similar to: Same username on SIP & IAX?

Hello all, I'm going to tackle learning C this week, and start writing my first * add-on/contribution; assuming it's actually worthy of contributing once it's done.. I think I've chosen a hefty project for my first go round here... I'd like to get some feedback from everyone on a FindMe/FollowMe spec I've put together. Before you read on, let me say, I don't want

Hello all, I'm having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I can't seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. Originally, I had configured Asterisk to run on the NAT side so that those within the office could connect easily, and those outside the office

Hi Adam, I appreciate your feedback, and understand totally where you're coming from as far as the database perspective goes. For the first "draft" of the app, I think I'm going to let it default to using a conf file for two reasons. First, my setup currently does not utilize a database. I would like to move to that type of a setup in the future however. Secondly, seeing as

I have been attempting to download, compile and configure spandsp to function with * without much luck. I am guessing that some assumptions were made regarding the users knowledge level of Linux. Sadly, I must not live up to those assumptions. My problem begins when after compiling spandsp I look for the app_rxfax.c, app_txfax.c, app_dtmftotext.c and makefile.patch files to place in the

I ordered 10 7960's with SIP today (YAY!), I should have them on Monday! So, to be better prepared come Monday morning, I was wondering if anyone knew of any * compatible screen configs for things such as browsing VM, etc, yadda, yadda. I checked out the wiki about ADSI but from what I see, that's not really applicable in a SIP setup? I'm guessing it's going to be a more XML

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

I have a UIP200 on the way for eval. Does anyone have tips or tricks to get it working right away with * ? I hate having to go through the pain someone else braver than I went through already. :) Tim --

I was wondering if there was a way of setting up the dialplan in a way that if you dial an extension that is NOT in the dialplan then it would play a not-in-service gsm file and then play congestion tones. I would rather like this better than just hearing a busy signal on my phones.. I DID search around on the wiki and using google and could not find anything. Thanks. -- Stephen Rosebush,

Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Hello to all: When I make a call of extension (FXS) to extension (FXS) (TDM40B) status of the call is the following one: #### Case call answered --------------------------------------------------- Log init -- Starting simple switch on 'Zap/4-1' -- Executing Macro("Zap/4-1", "stdexten|103|Zap/2") in new stack -- Executing Dial("Zap/4-1",

Alright, this may seem like something relatively easy to do but I must be missing something or had a neuron misfire. I am trying to get The Status lights on my Snom200 hardphones to display the status of each one of my PSTN lines in my Asterisk server. Current Config: 3 X100P cards Asterisk CVS-02/25/04-18:06:52 5 Snom200 phones I am currently using the following macro to dial out

I installed Samba 2.2.2 on my Linux box. The smb.conf file is as shown below. The permission on /var/spool/lpd/lp is set to 755; ie, it is writable only by the owner, which is lp. With the setup as is, when I try to print from the NT machine, the smb log on the Linux box indicates the following error: " print_job_start: insufficient permissions to open spool file /var/spool/lpd/lp". If

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

On 04/05/16 14:07, Cole Robinson wrote: > On 05/04/2016 06:54 AM, Paul Carlton wrote: >> Hi >> >> I'm trying to create a volume in an existing storage pool using python by >> calling createXML() on the pool object. >> If the file that is the subject of the new volume exists you get and error, >> also if it doesn't exist >> >> I worked out

On Thu, 4 Aug 2016 14:03:39 +0100 Nabeel <nabeelshikder at gmail.com> wrote: > I should add, a password is *always* asked if a password has been set. > There isn't a way to bypass that. Then something is wrong. http://darcy.vex.net/star98.mp3 -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net VoIP: sip:darcy at Vex.Net

I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",

>>>>> D'Arcy Cain <darcy at VybeNetworks.com> writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos <cloos at jhcloos.com>

I know that it runs on other systems but do other ports get the same attention? I have been running it on a NetBSD server for about a year now and while it mostly works it just crashes from time to time with no explanation or core dump. I have improved the situation by expanding my intrusion detection but it still stops every few days or so. I have a cron job that tests for it and restarts it

I have an extension that looks like this: exten => 5555551111,1,Verbose(Door buzzer calling) same => n,Dial(SIP/user1&SIP/user2&SIP/user3) The idea is that any of the three users can answer the phone to let someone in. The problem is that if, say, user2 unplugs his phone then the call immediately goes to his voice mail and the other two do not have the ability to open the door.