similar to: kernel: dahdi: Detected time shift.

Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is

Hello, I am getting the following error when compiling dahdi : make[2]: Entering directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64' Building modules, stage 2. MODPOST 0 modules make[2]: Leaving directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64' make -C /lib/modules/2.6.32-431.1.2.0.1.el6.x86_64/build

Hello, when compiling dahdi (CentOS 2.6.18-348.3.1.el5) I get the following error : In file included from /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xpd.h:26, from /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/card_bri.c:29: /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xdefs.h:152: error: conflicting types for

Hello list, when installing Dahdi, the following error comes up : You do not appear to have the sources for the 2.6.18-164.11.1.el5xen kernel installed. make[1]: *** [modules] Error 1 The running kernel version : -bash-3.2# uname -a Linux vds.hosting.net 2.6.18-164.11.1.el5xen #1 SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux -bash-3.2# ls /usr/src/kernels/

I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. When I plug one PSTN-line into a FXO-port I am able to receive calls on this line and I can also make calls from an internal SIP-phone to the external PSTN-network. Still I am bothered about something that appears on the CLI when I do a reload chan_dahdi.so : asterisk*CLI> reload chan_dahdi.so -- Reloading module

Hey list, I'm getting the following warning when installing dahdi-linux-complete-2.2.0.2+2.2.0 : make[2]: Entering directory `/usr/src/kernels/2.6.18-128.1.6.el5-i686' Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-complete-2.2.0.2 +2.2.0/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd for /usr/src/dahdi-linux-complete-2.2.0.2

Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init

Hello, I see a lot of these messages in the debug log : /[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Read factory 0xae173e0 and write factory 0xae17e18 both fail to provide 160 samples [May 3

Hello list, I get the following error : pbx_extension_helper: No application Page for extension Apparently I have no timing source installed. But I thought that Dahdi did not need to be installed for timing ?! And that there is some internal timing in Asterisk 1.6.2.10 ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I want to install Dahdi and Dahdi-tools on a CentOS 5.3 Xen host and I receive the following error : "You do not appear to have the sources for the 2.6.20-prep kernel installed." I have installed : - kernel-headers-2.6.18-128.4.1.el5.x86_64 - kernel-devel-2.6.18-128.4.1.el5.x86_64 - kernel-xen-devel-2.6.18-128.4.1.el5.x86_64 bash-3.2# uname -r 2.6.20-prep bash-3.2# ls -l

Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip

Hello list, can anyone tell me if this card : http://www.audiocodes.com/product/ipm-260-sip is compatible with Asterisk (DAHDI) for use as PCI PRI card ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/7160a55d/attachment.htm>

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

Hello, when trying to start Asterisk on a new server, I got the following problem : [root at sip asterisk]# /etc/init.d/asterisk status asterisk is stopped [root at sip asterisk]# /etc/init.d/asterisk start Starting asterisk: Cannot find specified TTY (9) [FAILED] I can start asterisk with /usr/sbin/asterisk -c but when I quit the CLI

Hello list, is it necessary to properly answer() an incoming call ? I don't want to answer a call because the caller has to pay even if the attached SIP-phones do not answer the phone call. Because I answer() the incoming call, the caller has to pay for 60 seconds of 'ringtone'. On the other hand, sometimes an incoming call is send to a macro where the caller is given the

Hello list, google returns a discussion on the dev-list when I search for how to mail a voicemail to multiple mail addresses. Is there yet a seperator that actually works to define multiple mail addresses ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! > How to do this ?? > To proceed with your answer on PICKUPMARK, where do I put this ??? Look at the example for Asterisk 1.4 on this page: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup Philipp