similar to: Changing caller ID on a Zap channel

For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

I was just browsing Digium's web site and noticed they are taking orders for the new IAXy. Has anyone purchased and tested one of these yet?? I have thought about buying one for testing, but want to make sure it isn't going to be a flop like the last one. Robert

I would like to setup a test number, that speaks back my phone number. How can I set this up? bye Ronald

Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference "FM" is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad

Hello All, I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy static. Even with the pots line disconnected, if I do a dial I still get static. This way I know it's not the line, but rather something on the card. I tried alternate pci slots. This card has a power connector, does anyone know what the power requirements are? The unit is in a small case with a 2.4ghz p-4 and

Hi, What do I need to do to get asterisk to NOT pickup a Zap channel when it rings? The channel in question is used for outbound calls only, and all incoming calls are answered by an analog phone elsewhere in the building that does not run through asterisk... so.. either make it not answer.. or make it delay for like 90 seconds.. I've tried wait's.. but it still seems to pickup the

Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com jeromy@voipempire.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/e668136a/attachment.htm

Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel.

Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine

Hi All, I am looking for a GSM VoIP gateway for use with Asterisk. I have come across VoiceBlue by 2N but it's price is beyond my reach. Are there any other alternatives out there? I've scanned across the mail achieves for an answer to this without much success, if the question has already been answered kindly point me to the resource. Allan.

I am running Asterisk@Home Voicemail works fine but does not email out the voicemail attachments. Any suggestion? ----------------------------------- Voicemail.conf [general] #include vm_general.inc #include vm_email.inc [default] 201 => {password},Jeff G Laptop,jrglass@columbus.rr.com,,attach=yes --------------------------------------------------------------------- Sip.Conf [201]

I have a data set like this I want to assign "outward" to Y if sc <90 and assign "inward" to Y if sc>=90. then cbind(p1982,Y) to get like these p aa as ms cur sc Y 1 154l_aa ARG 152.04 108.83 -0.1020140 92.10410 inward 2 154l_aa THR 15.86 28.32 0.2563560 103.67100 inward 3 154l_aa ASP 65.13 59.16 0.0312137 7.27311 outward 4 154l_aa CYS 57.20 49.85

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the

I wish to have inward-pointing ticks on my contourplot graph, but the colored background produced by the "region=TRUE" statement covers the ticks up, is there any way around this? Sample code below. --Seth library(lattice) model <- function(a,b,c,d,e, f, X1,X2) # provide model function for contour plot {J <- a + (b*X1) + (c*X2) + (d*X1*X2) + e*(X1^2) + f*(X2^2) pp

I have a full data set like this: aa bas aas bms ams bcu acu omega y 1 ALA 0 127.71 0 69.99 0 -0.2498560 79.91470 outward 2 PRO 0 68.55 0 55.44 0 -0.0949008 76.60380 outward 3 ALA 0 52.72 0 47.82 0 -0.0396550 52.19970 outward 4 PHE 0 22.62 0 31.21 0 0.1270330 169.52500 inward 5 SER 0 71.32 0 52.84 0 -0.1312380 7.47528 outward 6

I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03

I'm attempting to set up call recording with Asterisk. Using automon => *1 ; One Touch Record in features.conf does not appear to be working. I'm using Polycom 501's but when someone dials *1 while in a call, nothing happens. I'm wondering if the phone or Asterisk is even detecting the DTMF. I suspect that is the problem but don't know how to verify or

So I decided, for the formal asterisk rollout, to change over to less commercially-infringing MOH than the prior admin had thrown on the server. (plus: it was blown out and nasty sounding over the phones. Ew.) I changed the files in /var/lib/asterisk/mohmp3 to something else (can't dig up the link, but it was from the voip-info wiki). My musiconhold.conf looks like this: ; ; Music on

I have tried to get MRTG to graph my Asterisk box but have run into a problem. When I run the perl script provided at: http://karlsbakk.net/asterisk/ I get the following error: [root@tsr asterisk]# ./asterisk-mrtg -h myasteriskip.mydomain.com<http://myasteriskip.mydomain.com>-v -1 SIP -2 IAX2 -u 109 -p xxxx Asterisk Call Manager/1.0 Action: Login Username: 109 Secret: xxxx Response: