similar to: * 1.0.9 Voicemail record name does not playback in Directory()

Used the Directory application instead of the Directory AGI. -----Original Message----- From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Tuesday, January 23, 2007 11:29 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] * 1.0.9 Voicemail record name does not playback in Directory() On * 1.0.9 User logs into voicemail, dials Option Zero, then Option

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

Disclaimer: Not a troll I'm curious as to this obsession with uptime is. All of the posts of this type are along the lines of "After X days, Y thing does not work but if I reload or reboot, it's OK" - so why not cron a reboot? Is it considered bad form or something like that? I reboot every night whether it is needed or not, not afraid to admit it, and everything works fine for

>I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN?

I know this is not strictly an asterisk issue but it is related I guess. Just to let you know that after many calls to Mitel the consensus is that they will be releasing a new version of the 5220 that is dual boot (minet and SIP) next week or the week after. This firmware will only appear on NEW phones manufactured after the release date (no one could confirm but the 23rd of sept was

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,

I'm using a Snom 320 in a CAP position and the receptionist wants to do blind transfers. OK, no problem so far. Now she has asked me how to UN-transfer a call, as in, she transfers a call and wants to hook the call back before it connects (she wanted to tell the caller additional information for example) I don't think that this is possible as once my dialplan starts using Dial()

I want Asterisk to do the following: - call a voicemail system by dialing a number and playing a DTMF tone - record what is said by the called party and save the recording to a file - end the recording when a particular phrase is said by the called party - put that recording into an Asterisk voicemail box and notify the user I've made a start below (on the easy bit). Any further pointers on

AHA! # 1 is the case! Seems the user was fooling around with the phone after the firmware upgrade. Shame that that setting couldn't be locked out. Thanks to Mr Tahir and Mr Stredicke for their spot on responses. -----Original Message----- From: Usman Tahir [mailto:Usman.Tahir@snom.de] Sent: Friday, September 23, 2005 12:34 AM To: ColinA@landmarkmasterbuilder.com Cc:

Colin, Thanks for your assistance. Reading over your advice I seem to still be a bit confused. My agents are not on the Asterisk server; it appears in your advice that my the call will travel this path: WWW interface --> agent enters their DID, platform to use, and termination DID --> AST calls agent --> Agent calls termination DID If my agents are not on the Asterisk server

Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. ----- Mike Hammett Intelligent Computing

the user you are connecting as should have full rights to /var/run/asterisk: http://www.voip-info.org/wiki-Asterisk+non-root hth -----Original Message----- From: Erick Perez [mailto:eaperezh@gmail.com] Sent: Monday, April 03, 2006 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk.ctl

In the example I posted previous, there is an obvious gaping security hole, it would be trivial for someone to read the querystring and exploit it to make free phone calls, spoof caller ID (if you allow the CallerID to be set with a QueryString value), etc. You want to make damn sure that the URL is not publicly accessible or somehow obsfucate the querystring, or use POST. In my case, I

http://www.pcmag.com/article2/0,1759,1812887,00.asp Specifically, his assertion that ISP's would sniff traffic and block, say, the SIP port. You could play wack-a-mole with port numbers, no? Also a community based, Freenet style of encryption implementation for "free" VoIP traffic would address this issue. I raise this to the list because I'm sure there's a grain of

I want to have something for the kids to play with which just records until silence is detected, plays back what was recorded, then repeats. They are having fun with Echo() at the moment :) I have mocked something up with: exten => *93,1,Answer exten => *93,n,Record(/tmp/echo:alaw|1) exten => *93,n,Playback(/tmp/echo) exten => *93,n,Goto(2) But it has the shortcomings that a beep is

> They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and > 'worst' readings of 99.975586% and 99.963379% respectively. Might want to give PCI latency setting a try, it helped for me. My ZTTEST would drop occasionally to 99.95% until I set: setpci -v -s 01:01.0 latency_timer=ff

>Does IAXmodem allows you to receive faxes with any extensions >(auto-detecting incoming faxes). You just let Asterisk do the fax detection for you, and when it hears CNG, send it to the fax extension, and your fax extension would just Dial() one of the IAXmodems (using IAX) >>DRi@b-w-computer.de wrote: >> sure in an small office you can use iaxmodem/hylafax to receive faxes

I just experienced something I'd rather not experience again. Using a SPA-841 SIP phone connected to our Asterisk server, someone dialed their own extension, answered, and then transferred the call using the phone's "XFER" soft key. This does a SIP REFER. Now, the phone has dropped out of the loop, and Asterisk has connected the two call legs into a loop, as far as I can tell.

Sangoma support did something and the driver is there. Rebooting now with the new card. Hold me, I'm scared. -----Original Message----- From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Friday, December 22, 2006 10:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sangoma Wanpipe 2.3.4-3 compilation fails un der FC2

If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP.