search for: hadly

Does anyone know how to set the distinctive ring on the Linksys SPA941? I want to be able to dial one extension and have the phone ring with a certain tone and then dial another and have the phone ring with a different tone. I have tried the following ------------------------------------------------------------------- exten => 802,1,SIPAddHeader(call_info=Classic-4) exten =>

I recently posted a question RE the Sipura 941 and using different ring tones, Thanks to hads I managed to use SET(_ALERT_INFO=Classic-1) to achieve this but trying this on the GXP 2000's didn't seem to do the trick?? Has anyone one had any luck on this topic? Also haven't been able to find any info on an auto-answer for the GXP 2000, again, I have succeeded in doing so with the

Hi Can someone please help with the following, We are using asterisk@home 1.5 and SIP trunks to communicate to the PSTN network. We are having some problems with the call quality. Although we can hear the other person's voice quite clear when making or receiving a call, we get complaints from the people on the other end saying that our voices sound very unclear, low and that the voice

Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22

Hi I am making an extraordinary claim: rysnc seems to copy all my files, not just ones that have changed or new files. rsync version 2.6.0 protocol version 27 Debian 3.0 Woody I have tested this with one simple file, my example is shown below. Does anyone have any suggestions to rectify this? Regards Gareth example: I create a file called 'testfile' in /tmp/play containing the

Hi Hadley, thank you for providing this "scagnostics" primer.... I was trying to do some basic testing, and I see that I probably missed some points : first it's not clear for me if the argument of "scagnostics" should be raw data or "processed" data (results of calling "splom" or whatever...). If the first, I thought (from Wilkinson & al.) that if

I have written up an guide on how to do bulk provisioning of the Linksys phones and ATAs. http://voipspeak.net/index.php?option=com_content <http://voipspeak.net/index.php?option=com_content&task=view&id=73> &task=view&id=73 Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 -

Ok, I must be really stupid here - I'm playing with ael and svn trunk. given the following in ael: context isdn10 { 444601 => { Answer(); NoOp(${CALLERIDNUM}); Hangup(); }; }; isdn10 is the incoming isdn context. why do I get this on the console: -- Accepting call from '01702xxxxxx' to 'yyyyyy' on

hey all, I'm getting this warning on the console when I leave a voicemail on my test server: [Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6552 vm_exec: Prefixing the mailbox with an option is deprecated ('u250@local-vm-users'). [Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6553 vm_exec: Please move all leading options to the second argument. This is what voicemail.conf

I was planning on using a TDM400P with 3 FXO & 1 FXS, with the 1 FXS being used for a fax machine. It now appears that Digium doesn't support this, are there other manufacturers anyone can recommend that will support it? Has anyone used a TDM400P in this setup and had it work without much issue? Thanks for the help, Ken -------------- next part -------------- An HTML attachment was

Hi all, I was wondering whether anybody here would help me clarify this minor issue please. If I have the following setup; Asterisk ------ Sipura SPA 3000 (fxo) --------- Pstn Line Would a call coming in on the pstn line be answered by the ATA or just get passed through to the * server (depending on dialplan) to handle? So basically, the caller does

When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and

As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing

I have fixed this before, but I cannot for the life of me remember how I did it. I have a TDM400P with one fxo module and one fxs module. I setup Asterisk @Home and everything works fine, except when I try and call out. If I call out with a SIP phone it calls the zap extension and not the pstn line? If I take the zap extension offhook and call with the SIP phone it dials out the pstn line

Hello! This might be a FAQ, but I didn't manage to find an answer somewhere: why doesn't the linux-image-2.6.16-2-xen-686 (at least that one; that's the one I installed) create a accompanying initrd.img, but the user has to do it himself? (Like me, having troubles with the version of initramfs-tools from testing, resolved by upgrading that one to unstable's version.) Also, why

I am aware of Astlinux and the other embedded Asterisk solutions out there? Astlinux is nice but the problem is that when I hit a snag and need to incorporate a patch and what not I cannot do that with Astlinux because I cannot compile my own version. How hard is it to create my own version of Linux/Asterisk to run on Compact Flash. I have seen 1GB Sandisk CF for as low as $50 recently so small

Hi, I would like to see if during a call a new voicemail was recorded. I want to send a SMS to mobile phones if someone recorded a message on our voicemail system. I can use VMCOUNT to see if there are new messages in the Inbox but this will result in new SMS being sent even if the caller hangs up during the Voicemailpromt, at least if there are still unread/unheard messages in the inbox. Is

Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __________Asterisk fxo ---- line -----| -----------------Analog phone The analog phone rings immediately when calling, while asterisk shows the message

I figured I'd answer my own thread and document what it took to get rid of the echo at my location. For those of you trying to get rid of echo, let me tell you, "what worked for that guy, probably won't work for you". I think we've all heard that before, and it's true. Let me assure you that echo can be removed from your phone lines. At 20 hours into my 40

First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. With that out of the way.. Is anyone aware of any type of failover device for PRI on asterisk? I've found the ISDNGuard, however it is currently not made in the U.S., nor does it run on U.S. power. Is anyone aware of a device that will