similar to: Identifying which call an event belongs to

An application commanding asterisk with AMI is going to launch lots of concurrent calls in very few seconds using the Originate AMI command but it's also going to need to be able to cancel very quickly any call of them even before each OriginateResponse event comes in. All the calls will be done by the same trunk (a trunking enabled channel). But there's a problem for canceling any call:

Hi all, When the 'Originate' command is issued with 'Async' open set to 'yes', I got the response right away with the correct 'ActionID'. What follows is the 'Newchannel' event with a 'Channel' ID, but their is no 'ActionID' to tie it back to the command. How do you guys deal with this? -------------- next part -------------- An HTML

Hi Guys, i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1) and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue i'm having is that i'm able to receive faxes from a website (that offer this service) but not able to receive from a regular fax machine (that is working perfect). [fax-rx] exten => receive,1,NoOp(**** FAX RECEIVE ****) exten

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

Hi Guys, i spent some time to figure this out (since i love how dialplan is written) but i decided to ask for your help guys. i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in pbx2 extensions.conf: i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr) in pbx1, i

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, I have asked this question months ago - i have "toggled down" all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like "A"

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

Hi Guys, i am using the following config in pbx1: register => pbx1:endopass at 172.16.200.175 <pbx1%3Aendopass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=password context=[default] deny=0.0.0.0/0.0.0.0 permit=172.16.200.175/255.255.255.128 in pbx2: register => pbx2:endopass at 172.16.200.176 <pbx2%3Aendopass at 172.16.200.176> [pbx1] type=friend

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

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

See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Thanks. -- James <--- SIP read from zzz.zzz.zzz.44:9363 ---> NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: "xxx-xxx-xxxx" <sip:xxxxxxxxxx at pbx1.mydomain.com>;tag=467525dd6fac949do0^M To:

From: Haiyang Zhang <haiyangz at microsoft.com> Subject: 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 atomic counter to ensure all channels are ready before vmbus_init() returns. So another module won't have any

From: Haiyang Zhang <haiyangz at microsoft.com> Subject: 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 atomic counter to ensure all channels are ready before vmbus_init() returns. So another module won't have any

Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid

Hi, I am trying to understand why some of my call transfers fail. My scenario is as follows: Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2 Step1: PBX1 extension 101 calls PBX2 extension 102 Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension 103 Step3: PBX1 extension 103 answers the call and transfers it to PBX2 extension 104 Step3 fails and extension 103