similar to: Inclusion of the ILBC codec starting from asterisk 1.8.10/10.2

Hi, I explain what I want to do.. All the operators share their phones. The number of the operator isn't constant, so it's possible that two operators share all the phones. They need to move all around, so they pick up the first phone they find. If there are only few operator is very annoying for them to ear the other phones ringing while they are at the phone! So I'dd like to limit

Hi, this is an old outstanding problem, unfortunately I don't remember how to walkaround it. I use asterisk 1.8.3 and I have a public IP in my network interface. As soon as the Internet connection goes down, phones stop working. I want to be able to use pstn, isdn and the gsm gateway even if the Internet connection goes down, how can I achieve it? Thank you, Darkbasic

Package: xen-hypervisor-4.0-amd64 Version: 4.0.0-1 Severity: grave The only thing I can read is: ERROR: Unable to locate IOAPIC for GSI 2 ERROR: Unable to locate IOAPIC for GSI 9 ERROR: Unable to locate IOAPIC for GSI 9 cannot read /etc/fstab: no such file or directory mounting [...] failed: no such file or directory I boot with: moltiboot [xen] module [kernel] root=[uuid or device] ro quiet

Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccol? -- http://www.linuxsystems.it

Il 27/04/2011 14:37, Ian Campbell ha scritto: > Which specific revision of qemu-xen-4.1 are you using? I'm using a tarball from http://bits.xensource.com/oss-xen/release/4.1.0/xen-4.1.0.tar.gz It seems Thomas used the tools/ioemu-qemu-xen subdirectory as source for xen-qemu-dm. At a first glance I didn't notice debian/mypatches/add-rules.patch: it adds Rules.mk from the tools

Distro: debian unstable Xen version: 4.0.0-2 This is the config: #HVM #still no qemu-dm in debian #kernel = "/usr/lib/xen-4.0/boot/hvmloader" #builder = ''hvm'' #memory = 2048 #name = "webserver" #vif = [''bridge=eth0''] #disk = [''tap:tapdisk:aio:/srv/xen/webserver.img,xvda,w'',''phy:/dev/cdrom,xvdb:cdrom,r'']

# dd if=/dev/zero of=/srv/xen/webserver.img oflag=direct bs=1M seek=4095 count=1 # dd if=/dev/zero of=/srv/xen/webserver.img oflag=direct bs=1M count=4096 root@debian:~# grub-install /dev/xvda Searching for GRUB installation directory ... found: /boot/grub The file /boot/grub/stage1 not read correctly. Darkbasic _______________________________________________ Xen-users mailing list

Hi, I have a Portech MV-374 GSM Gateway and I'd like to send SMS from a web page to confirm the subscriptions. How can I achieve it? Is Asterisk of any use to send SMS with the Portech? I really have no idea because I know nothing about the whole SMS thing... Thanks, Niccol? -- http://www.linuxsystems.it

Hi, I continue the discussion from https://issues.asterisk.org/view.php?id=19103 If T.38 reinvite detection should still work, why it doesn't? If I use faxdetect = t38 it does never detect the fax, even using alaw. Cheers, Darkbasic

Do you know if squeeze has xen support? _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Is there a fix for this in the works/coming for squeeze or do I have to patch the package myself? If so, how? Thanks, a grateful Debian user that hasn't patched packages yet! Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.alioth.debian.org/pipermail/pkg-xen-devel/attachments/20101122/8433308c/attachment.htm>

Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the "Idle" state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him

I've contacted Riello and they told me that the NetMan 204 is the new version of the Netman Plus 102 and is indeed compatible with NUT. I don't know how to update the Hardware compatibility list page, but the 204 should be added to the compatible ones. Niccolo' Il 2022-06-17 12:02 Niccol? Belli via Nut-upsuser ha scritto: > Hi, > I'm looking for a small 230V Line

I've contacted Riello and they told me that the NetMan 204 is the new version of the Netman Plus 102 and is indeed compatible with NUT. I don't know how to update the Hardware compatibility list page, but the 204 should be added to the compatible ones. Niccolo' Il 2022-06-17 12:02 Niccol? Belli via Nut-upsuser ha scritto: > Hi, > I'm looking for a small 230V Line

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid

Hey all, I have moved 10_Linux to 50_Linux and ran update-grub which is fine. The problem I have is that there is xen-syms in /boot And that is being put first in the boot list which is incorrect. How do you make /xen.gz at the top so it is booted first? Thanks Ian _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

Hi, I'm having some questions related to echo cancellation configuration on a Digium board enabled systems (B410P, TE420, TE420B, ....) for cases when a hardware ech canceller is present or not. I read in TEXXX manual that when setting echocancel=yes in chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo cancellation was enabled. 1. I'm correct thinking that it is then

Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Thanks! Regards, Ronald -------------- next part -------------- An HTML attachment was scrubbed... URL: