similar to: no audio between sip channels * 1.2.6

Hello all, I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use "Get Data filename, timeout, maxdigits " in the agi script. In * console I get message saying playing filename but it

Hello all, When I call the Broadvoice number all is good. When I try to call out through DISA on my broadvoice line i get the following: Executing Dial("SIP/147.135.0.129-0815bc60", "SIP/16037862111@proxy.bos.broadvoice.com|30") in new stack -- Called 16037862111@proxy.bos.broadvoice.com -- Got SIP response 480 "Temporarily Not Available" back from

Hi, I am using SIP extensions connected to the PSTN with the CAPI Channel driver. All works fine except that one of the sip phones keeps ringing when the caller hangs up before extension is answered. The phones are grandstream 100, though we get the same behaviour using other phones (X-lite, Kphone). It behaves the same regardless of whether the incoming call is from a SIP extension or an

I recently encountered this in the logs of a new Debian Xen Dom0, and having now spent the better part of a day researching and testing, I've come to the conclusion that this is not a bug in xen-utils-common or even iptables; it's merely the consequence of structural changes to the core netfilter code starting in the 2.6.20 kernel. This is rather long, but the issue is complicated. Please

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the 480i

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the

Eero and others, I've already emailed this to Michael Wood: > Most likely your D-Link router has a built in 4 port switch and does > not control access between these 4 ports, so I doubt you need to do > anything on the D-Link router. Okay. I am now thinking it may be routing on the Windows box, so I will next put my focus/attention more to that. > > ? ? For example, I

Hello everybody, I have a problem and already browsed the mailing list archives but didn't find any help. So I ask here.... My new * Box ist up & runnig. Got access to the SIP server of my Internet provider (Userid, password, phone number, ...). And yesterday I tried my first calls to the outside world. (Internal calls work). Now when I call from the SNOM-360 connected to Asterisk to my

Thank you Melissa. I love the phone but the dial keypad is a little bouncy. I was hoping for a more solid feel like on the analog PT390's or my quality standard, the Nortel 9417CW. Other than the MWI problem, I'd like more documentation on the configuration paramters. I have found little online configuration documentation other than very basic stuff on the Sayson website. I'd

https://bugzilla.mindrot.org/show_bug.cgi?id=2725 Bug ID: 2725 Summary: can't login Product: Portable OpenSSH Version: 7.4p1 Hardware: 68k OS: Mac OS X Status: NEW Severity: normal Priority: P5 Component: ssh Assignee: unassigned-bugs at mindrot.org Reporter:

I'm sure I'm doing something wrong here, but I would appreciate a pointer. I have tried to rtfm, but find the docs a little sparse wrt pxechain.comand gpxelinux.0. I am trying to setup a pxe chain server (aka chainloading?) where one of the entries on one pxe server forwards to another (cobbler, in this case). I know this works fine with traditional pxelinux.0 images, but it's

On Wed, Aug 7, 2013 at 3:58 PM, Hans Lellelid <hans at velum.net> wrote: > I'm sure I'm doing something wrong here, but I would appreciate a pointer. > I have tried to rtfm, but find the docs a little sparse wrt > pxechain.comand gpxelinux.0. > > I am trying to setup a pxe chain server (aka chainloading?) where one of > the entries on one pxe server forwards to

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

Hello, I am just curious how many people are hooked up with BroadVoice and have recently been experiencing a lot of dificulty. Joel

Hi, I have spent a fair amount of time trying to solve this problem and tried things like disabling checksums etc. The problem is like this: 1. I boot my dom0 without xend running and I can resolve and ping google. 2. I start xend (without and domU domains up) and after a while I can no longer resolve domain names, but I can ping the public IP addresses. 3. I start up a domU (xm create -c xyz)

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

I have Suse 9.1 + Samba 3.0.9 + winbind joined to a Windows 2000 AD domain "MyCompany". Yesterday for some unknown reason (appears to be Win2K issue) connections to Samba and the ability to browse the domain "MyCompany" from Win2K/XP workstations failed but connections to the public share offered by the Win2K server were OK. Also, all Win98 workstations and were unable to

Hello all, Okay when you are done laughing at the simplicity of this question could someone show me please what I have wrong in the following statement? GoToIf($[${numdial} != [1-9] ]?15:3); What this is supposed to do is if numdial is not a single digit from 1 to 9 inclusive goto 15, if it is a singledigit from 1 to 9 inclusive goto 3. Should be pretty simple but not working for me, always

Hello all, I am looking into building a calendar server (due to business requierments I can not use public hosted calender like Google), and am looking for suggestions based on experience with different calendar applications/servers available for Linux that you have integrated with Asterisk. If you can give a quick, simple list of what worked and what didn't I would be very grateful.