similar to: SNOM Red LED on DND or unregistered Phone

Hello, I would like to double check what Echo Cancellation my Digium Card uses. I thought I bought the little more expensive card that integrates EchoCancellation. How can I check? root at sn1:~# zaptel_hardware pci:0000:0b:08.0 wcte12xp+ d161:8000 Wildcard TE121 root at sn1:~# ztcfg -v Zaptel Version: SVN-branch-1.4-r4309 Echo Canceller: MG2 Is MG2 the correct one that I am supposed

JFYI - I run (successfully) agx-addons with 1.4.22 and Etch. Make sure you have the right version of SpanDSP installed (as well as the tiff libraries). -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Loic Didelot -->> Sent: 19 December 2008 20:33

Hi, I noticed that my hint priority stops working when I add to many extensions/channels. It looks like everything exceeding 80 characters is discarded. By stop working I mean the status is and stays "Unavailable". This works exten => *1,hint,SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1 This does not work: exten =>

Hello, I wanted to relaunch my problem with a quite precise and small question. Could the command "lsusb" affect interrupts when executing it regularly (cron) and if the zaptel pci card is sharing an interrupt with a usb bus? I have a Digium PRI card and a few Xorcom USB Devices. The Digium card seems to share the IRQ with some USB bus. cat /proc/interrupts CPU0 CPU1

...described in the README file ;). SpanDSP - 0.0.4 pre 16 LibTiff - >= 3.8 but <4.0 I had to trawl around for the right SpanDSP - but I can e-mail a copy to whomever wants one (drop me a personal e-mail and I'll attach it by return) HTH Andy -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of

Hi, I just got a Xorcom Astribank with 8 FXS but it does not work. So I tried resetting and loading the firmware. But loading just times out. /usr/share/zaptel/xpp_fxloader usb --------- FIRMWARE LOADING: (usb) [1 devices] 'xpp_fxloader'[11561]: USB Firmware /usr/share/zaptel/USB_FW.hex into /dev/bus/usb/002/003

Hello, I played with the externalIVR application. So far I am able to read digits and play sound files. But how can I leave the application and continue in the dialplan so that I can execute other actions like going to voicemail, or ringing users etc.... Best regards, Lo?c. -- Lo?c DIDELOT MIXvoip S.a. Tel: +352 20 3333 20 Fax: +352 20 3333 90 ldidelot at mixvoip.com http://www.mixvoip.com

Hello, I recently got a problem I have never seen during my previous installations. I have a Xorcom device with 4 BRI and 8 FXS ports. I use ubuntu Linux and asterisk 1.4.22. Everything works fine but after some time zap channels disappear in asterisk. root at pbx1:~# cat /proc/zaptel/* Span 1: XBUS-00/XPD-00 "Xorcom XPD #00/00: BRI_TE" AMI/CCS NOTOPEN 1 XPP_BRI_TE/00/00/0 Clear

, address 2 Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.344611] ERR-xpp_usb: XBUS-00: x= usb_listen: usb_submit_urb failed: -19 Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.353325] INFO-xpp: XBUS-00: [] D= isconnecting Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.353373] Zaptel: Master changed = to XBUS-00/XPD-01 Dec 30 21:42:08 wedekindpbx1 kernel: [ 3746.353382] Zaptel: Master changed = to

Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:

Hello, I am testing the Openstage phones from Siemens but I can not find a solution on how to update the caller-id after a successful attended transfer. Of course, I mean an attended transfer by using the phones functionality, not something defined in asterisks features.conf. Any idea on how to achieve this, or any technical document from Siemens on on how this is support to work would help.

Hello, I am having some troubles with Snom phones and maybe someone can help me. Let me say this: BLF and pickup works great with Polycomes and Grandstream etc... So I think my problem might not be Asterisk related but I am not 100% sure. The snom phones subscribe to my extensions (hint priority) as expected. The light blinks (ringing) or is turned on (in the call) as expected. My problem is to

Forgot on the Snom 200 it won't reboot if under the Memory tab in the web interface, Connections > 0 then remote reboot is not possible. Manually cycling the power allows the phone to be rebooted by Sipsak remotely. HOWTO: Reboot a Snom with Sipsak Checklist: 1. Under Advanced in the web interface, is Network Identity set to 5060? 2. Under Advanced in the web interface, is Challenge for

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Hello all, this may be slightly offtopic :-) I have some Cisco 7940 phones with SIP firmware, connected to an Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g (HorstBox Pro with custom extensions.conf). On some of the phones, two lines are configured, one for business, one for private calls. When forwarding a line to another destination (e.g. to voicemail), we can't use the phone's own

Hey all, First of all, thank you for the help I've gotten on this list in the past. Very helpful, and I apprecaite it. Now, what I'd like to do is send a message to my snom 320s. I'd like to have the message display regardless of what the phone is doing. I have been trying SMS, or the sipsak method on the wiki but I have had no luck thus far. Does anybody have this working,

Hi, I've read README file in agx-ast-addons-1.4.17.5.tar.bz2 It says "Install libTiff >=3.8 and <4.0" Should you really use this agx-ast-addons to get app_rxfax and app-_txfax running with latest 1.4.22 ? If positive, should you take this libtiff warning into account ? If positive, where can you find such libtiff version as Debian repository (I didn't check alternate

I'm afraid I already know the answer because I've done a lot of searching, but does anyone know of a softphone that supports a central phone book and paging (like the sip autoanswer option of some hardphones) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could