similar to: Xorcom and Doorbell

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 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

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

Hello, I am trying to update to dahdi 2.2.1. But somehow the xorcom (2BRI) device does not want to work. I pasted below the information I have. pbx at loicvoip1:~# /usr/share/dahdi/xpp_fxloader reset pbx at loicvoip1:~# /usr/share/dahdi/xpp_fxloader usb 'xpp_fxloader'[4937]: --------- FIRMWARE LOADING: (usb) [1 devices] ..Got all 1 devices 'xpp_fxloader'[5003]: ---------

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 =>

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 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.

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 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

Hi, Does anybody know of a VoIP compatible doorbell or door intercom unit? I've contemplated buying a cheap SIP phone, ripping it apart, and putting it inside an IP66 sealed unit... It would need: - At least one speed-dial key, or some way to make every button dial the same extension number - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart

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.

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

Hello, I have a customer who has their front door integrated to their current phone system. If someone presses the buzzer, the secretaries phone will ring, and she can talk to the person at the door. By pressing ** she can release the door. Anyone have any sort of integration like this. Are there IP devices anyone is using? They have a pizza box server as their asterisk server with

Given the need for 4K doorbell such that QEMU can easily map, ect, and assuming that I have a HW device which exposes 2 VQ's, with a notification area off of BAR3, offset=whatever, notifier_multiplier=4, we don't need to have 2 x 4K pages mapped into the VM for both doorbells do we? The guest driver would ring DB0 at BAR4+offset, and DB1 at BAR4+offset+(4*1). The 4K per DB is useful how?

Hello, I have several customers describing something like annoying "silence suppression". So I did some tests and I can confirm. After disabling echo cancellation in zapata.conf the "silence suppression" effect is gone, but there is a little echo of course. I do not have this problem on other boxes where I am using oslec as an echo canceller. All my calls are SIP to ZAP (E1).

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not

Hi all: This series introduce basic functionality of doorbell mapping support for vhost-vDPA. Userspace program may use mmap() to map a the doorbell of a specific virtqueue into its address space. This is help to reudce the syscall or vmexit overhead. A new vdpa_config_ops was introduced to report the location of the doorbell, vhost_vdpa may then choose to map the doorbell when: - The doorbell

Hi all: This series introduce basic functionality of doorbell mapping support for vhost-vDPA. Userspace program may use mmap() to map a the doorbell of a specific virtqueue into its address space. This is help to reudce the syscall or vmexit overhead. A new vdpa_config_ops was introduced to report the location of the doorbell, vhost_vdpa may then choose to map the doorbell when: - The doorbell