similar to: Only silence trying to play streaming MOH

Hi, I'm using scp from openssh-5.1p1 on opensuse 11.1. trying to copy a file to a nonexisting directory with scp gives the wrong error message: harald > scp /etc/passwd remote:nodir/ scp: nodir/: Is a directory harald > ssh remote ls -ld nodir ls: cannot access nodir: No such file or directory IMHO the error message should read e.g. "directory does

Dear All I have this setup [asterisk]----[mediant2000]-------E1 Trunk----------[Avaya] When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error *CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip

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

support full text searching you can download it from http://61.152.238.107/download/nodir.api.rubyonrails.zip (no directory list but better) http://61.152.238.107/download/api.rubyonrails.zip description is here if you can read chinese :) http://www.blogjava.net/cap/archive/2006/03/22/36796.html -- Posted via http://www.ruby-forum.com/.

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

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")

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

I've tried a number of solutions, but I've been unable to get Asterisk working with streaming MOH without running into the "buffer" issue. I've tried using various combinations madplay, mpg123, mpg321. I've also tried streamplayer by itself, and in combination with play-fifo ( http://www.freeswitch.org/asterisk_stuff/play-fifo.c ) to try and eliminate the issue. For

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10, 11.15.1, 12.8.1, and 13.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10, 11.15.1, 12.8.1, and 13.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these

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. > >

Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis. This is on a BT Business Highway line (which is essentially an ISDN2e line with two built-in

Hello everybody, from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in verbose and SIP logs. Could even be some weird intermittent firewall issue I guess. Apart from logging all traffic 24/7 via tcpdump (not really

I have been racking my brain attempting to get the remote callerid information for calls made to extensions on another Asterisk system connected via IAX2 but nothing has worked. To clarify, I would like to display the number AND name on the calling phone when calling extensions on another Asterisk system. I seem to be able to 'send' all the information I want to the system I am calling but

We are having a strange problem today. We have a SIP trunk from a provider and incoming calls are being dropped after the IVR when attempting to connect to any internal phone. If a you dial a DID that goes directly to a phone you can talk but the call will drop when you attempt a transfer. The trunk provider tells us that they are dropping the call because according to them we are

hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC]

Asterisk Project Security Advisory - AST-2015-002 Product Asterisk Summary Mitigation for libcURL HTTP request injection vulnerability Nature of Advisory HTTP request injection Susceptibility Remote

Asterisk Project Security Advisory - AST-2015-002 Product Asterisk Summary Mitigation for libcURL HTTP request injection vulnerability Nature of Advisory HTTP request injection Susceptibility Remote