similar to: four ringing and hangup with error

Dear ALL I have audiocode MP -124 with configure in asterisk Endpoint configuration means every analog phone register in asterisk now thing is that i have one more SIP trunk with mediant 2000 [auodiocode-mp-124]-----[ * ]------[mediant 2000]-----E1 When i call from audiocode MP -124 phone i got this error -- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1")

Hi, I've solved the flash transfer problem changing the flash time in the zapata.conf file, I've set: flash = 200 (the defualt was 750 ms) in the extensions.conf the code is for example: exten => 42,1,Flash() exten => 42,2,SendDTMF(42,250) exten => 42,3,Hangup() now the transfer with flash works correctly. About the question whether my PBX expects a hook flash for

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR: TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk >From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can

I've found some manuals and tried this to do : Sip.conf [test] type=friend username=test1 secret=test1 host=192.168.1.238 context=tutorial fromuser=SIP Phone callerid=101 nat=no canreinvite=yes dtfmode=info disallow=all allow=ulaw [test] type=friend username=test secret=test host=192.168.1.240 context=tutorial callerid=100 nat=no canreinvite=yes dtfmode=info

Hello R-World, i am trying to calculate data for a DEM (Digital Elavation Model) which i also want to plot in R. i have the coordinates for the lower left corner which look something like this: x<-42,2 y<-50,5 besides i have the cellsize of the grid, which is: z<-1,1 what i do is to calculate the corrdinates of the cells to the right and top, what i can do by specifying the number of

I am having trouble getting streaming MOH to work. As far as I can tell I have everything configured properly but there is only silence. Your help is appreciated. I am running Asterisk 1.8.11-cert10 with mpg123 1.12.1 to play the stream (I have tried madplay, and mpg321, and I compiled streamplayer as well with the same results). I started by finding a working stream and tested this from the shell

I have a setup with lot's of small files (Maildir), in 4 different volumes and for some reason the volumes are full when they reach 60% usage (as reported by df ). This was ofcourse a bit of a supprise for me .. lots of failed writes, bounced messages and very angry customers. Has anybody on this list seen this before (not the angry customers ;-) ? Regards, =paulv # echo "ls

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Remove some pointless conditionals before kfree(). Signed-off-by: Wei Yongjun <yjwei at cn.fujitsu.com> --- fs/ocfs2/alloc.c | 3 +-- fs/ocfs2/cluster/heartbeat.c | 6 ++---- fs/ocfs2/cluster/tcp.c | 6 ++---- fs/ocfs2/dlm/dlmdomain.c | 3 +-- fs/ocfs2/dlm/dlmrecovery.c | 6 ++---- fs/ocfs2/extent_map.c | 3 +-- fs/ocfs2/journal.c

Hi all ,? I have managed to install and configure the? 1. asterisk-1.8-current 2. dahdi-linux-complete-current I did not faced any issues during the installation. After that I installed X-Lite soft phone in two different PCs and tested the setup. every thing was success. I was able make calls from each?extensions. But when I observe the log files , i could see some messages ......

Hi all, I have the following problem, working with a SIP provider, if i setup my SJPhone to register directly to their STUN server and working over a 384/128 ADSL i have a really good quality, but then if i configure Asterisk to register to the same provider over the same 384/128 circuit the quality is REALLY BAD. The obvious difference is that using directly the SJPhone i am using STUN, while

Hi all, I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result. Here is the problem: I have a peer -which is peer AND user- setted up like this [MyPeer] ; type=peer host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142 permit=yyy.yyy.yyy.yyy/255.255.255.255 context=from-MyPeer dtfmode=auto disallow=all allow=ulaw,alaw

Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

Hello, I have a question . dose asterisk supports "302 Redirecting..." ? I have SIP Server "Not Asterisk" and my Asterisk is registering as a client for this device . when i try to call another client registered to the same SIP server i got Busy Tone and here is the asterisk CLI output ----------------- -- Got SIP response 302 "Redirecting..." back

On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >

Hello, I am having trouble passing DTMF digits from a Polycom 330 SIP phone to my FXS/FXO lines. I am running Asterisk 1.4.21.1 In sip.conf I configured dtmfmode=inband. RTP traffic (voice) goes perfectly from SIP to FXS, but in the SIP phone I only hear a continuous noise. However, when I press any digit in the pone (FXS) I hear the DTMF tone fine in the SIP phone (the noise goes away for as

Hello all, I have a population of 112 servers that are experiencing different levels of packet loss. I don't want to poll all 112 of them (the analytical tools must be manually run on each individually) so it seems best to sample among them; then I plan on using R to run comparisons of the data pulled from each one. I'm not clear on the most sound way to go about this and I don't