similar to: SipTone 4 Sale...

I'm just wondering if anyone knows the story with these... http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5721202362&ssPageName=STRK:MEWA:IT He claims they support IAX2 and SIP... but almost no history on the account selling them. I didn't see anything in the wiki about this company either.. Does anyone have any history with these phones? Thanks, Jared

Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM

Anybody used the above phone with asterisk I have one working ok for calls, but having a problem with voice mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? -- Clive Email :

Here is a SIP phone I haven't seen before. Does anyone have any experience with this one? -----Original Message----- From: George Richardson [mailto:georger@netxusa.com] Sent: Wednesday, April 02, 2003 4:56 PM To: clay@ctitec.com Subject: ipDialog Ethernet SIP Phone $199 pad <http://us.st1.yimg.com/store1.yimg.com/Img/trans_1x1.gif>

If receive a call on the IP Dialog SipTone II, and the other end hangs up first, the siptone immediately enters into the congestion tone. If I initiate the call from the siptone and the other end hangs up first, same thing -- congestion. The same thing happens if we make calls from the analog phones attached to the Mediatrix 1102. This does not happen on our Snom 200 phones, which have

On Wed, Mar 16, 2011 at 04:58:45AM +0100, olafBuddenhagen at gmx.net wrote: > Hi, > > On Tue, Mar 15, 2011 at 11:48:16AM -0600, Rob Savoye wrote: > > > In master, we need to do something to make this work correctly, and I > > think forcing everyone to clone swfdec is not a good idea. At least > > not the way it "works" now. > > Perhaps it would be

Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi

Hello Everyone, For the past few days I have been working on porting asterisk to the win32 platform using Cygwin. Asterisk is already capable of being loaded itself, but the modules are presenting a problem. On Windows a DLL can not access any functions of the program it was loaded from, which asterisk modules like to do. I have however come up with a crazy idea of turning asterisk into a DLL

I was wondering if anyone has seen anything like this standby power supply for Dell servers. http://www.ebay.co.uk/itm/200824909297?ssPageName=STRK:MEWNX:IT&_trksid=p3984.m1497.l2649 I made up a cable and can connect to it at 9600 baud 8N1. The problem is that all it ever does is respond with a "?" and ends it with a LF it does this whenever it recieves a LF - I'm begining to

Hi All, * is working very well for us now. But I have an issue that I cannot find the answer to - enter guru's!! When our receptionist does a blind call forward I receive the Caller ID, however I do not know if the call is fresh (i.e. ringing in) or forwarded. What I would like to do is to have * prefix the CID External (so that I can tell that it is a fresh call) or Internal (to tell me

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=230492577678#ht_500wt_1076

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, May 25, 2004 5:30 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or

I am using asterisk 1.6.1.6 and have been setting up a system to use a Postgresql database as the realtime DB via the ODBC route. I have got extensions and voicemail working but am having trouble with SIP The problem seems to be with using a schema. If I put the table "sip" in the schema "foo" then I add this entry to extconfig.conf sippeers => odbc,psqldb,foo.sip Restart

I am using a IPDialog siptone II. I can take incoming calls, but when I try and make an outgoing call I get a SIP 407 error. Can some kind soul explain to me what I am doing wrong? Here's what I found in the wiki: If a proxy does not accept the credentials sent with a request, it SHOULD return a 407 (Proxy Authentication Required). The response MUST include a Proxy-Authenticate header

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

Are there any IAX speaking "hardphones" out there? If so, can anyone offer comment on their quality? Thanks! -Dorn

I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

Unfortunately I can''t make the June 22-25 conference in Chicago. If interested in purchasing one 4 day pass for $400 please contact me at jonlivingston at gmail dot com or call me at 904.686.0056. -- Posted via http://www.ruby-forum.com/.

I just purchased this ticket yesterday from another list member here for regular list price. I had some terrible family events come up late last night, and will not be able to attend. This comes at a poor time for me as I was working with some great guys getting a project built for RailsConf. The ticket is just the same price as normal, and what I paid, $400. Please let me know if interested.