similar to: Asterisk with SuSe 10

Is it true that asterisk 1.2.6 does not use mpg123? I just installed asterisk 1.2.6 and while I do have music on hold (through format_mp3?) I do not have an mpg123 process running. I seem to be having serious audio issues when going through one of my providers (and just through that provider) when using mp3 for hold music, however when using wav files it is fine. The processor is only at about

Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration ====================== Channel map: 0 channels configured.

Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12<http://1.0.1.12>to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls

Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together. Regards Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/e71c34a3/attachment.htm

Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals? Regards L:ee ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ -------------- next part -------------- An HTML attachment was scrubbed...

Hi, is it possible to set a wcte11xp card to use a certain IRQ? I've tried a few things but it always shares the IRQ with eth0 even though the system has 4 spare ones. I can't set it via the BIOS. Regards Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050902/9ce217db/attachment.htm

Hi, It seems like my ATA is making a ringing noise... (as it used to), but now (After the upgrade from 1.0.7 to 1.2) asterisk also is either making the ringing, or passing the PRI ringing from the telco on to me. Any suggestions on how to fix this?

Has anyone had any problems with SATA, either on board or 3rd party setup? I've currently got a problem where an AMD non SATA FC2 system is working fine but an Intel system with a 3Ware SATA card and FC3 is radomly not syncing with the ISDN30. It allows and receives calls but at random intervals drops them. Regards Lee -- No virus found in this outgoing message. Checked by AVG

Hi, I would like to see the DNID in my MySQL CDR logs. I have read one big thread in the Asterisk Developer List, but I could not figure out how to implement it ? Is there a simple step-by-step. Thx in advance. Vai

Stopping and restarting Asterisk will lose the hints, then you will have to wait until the phone registers again. With 1.2.7.1 a reload shouldn't lose anything. Change the register time on the phones to something less that 60 minutes if it's a big problem. Instead of factory defaulting the phones you might find a simple restart will re-register the phone and BLF will work on the phone.

"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas

Actually it worked, but only after I defaulted all the settings on the phone and let it pick the config up fresh. Anyone know if there is any headset config options to default to headset/speaker? Thanks Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lee Archer Sent: 03 January 2006 14:49 To:

On a different note using Fedora Core 3 I get CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_chan_write': /usr/src/zaptel/zaptel.c:1745: warning: ignoring return value of `copy_from_user', declared with attribute warn_unused_result /usr/src/zaptel/zaptel.c: In function `ioctl_load_zone': /usr/src/zaptel/zaptel.c:2392: warning: ignoring return value of

Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dave Cotton Sent: 16 February 2006 14:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Firmware version 1.3.1 released for AastraIPphones On Thu, 2006-02-16 at 13:28 +0000, Lee Archer wrote: > There is no

Hi, is userfield the only extra CDR field that can be added or can others? Regards Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091103/efff367b/attachment.htm

Hi, I installed the Vestec module to one of my development Asterisk servers a few months ago but now I need to move the license to another host. Does anyone know how to do this? I've had a look on my Account page on the Digium website but it only shows the Language Pack, and I can't do anything with this either. Can anyone point me in the right direction please? Thanks Lee

I installed zaptel 1.2.17 and shortly afterwards got a problem of calls not clearing properly. I ran dmesg which showed Unable to handle kernel NULL pointer dereference at virtual address 0000009c printing eip: f8a79fa8 *pde = 00000000 Oops: 0000 [#1] Modules linked in: zttranscode button battery ac ipv6 edd wcte11xp zaptel crc_ccitt i2c_i801 i2c_core tg3 generic shpchp pci_hotplug

Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but it doesn't seem to be doing anything as the script is still exiting on a hangup and not completing properly. I am using 1.4.35 and have tried various combinations. Can anyone shed any light on this? Regards Lee -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, does anyone have an info into what could cause [Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:26:23] ERROR[13098] utils.c: write() returned error: Broken pipe Is it a write process or a problem with one

There is no release note, just a text file that says AASTRA TELECOM INC. February 2006 FC-000046-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095 for customer release. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gareth Owen Sent: 15 February 2006 02:00 To: asterisk-users@lists.digium.com