similar to: snom mass deploy help

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

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

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 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

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'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

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

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

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.

I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I can see the channel bank with lsusb, but when I tried to use zaptel_hardware, or when I try the /etc/init.d/script, they don't see my Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's dependecies, fxload and libusb-dev. Anyone have a similiar experience ? Best Regards, -- Guilherme Loch G?es

Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091213/1f81f008/attachment.pgp

Hi I am having some problems installing a paravirt RHEL guest on a deb amd64 xen machine uname -a = Linux debxen 2.6.20-1-xen-amd64 #1 SMP Tue Apr 24 22:23:22 UTC 2007 x86_64 GNU/Linux ============== xm info ============== host : debxen release : 2.6.20-1-xen-amd64 version : #1 SMP Tue Apr 24 22:23:22 UTC 2007 machine : x86_64

Hi, I have the following problem. I have 5 phones behind the same nat (canreinvite=yes). it works fine to receive calls and to make calls. sound quality is good, so everything works fine. The poblem is that the phone behind nat cant call each other. It works if canreinvite=no. But i want to do this. Does anyone have an idea? Regards, cjk.

Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again ..... Alex

Hi I am in the process of installing a new box and using dahdi. I have a tdm410 + hardware echo canceller. I have just read in the read me for dadhi that VPMADT032 support has been removed and unlike with the zaptel stuff i could just download and install the firmware I can't with dahdi what is the best way forward to recompile with hardware echo canceller support. Thanks Alex

Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan

Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I have read up about a PWR2400b and it seems to use 2wire pin, I am guessing to connect to P8 just behind the molex connector on the tdm410. can any one here confirm this, or have any info on the

Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus