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Hello, I am wondering how to send a call to a specific IP address that is different than the host of the URI. For example, an invite to the URI is " john at phone.com" needs to be sent to the IP address 123.456.789.255, not to the IP address of phone.com. How is this done? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Day list I am looking to buy a few Wooksung Video phones to try with my asterisk box.... http://www.wooksung.com/eng/html/pro/pro_001.html has anyone had any experience using these with asterisk? Thanks Ron

Can anyone please direct me to UK based suppliers of equipment. Website URL's would be appreciated. TIA -- ***** Not everyone is touched by an Angel.... .... Those that are, never forget the experience *****

...lt;lee.redmayne@nwva.org> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] UK Suppliers > Date: Sat, 13 Sep 2003 13:11:52 +0100 > Reply-To: asterisk-users@lists.digium.com > > I bought some Snom phones which work nicely with Asterisk from: > > ProVu Communications Ltd > Bank House > Marsden > Huddersfield > HD7 6BR > > 01484-840048 > info@provu.co.uk > www.provu.co.uk > > -----Original Message----- > From: Angel Gabriel > Sent: 13 September 2003 13:02 > To: * Users > Subject: [Asterisk-Users] UK Suppli...

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

Hi, Is FAXing with Asterisk a practical option ? Or is it better just to use a plain fax connected to an FXS and just switch with Asterisk. I specifically wanted to know if there was any experience using just the fax scanner to send faxes and receive them via asterisk and the to e-mail. My idea was to take my old fax connect it to an FXS port and send faxes with the fax machine (using the fax

Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -------------- next part -------------- An HTML attachment was scrubbed...

Dear Expert, I am stuck when trying to register SPA-3102 on AsteriskNow .. could any body please advise .. where can I find the article for doing this? .. I googled but got nothing.. Regards bie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071203/6bbbecd8/attachment.htm

Hi All, What is the going rate for the Snom 100 in the UK? I've found a couple of suppliers with prices around the ?170 (exc vat) mark. Regards, Nathan.

Hi All, Can anyone recommend a supplier and/or a particular a model of SIP voip phones in the UK? I don't seem to be having much luck locating any. Thanks, Nathan.

Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. I don't know anything about electronics and would like to

Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). ---------snip------------------ == Using SIP RTP CoS mark 5 -- Executing [10000 at

My wife and daughter, and to lesser extent myself and my daughters boyfriend would like a communications system which allowed us to talk to each other, both on a one on one basis, but also occassionally in conference. My wife and I live in a house with an internal LAN with each of us with a desktop machine (hers in Windows XP, mine runs Linux) and a Linux server acting as firewall and NAT

Hi all, I have had (what I consider) an odd request. The installation I'm working on now is an office on a multi-floor building. They 're looking for some kind of solution with the phone system to provide door control. We are a non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius

We have a client that has sporadic problems with the *8 pickup facility. The server they are using is 1.8.5 and they are using Snom phones. Every now and then when they try to do a pickup from another phone they get a forbidden message on the phone and I can see the following in the logs. [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL [Aug 8 11:51:53] ERROR[19314] astobj2.c:

...ers Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <442fbb120708200019n4807cd08w8465a978edd022dd at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > 2007/8/13, Paul Hayes <paul at provu.co.uk>: > > > > > > It's not currently possible but Siemens are working on new firmware for > > at least the S450IP model which will support auto-config using http. > > I'm not sure when it's due for release though. > > > Thanks for the tip ! &g...

Hi, I have a SIP phone which does not natively support SIP transfers (REFER etc...). So far all that is possible is to enable blind transfers using the t and T arguments in Dial from the # DTMF key. The phone has an R button on it and this can be setup to either send an RFC2833 hook flash message (value 16) or a SIP INFO message which you can edit the contents of (since there seems to be