. The reason for this, is both to be able to limit the number of calls to one agent, and to have fail-over-lines on the agents. (for example, if dahdi fails, go sip) But for a few years ago, I did some testing with Local/ channels, and they seemed somewhat unstable in large quantity. Are they more

Hi, This is the output from queue show 28: 47 (DAHDI/g0/12345678) (realtime) (Ringing) has taken no calls yet Why is the devicestate "Ringing" when no channels is calling this number, and the queue says "has taken no calls yet"? Is it picking up the general state of a random channel on g0 in dahdi? Or what is happening? It only seems to happen with this particular

Hi, I'm having problems with CDR's and Queues in Asterisk 1.6.1. Heres three examples: Normal call: User A calls in to asterisk, gets a PlayFile, and hangs up. This gives 1 CDR as expected. Call to a Queue and then a playfile afterwards: User A calls into asterisk, goes into a queue, asterisk calls user B. When user B hangs up a CDR for User A is generated. (no CDR for user B)

System have been working great for weeks, using an average 40 of 120 dahdi channels. But today, I suddenly see scary things like this: -- Moving call from channel 5 to channel 7 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing

I want to create a script for IVR that compiles responses, aggregates them to a total number. Then, run an equation based on the result. Press 1 for X (X is a positive number 500) Press 2 for Y (Y is a positive number 200) Press 3 for Z (Z is a positive number 300) Press 20 to calculate the results = 500+200+300 =1000 then, exten => s,n,Read(NUMBER,,1000) exten => s,n,SayDigits(${NUMBER})

Hi, Is it possible to have early audio while waiting for answer in a Dial()? Say that I want to do this: 1,Progress() // Establish early audio possibillites 2,Dial(SIP/user,20,z(repeated-musicfile)) Where z would be like a function for playing early audio. Or z would just start MOH without Answer()'ing first. H?kon Nessj?en Loopback Systems AS -------------- next part

Hi, I've uploaded a new patch at https://issues.asterisk.org/view.php?id=16732which adds two new AMI commands, called "DeviceStateSet" and "DeviceStateGet". These commands let you update Custom device states, and read all devicestates from AMI. It would be very nice if someone could help me test this feature, and report back to the issue tracker. To test, log into AMI

Hi, I've just uploaded a patch here: https://issues.asterisk.org/view.php?id=16925 This patch introduces a new parameter; "congestion" to both RINGNOANSWER in queue_log and AgentRingNoAnswer AMI event, which is set to 1 if the call failed to go through because of technical difficulties. And it also is more verbose than app_queue has been earlier, since app_queue usually silently

I'm trying to setup my asterisk system for the least overhead as possible. My understanding (and experience with other systems) leads me to believe I can run any MOH using a certain class through a single 'player' as opposed to starting an independent stream for each MOH instance. However, try as I might, I can not get it to work. When I throw two calls into a queue, they are both

Just been getting lots of failed SIP registrations to a system here. All coming from Taiwan. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

It would be cool to get one of these and see if it can be hacked and loaded with your favorite SIP or IAX softphone. Looking at the pic, it looks like the dongle is both a soundcard and memory stick. Heck, I would be glad to have it if I could get the soundcard to work. Might as well since it is free after rebate. http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem

Hi, Does anyone know how to code newton's method for finding the roots of polynomial functions? im not sure whether i need to do this manually, or just code something with a loop to stop when it gets to the desired result thanks guys! _________________________________________________________________ Looking to move somewhere new this winter? Let ninemsn property help [[elided Hotmail

Hi, Does anyone know how to code newton's method for finding the roots of polynomial functions? im not sure whether i need to do this manually, or just code something with a loop to stop when it gets to the desired result thanks guys! _________________________________________________________________ Looking to move somewhere new this winter? Let ninemsn property help [[elided Hotmail

Hey guys, sorry im drawing a mental blank. i have coded the function: y <- function(x) 6*x^4-(1*x^2)^3 How do i evaluate this function for a given x? i know this is a ridiculous question, but thanks anyway _________________________________________________________________ [[elided Hotmail spam]] [[elided Hotmail spam]] =Domain_tagline&_m=EXT [[alternative HTML version

Testing on an 8-node 128-way NUMA machine has exposed a problem with Xen''s nr_cpus calculation. In this case, since Xen cuts off recognized CPUs at 32, the machine appears to have 16 CPUs on the first and second nodes and none on the remaining nodes. Given this asymmetry, the calculation of sockets_per_node (which is later used to calculate nr_cpus) is incorrect:

Den 12. feb. 2017 21:44, skrev ebroch at whitehorsetc.com: > > > Any opinion on dspam's interoperability with this? > > Just follow the wiki and replace sa-learn scripts with calling dspam. Dspam direct pipe needs mail-line-endings (\r\n) translated into unix line-endings (\r). like so: ----- learn-spam.sh --------- #!/bin/sh sed -e 's/\r$//' | /usr/bin/dspam

Hello, I just did a fresh svn install of 1.4 trunk everything. Everything compiles and installs just fine. When I get to asterisk-addons, I cannot select chan_mobile in menuselect. Chan_mobile is not even an option in menuselect for asterisk trunk. I tried the latest patch which failed in many places but did add an option for chan_mobile in menuselect for asterisk but it still cannot be

So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry