similar to: FW: ipDialog Ethernet SIP Phone $199

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

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

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

Hi, I recently upgraded samba from 3.0.11 3.0.20. It's integrated with squid 2.5 STABLE 7 with patches for NTLM bugs. I am facing problem with NTLM authentication. The browser hangs and I get following error in log.winbindd [2005/09/02 13:53:10, 0] nsswitch/winbindd.c:process_loop(803) winbindd: Exceeding 200 client connections, no idle connection found [2005/09/02 13:53:10, 0]

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

I just got 2 IpDialog phones for use with my Asterisk system. I have been able to get the phones to just dial local extensions but it is not able to register with my system correctly. I would like to know if someone has set these phones up before and how they did it! Is there any examples for use with Asterisk? They seem simple enough to config with there web interface. Thanks

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

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

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 :

Hey Folks, I'm selling my SipTone on eBay... starting at $100, 17 hours left. It's been modified (the firmware) so that you are able to telnet into it and possibly (thanks to cross compiling) run your own software on it. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5711945656&ssPageName=STRK:MESE:IT Just so this post doesn't seem all to be about selling it

Hello, I am interested in knowing what are the advanced topics that can be learned in Asterisk. It would be helpful if there are any reference books or tutorials on Asterisk that cover advanced concepts on Asterisk. Thanks in advance! A Successful Person Is The One Who Can Lay A Firm Foundation With The Bricks That Others Throw --------------------------------- Why delete messages?

Hi, I''m looking for a star rating component for RoR, a bit like Votio (http://redalt.com/downloads/ - find the votio heading) or the star rating used on Amazon. I don''t really need the AJAX capabilities, just the ability to bind the results to a hidden drop down, or radio inputs. Multiple raters per page is also an issue. Any recomendations? -- Posted via

I use Centos 4.2 with all service pack installed. I verified traffic on link WEB and I see port TCP 80 with many traffic. I accessed lod /var/log/httpd/access_log and show below. ca.com/members/index.php HTTP/1.0" 401 - "http://members.sapphicerotica.com/members/index.php" "Mozilla/5.0 ( compatible; MSIE 5.01; Windows XP; NetCaptor )" 68.119.110.138 - -

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

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Dear List, Can anyone recommend a sip phone for residential use? (asterisk home pbx) Thanks!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040819/7b107ebc/attachment.htm

Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike

-----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

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Hi,  I tried fitting loglinear model using the glm(catspec). The data used is FHtab. . An independence model was fitted. Here summary() and fitmacro( ) give different values for AIC.   I understand that fitmacro( ) takes the likelilhood ratio L2(deviance) to calculate AIC and uses the formula AIC= L2- d.f(deviance)*2 and this AIC is used for comparison of nested models. (Am I right?)   The value