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...t; (very annoying!) To configure the new phones you have to press and HOLD the up arrow whilst plugging in the power/ethernet, you should then get the option to select SIP. will let you know how I get on with the 5220 with asterisk as and when I get my phone :-) Sam Colin Anderson <ColinA@landmarkmasterbuilder.com> wrote on 01/09/2004 16:36:24: > > > > I will get a packet sniffer on one in a minute.... > > Don't bother, won't work. I already tried. Spoke to some Mitel mucky-mucks > too, and they said nope. You have to get Mitel's SIP-specific phone which *...

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

...ing 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: asterisk-users@lists.digium.com Subject: Re: SNOM 190 '486/Busy here' after upgrade to re 3.60s Hi Colin, There are a few reasons why a phone would deny a call with reason=busy: 1. If redirection is somehow on without a redirect target set. An incoming call in this scenario can...

Hi there, anyone in the community has manage to configure click to call features? Care to share. I have tried on this manual , seem got some software error like http://www.voip-info.org/wiki/view/Asterisk+click+to+call Software error: > > Unable to determine call statusMessage: Originate with 'Exten' requires 'Context' and 'Priority' > > For help, please

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,

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

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()

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

.... Thanks, -- -- -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: caloi@usadatanet.com <mailto:caloi@usadatanet.com> -- -- -- _____ From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Friday, February 17, 2006 10:42 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] A unique 'click to call' project - Could usesome advice You create a context in your dialplan that accepts the DID to call as a variable using...

...service can work with Asterisk. We tried setting it up like how you would for iconnecthere However, we even failed to register in the first place! (Of course password and username are correct). Anyone else on the list successfully used Primus' SIP with Asterisk? David >>> ColinA@landmarkmasterbuilder.com 1/20/2004 12:25:50 PM >>> Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20...

...service can work with Asterisk. We tried setting it up like how you would for iconnecthere However, we even failed to register in the first place! (Of course password and username are correct). Anyone else on the list successfully used Primus' SIP with Asterisk? David >>> ColinA@landmarkmasterbuilder.com 1/20/2004 12:25:50 PM >>> Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20...

> 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

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

...Thanks, -- -- -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: <mailto:caloi@usadatanet.com> caloi@usadatanet.com -- -- -- _____ From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Friday, February 17, 2006 12:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] A unique 'click to call' project - Could usesome advice Same as before but instead of SIP as the origination channel you pass ZAP/g0/XXXXXXXXX...

I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

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

Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20 Cdn a month!! Local number portability!! Cheapo Primus LD rates!! They don't care where geographically you plug it in!! When you sign

>how does the phone know where to find the TFTP server..? Dude, option 150 in your DHCP server: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186 a00800942f4.shtml We use the same option for our Mitel phones. HTH.

If you are a Canadian VoIP provider or CLEC, help make a difference by joining the CAVP: www.cavp.ca >OMFG, >I thought April Fools day was over. This is hard to believe. If true it >tells me that Bell has not changed at all. They are still trying to >manipulate and take advantage of the parts of the market they have absolute >control over. The CRTC was right to continue to

We use Motorola v551's as "extensions" on our Asterisk system with a homebrew find me/follow me dialplan. It works great except where coverage is poor then of course the inbound call hits voicemail. This has nothing to do with Asterisk and everything to do with our cellular provider, but since you guys are telephony pros I'd like to ask if anyone has had any good or bad