similar to: Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

Hi list, in asterisk 1.4 and maybe earlier it was possible to use voicemail system with mailboxes starting with some special characters like *. The line in voicemail.conf was like this: *123 => , AB,,,tz=cet|attach=no| Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4. In Asterisk 1.8 the above scenario is not working any more. The Voicemail application reports an

Hi I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI Already installed). according to https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Asterisk 1.8 include this application but I cannot see it with "core show applications" Do I need to install mISDN or other modules for using that ? Regards M.Shirazi

Karsten Wemheuer <kwem at gmx.de> schrieb: Hi Karsten! > the timeout value of 15 minutes directs me to an issue with session > timer. Try to refuse them by putting the line > session-timers = refuse > into the general context of sip.conf. Reload the sip stack with "sip > reload". Sorry, I forgot to mention that... I already have this setting:

Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

Hi, I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN card. The system starts without any errors. I discovered a severe issue. The kernel panics on a very small load. The first call normally gets through. If I start the second or third call

Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Description Handshake Failure). The following line appears in the log: ast_iostream_start_tls: Problem

Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but

Hi list! My Problem: all calls to international numbers will be dropped after exactly 15 minutes... I have a VoIP-account by Deutsche Telekom. This is what I see when I call someone (my parents) and the connection will be dropped: == Using SIP RTP CoS mark 5 -- Executing [+39015222222 at default:1] Set("SIP/00493511111111-00000125", "newNumber=0039015222222") in new

Hi, while testing current release candidate 1.8.0-rc2 I stumbled on a weird behavior. I did not find any hints in the archives or at the bug tracker. Two SIP-Clients are connected (both on the local net, no NAT). The RTP stream flows directly between the phones. If I set phone A on hold, the music on hold is played. On the CLI I see the following message running: WARNING[2470]:

From: Haiyang Zhang <haiyangz at microsoft.com> Rename camel case variables in channel.c Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c | 733 +++++++++++++++++++++--------------------- 1 files changed, 370 insertions(+), 363 deletions(-) diff --git

From: Haiyang Zhang <haiyangz at microsoft.com> Rename camel case variables in channel.c Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c | 733 +++++++++++++++++++++--------------------- 1 files changed, 370 insertions(+), 363 deletions(-) diff --git

Rename camel case variables in channel.c and changed them to lowercase. Sending this from my own accounts till we have a proper mail server set up that allows us to send out patches without issues. This patch was created by Haiyang Zhang. Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> ---

Rename camel case variables in channel.c and changed them to lowercase. Sending this from my own accounts till we have a proper mail server set up that allows us to send out patches without issues. This patch was created by Haiyang Zhang. Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> ---

Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don?t in the initialization... I have in my file that comands: touch /var/lock/subsys/local modprobe zaptel modprobe wcfxo safe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening Somebody

From: Haiyang Zhang <haiyangz at microsoft.com> Subject: [PATCH] staging: hv: Fix race condition on IC channel initialization There is a possible race condition when hv_utils starts to load immediately after hv_vmbus is loading - null pointer error could happen. This patch added an event waiting to ensure all channels are ready before vmbus_init() returns. So another module won't have

From: Haiyang Zhang <haiyangz at microsoft.com> Subject: [PATCH] staging: hv: Fix race condition on IC channel initialization There is a possible race condition when hv_utils starts to load immediately after hv_vmbus is loading - null pointer error could happen. This patch added an event waiting to ensure all channels are ready before vmbus_init() returns. So another module won't have

From: Haiyang Zhang <haiyangz at microsoft.com> Subject: [PATCH 1/2] staging: hv: Fix race condition in hv_utils module initialization. There is a possible race condition when hv_utils starts to load immediately after hv_vmbus is loading - null pointer error could happen. This patch added an atomic counter to ensure the hv_utils module initialization happens after all vmbus IC channels are