similar to: Shared Line Appearance - Aastra 55i - Does it work?

On 8/7/07, raviprakash sunkara <sunkara.raviprakash.feb14 at gmail.com> wrote: > > Hello Russell, > Nice To meet U and Good Morning. I got u r mail-Id from > http://www.asterisk.org/node/48325 > Recently i started the SLA configuration. But i didn't understand the > Flow of its Functionality > One of the My Client Ask to have do deploySLA feature > He Using

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

Hi, I have a strange error, that I haven't yet found in another posting on the archive. Here's my sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls callerid=No CallID [stefan] type=friend secret=blah username=stefan

Hello all, I understand that during PXE boot, it goes into the pxelinux.cfg and it states: 1. "it will search for the config file using its own IP address in upper case hexadecimal, e.g. 192.0.2.91 -> C000025B (you can use the included progam "gethostip" to compute the hexadecimal IP address for any host.) If that file is not found, it will remove one hex digit and try

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

Fellow asterisk users, I am trying to get Single Line Appearance functionality working on a set of Linksys SPA942 phones and have not been successful. It looks like sla.conf is not getting read, only one phone reads as registered for the shared line, and a busy tone every time the shared extension is dialed. I have followed the documentation [1] and followed through other threads I saw

I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight. I took the additional step of nuking my modules directory first... When I used the command asterisk -v to start asterisk, it seemed to go along and get to the point where asterisk is running(ie Asterisk Ready). At that point it was eating all available CPU. I went ahead and tried to register a softphone to it via IAX2, which

From eb20603913ff7350cd25b39d1eb37b8fddd16053 Mon Sep 17 00:00:00 2001 From: Zhang Xiantao <xiantao.zhang@intel.com> Date: Sat, 25 Aug 2012 04:11:08 +0800 Subject: [PATCH 2/2] Nested: VM_ENTRY_IA32E_MODE shouldn''t be in default1 class for IA32_VM_ENTRY_CTLS_MSR. If set to 1, L2 guest''s paging mode maybe mis-judged and mis-set. Signed-off-by: Zhang Xiantao

Hi again, Kind of sheepish about asking for help, as I have only spent a day banging my head off this... I got my new Welltech 3701a, 1FXS,1FXO gateway. I flashed it with what is seemingly the appropriate firmware (SIP V1.04). This seems to have gone ok, and it is now registering both ports ok with asterisk. For 1 minute I thought I was home free and and everything was just going to work

Well, it's not my native language so I'm not pretty sure of my english. :) Did somebody try the xsls ? > Poor english? Where? :D > Deven > On Sun, 2004-01-25 at 11:08, Gilles Dartiguelongue wrote: />> Hi all, I had a look at the web interface yesterday and I was a little / />> bit disappointed / />> by the code I found. So I started to make modifications

Hi, Could you please help me!! I am trying to configure the Asterisk server. I have a analog phone connected to a FXS port of a Cisco 3745 router. This router is connected to a Asterisk server via Fast Ethernet interface. I am trying to make a call from the analog phone to a SIP client. This SIP client is registered to the Asterisk server. Analog phone number: 999 SIP client : 202 Sip client IP

Hi ?? I have been trying for several days get 3 Cisco spa508g phones (firmware 7.5.5) to work with asterisk 11.6 cert1 and sla. I can get the phones to all ring when an incoming call arrives, and I see the slatrunk working. However the blf function does not work. If one extension picks up the call the others do not show the trunk in use.? And as you might expect the hold and outbound dialing does

The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

flight 13627 xen-unstable real [real] http://www.chiark.greenend.org.uk/~xensrcts/logs/13627/ Regressions :-( Tests which did not succeed and are blocking, including tests which could not be run: test-amd64-i386-xl-win-vcpus1 12 guest-localmigrate/x10 fail REGR. vs. 13625 test-amd64-amd64-win 5 xen-boot fail REGR. vs. 13625 test-i386-i386-xl-win 5 xen-boot

Hi Gilles, > Well, it's not my native language so I'm not pretty sure of my english. :) My neither. Hey this is a multinational community. ;) > Did somebody try the xsls ? yup. Tried your xlst-files for the stats. Noticed some improved look of the GUI. Did not try the admin xlst's coz I didn't want to mess up my productive system by some silly mistake. ;) Now looking at

Thomas B. Ruecker, DM8TBR wrote: >Hi Gilles, > > > >>Well, it's not my native language so I'm not pretty sure of my english. :) >> >> >My neither. Hey this is a multinational community. ;) > > > Yeah, that's right. >>Did somebody try the xsls ? >> >> >yup. >Tried your xlst-files for the stats. Noticed some

Hi! I have small problem. On my sound tab i have only OSS and i didn't have sound in "steam" games. Emulation not work(OSS too). Somebody now why? :) Ubuntu 8.04. greetings! redlin

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * ---

Hi Everyone, We use the top buttons on Aastra 55i to login and logout from Queues. This is the order: Button 1 = Login to English Queue Button 2 = Login to Spanish Queue Button 3 = Logout of English/Spanish Queues There are indicator LEDs on each of these buttons. Is there anyway we can send a SIP request or some other communication to get the Aastra 6755i phone to keep the LED for login set