similar to: AMI interface problem

Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel variables when I originate a call via AMI. All the documentation I have found says to do it like this: Variable: variable1=value|variable2=value|variable3=value However when I do this it runs them all together and I end up with:

Hi all! Am I missing some configuration or is it simply a bug: If Asterisk chan_sip is configured with bindport=5062, the port is missing on the outgoing SIP messages contact header. This resulting in in-dialog messages sent to port 5060 ... where there is no Asterisk on that host... Tried externip = 1.2.3.4:5062 with no success. Version 1.6.0.3. br Walter -------------- next part

Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in

Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c -> manager.o manager.c: In function ?action_getvar?: manager.c:1732: error: ?SENTINEL? undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears

Yesterday, a low-duty production server that I maintain core-dumped. At the time there were only around 2 calls going through it. The strace on the screen made it look like it was caused by Dahdi. The machine is running asterisk-1.6.0.3 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-addons-1.6.0 Kernel version 2.6.28 There is a genuine TDM400P (populated witrh 2xFXO cards and 2xFXS cards.

Hi guys, Sorry if I'll be very very stupid but really I write to this conference first. I have problems with configuration of app_meetme in realtime environment. I use last stable release of asterisk 1.6.0.3 Now situation is following. I create database and table in it. Th table is CREATE TABLE IF NOT EXISTS `booking` ( `bookId` int(10) unsigned NOT NULL auto_increment, `clientId` int(10)

The Asterisk.org development team has announced the release of Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a security fix for chan_iax2. Please see the associated security adivisory for more details: http://downloads.digium.com/pub/security/AST-2009-001.html These

Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR

System: FreeBSD 7.1-STABLE i386 (revision 187025) Panic message: kernel trap 12 with interrupts disabled Fatal trap 12: page fault while in kernel mode fault virtual address = 0xd2006ad0 fault code = supervisor write, page not present instruction pointer = 0x20:0xc05623aa stack pointer = 0x28:0xdd4f6c34 frame pointer = 0x28:0xdd4f6c40 code segment

The Asterisk Development Team has announced several Asterisk-Addons releases, including Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases are an incremental release after some community reported issues were resolved, primarily in the MySQL and chan_mobile realms. * Using

The Asterisk Development Team has announced several Asterisk-Addons releases, including Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases are an incremental release after some community reported issues were resolved, primarily in the MySQL and chan_mobile realms. * Using

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I'm have a bit of a problem with temporary greetings. I'm using 1.6.0.1 with IMAP storage. If I go into comedian mail and record a temporary greeting, I get a Message [0] on my IMAP server. In addition, I get /var/spool/asterisk/voicemail/default/134/temp.[gsm|wav|WAV] files. I have imapgreetings=no in the voicemail configuration file, as

Hello users, have been facing problems with t38 passthrough using asterisk 1.6.0.3. observed also that in case of SendFAX we are not having major issues, its almost successfull. ReceiveFAX has problems most of the time. we have been using a ringcentral account for testing this receivefax. so ringcentral is trying for 3 times if the sending fax failed for the first time. what i observed is

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

I have this extension: exten => 1322,1,Answer() exten => 1322,n,Set(CfMC_AMDValue="NotChecked") exten => 1322,n,GotoIf($["${CfMC_DoAMD}" != "Yes"]?NOAMD) exten => 1322,n,AMD() exten => 1322,n,Set(CfMC_AMDValue = ${AMDSTATUS}) exten => 1322,n(NOAMD),Wait(1) exten => 1322,n,UserEvent(E1322-1,${CfMC_ActionID}=${CHANNEL} & ${CfMC_AgentToUse}

I issue this command: RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ ueue^SIP/GXP280_18)) Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds. Asterisk rings again for 10 seconds. I would expect this to happen a total of 4 times. The problem is that after the second ring for 10 seconds Asterisk exits the RetryDial step with HANGUPCAUSE=0 and

Prototype v1.6.0.2 If any of an element''s ancestors are hidden, Element.getDimensions() returns misleading values. In order to address this in my own project, I made the following modification: getDimensions: function(element) { element = $(element); var display = element.getStyle(''display''); + var hiddenAncestor = element.ancestors().any(function(e) {

We don't really have a max tx packet size limit, so allow configuring the device with up to 64k tx MTU. Signed-off-by: Mark McLoughlin <markmc at redhat.com> --- drivers/net/virtio_net.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/drivers/net/virtio_net.c b/drivers/net/virtio_net.c index e6b5d6e..71ca29c 100644 --- a/drivers/net/virtio_net.c +++

Hi, I have a system with the following configuration: - Asterisk 1.6.0.3-rc1 - DAHDI Linux 2.1.0.4 - DAHDI Tools 2.1.0.2 - wanpipe-3.3.15 - Sangoma A108E card with ISDN PRI interfaces Calls arrive over the PRI interface and users need to supply DTMF digits before asterisk decides what to do with the call. The DTMF digits are read with the Read application. With low call volumes everything works