search for: spa921

Displaying 12 results from an estimated 12 matches for "spa921".

Did you mean: spa922
2007 Oct 18
2
A linksys SPA921 behind NAT and firewall
I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is...
2009 Dec 07
0
SPA921 Help
Hi, We've got 6 Linksys SPA921s. Does anyone know how to get the "Call Park" soft button to appear? During a call, it is not on any menu. Currently, we have to dial the access code manually. Also, the manual mentions that the user menu has an option labelled "Call Park Status" but it's not in any of the...
2011 Mar 05
1
2 ip phones and 1 normal, can't neither send nor receive calls at all...
I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines. 1.- spa921, 401 ext 2.- spa921, 402 ext 3.- normal phone connected to spa8800 404 ext. It had a very strange behavior when I was configuring call transfer and call pickup. These are steps to repeat it: 1.- from 401 c...
2009 Oct 18
4
Customising Firmware
Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm
2009 Oct 14
3
Extension Paging
Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.d...
2011 Feb 12
11
SIP Hardphone that works well with asterisk
Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110212/ccd9d985/attachment.htm>
2007 Jun 06
4
Best Codec
...rimarily DSL as internet connectivity? I know that g729 is the king-all, but I want to know what the rest of the professional are using out there. g729 has a cost involved, so does the cost really offset the performance? Or is it better to go with g711 to start off? We plan on using Linksys SPA921 as the primary phone and asterisk open source as the softswitch. Any information you can pass would be appreciated.
2010 Apr 29
3
Calls Dropping
...tp data coming from the remote location stops arriving at my sip server. So after about 30 seconds, the call gets terminated by my provider because i'm not sending any rtp packets to them. Any ideas why the rtp data should stop coming in, and how can I resolve it? Asterisk v1.4.30 6 x Linksys SPA921 Router at remote site is a Thomson TG585v7 Any assistance will be greatly appreciated. Many thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100429/3c2d057f/attachment.htm
2007 Jun 12
0
On multiple dial phones continue ringing after picked up
...Dial(SIP/4029&SIP/4030,15,tTr) As soon as one of the phone is picked up, all the others should just stop rining. But the fact is that they continue to ring for several seconds (4-5s), and this is quite annoying as all phones are in the same room. Phones are Siemens Gigaset C450IP and Linksys SPA921. Do you have any idea what I could do to fix this ? Regards, Yves.
2007 Dec 18
1
Dropped Calls
Hi all, I have a problem with some asterisk boxes. I have a standard installation with 1.4.14 (I also test with 1.4.4) in core duo Mac Mini on Debian Etch. I use SJphone softphone, Linksys SPA921 or Thomson 2030 for phones. All my phones are in a LAN with good status of 2ms max. Randomly I have dropped calls during communication. No absolutetimeout or other calling limitation options. Any ideas on how to solve this problem? Thanks in advance, Jeremy
2009 Feb 19
0
sip phone cant hear the caller
Hi Im using a sip phone SPA921, and the one that calls me can hear me but I cant hear them, when I make the call I can hear them. Im running asterisk 1.6 behind a firewall, I have port 10000-20000 for rtp and 5060 for sip forward to my asterisk. Any tips? Regards /ralf ________________________________________________ Ralf Tr...
2007 Oct 23
0
Internal Data Stream Error
...terisk-users-request at lists.digium.com You can reach the person managing the list at asterisk-users-owner at lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: A linksys SPA921 behind NAT and firewall (joakimsen at gmail.com) 2. Re: Making Asterisk a "Voice Router" (end1r) 3. Split asterisk in two ?? One TDM and One IP only?? (Steven) 4. Authenticate by IP? (Carlos Chavez) 5. Polycom 601 + Headset (Dovid B) 6. Re: tech prefix (Philipp Kempgen)...