similar to: SipTone II and Choppy/Stuttering Audio

-----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

hi, I recently changed my filesystem to ext3 (from ext2), and since then, my hard disk sometimes seems to 'hang' a few seconds (in fact the whole system hangs then, and only the hd LED is on), before continuing normally... This happens for example (not always though) when I untar an archive and try to ls in the newly created directory... Is there an explanation for this? And if so,

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

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

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>

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, 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

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

Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, Could somebody please help me to understand the following: - We have 8 msn's 383590, 383591 etc. What I would like to do is for the person at extension 1 dial out on 383 590, the person at extension 2 dial out on 383 591 etc. I have got myself so confused that I need major help!!! If you could give me a simplistic example, including which files I put the coding in (i.e.

Hi All, Many thanks to Marc who helped me with a previous Capi Dialout plan - however..... What I now would like to be able to do is: - We have 8 msn's 383590, 383591 383592 etc. What I would like to do is set up an If Then Else type statement along the following lines: - If extension 7957 Then Dialout on Capi msn 383590 ElseIf extension 7958 Then Dialout on Capi msn 383591 ElseIf

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

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Hello ext3-users, We have a RH71 machine running 2.4.16 kernel with e2fsprogs 1.25. I noticed many of these errors in our logs. EXT3-fs error (device sd(8,1)): ext3_readdir: bad entry in directory #884828: directory entry across blocks - offset=0, inode=404600689, rec_len=23080, name_len=59 EXT3-fs error (device sd(8,1)): ext3_readdir: bad entry in directory #966714: rec_len % 4 != 0 -

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

Hi there, I recently found a workaround for the "stuttering legs" bug in WoW, which is currently described in AppDB. It should work for everyone using gnome - so, since this is widespread, I thought it would be a good idea to share the workaround with you. I noticed that the running animation is all okay when activating auto-run - and that holding down the "run forward" key

I just installed 1.3.20 from the PPA on my Ubuntu 11.04 64-bit installation. One of the new features is the "Option to clip the mouse inside fullscreen windows". I wonder if this is related to the problem that has now cropped up since 1.3.19. In World of Warcraft when I move my mouse the position resets at constant intervals (~ 2 seconds). (When moving the mouse ever two seconds the

Hi, I've been getting reports of stuttering audio in my VoIP application that uses Speex 1.1.11. Since I haven't had this report with previous versions of Speex I was wondering if one of the bug fixes in 1.1.12 solves a bug which causes the stuttering. To me the stuttering report sounds like the decoder state that speex uses gets messed up. Some info, I use the floating point version of