search for: 2201

...andard ones that happen with a fresh install. Does anyone know if this performance degradation is normal? If not, does anyone have hints on what I should do to track down the problem. Thanks. Ray ----------The log of my runs is shown below------------ # time dd if=big2 of=/dev/null bs=1024k 2201+1 records in 2201+1 records out real 23.8 user 0.0 sys 11.1 # time dd if=big2 of=/dev/null bs=1024k 2201+1 records in 2201+1 records out real 24.5 user 0.0 sys 8.9 # time dd if=big2 of=/dev/null bs=1024k 2201+1 records in 2201+1 records out real 27....

...chgn Property Rpr Rcil LLCI 12 2036 12.190220 UNSOLD 0.9999237 12.190220 16.09097 13 2036 14.741559 UNSOLD 0.9992714 14.741559 16.09097 14 2036 15.882518 UNSOLD 0.9955750 15.882518 16.09097 15 2036 16.090965 UNSOLD 0.9807892 16.090965 16.09097 34 2201 -6.542363 UNSOLD -1.0000000 NA NA 55 2201 -5.060431 UNSOLD 0.9848659 -5.060431 NA 56 2201 -5.056231 UNSOLD 0.9699841 -5.056231 NA 57 2201 -4.994895 UNSOLD 0.9441028 NA NA 58 2201 -4.982279 UNSOLD 0.9...

...ine. I alias alsa device hw:1,0 to card1 in /etc/asound.conf to get rid of the pesky ',' (see below) Any ideas on why this would happen? I am using Asterisk 1.0.5 on Debian testing (package version 1:1.0.5-2, which is the latest) --snip SIP to Console-- -- Executing Macro("SIP/2201-621f", "stdexten|1234|ALSA/card1") in new stack -- Executing Dial("SIP/2201-621f", "ALSA/card1|20") in new stack Mar 17 12:24:33 WARNING[23967]: channel.c:1901 ast_request: No channel type registered for 'ALSA' Mar 17 12:24:33 NOTICE[23967]: app_dial.c...

My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 & 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call out from my BT100 to other extensions and through the SPA to t...

I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/2201@gs1.uucp,20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as unavailable. This one does not work: exten =...

Hello all, I HAD video working before I upgraded to 1.08 (latest stable with Gentoo) and now it won't work. I just see noise bars and not the video. I know the camera works as I can use it in other programs such as AIM & Yahoo. I have the following setup: sip.conf [general] videosupport = yes port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind

https://bugzilla.mindrot.org/show_bug.cgi?id=2201 Bug ID: 2201 Summary: -R tunnel disappears Product: Portable OpenSSH Version: 6.1p1 Hardware: Other OS: Linux Status: NEW Severity: normal Priority: P5 Component: ssh Assignee:...

...outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed to rgadwell I'm seeing no sip traffic whatsoever on asterisk. My aim is to have inbound calls ring SIP extension 2201 I'm guessing this is something pretty straightforward, but any help would be much appreciated. Thanks, Russell. sip.conf [general] context=incoming ; Default context for incoming calls register => 7960xxx:yyyy@sip.gradwell.net/2001 register => 9479xxx:zzzz@sip.talklite...

Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': Found Sheriff*CLI> Disconnected from Asterisk server -- Dave Cotton <dcotton@linuxautrement.com>

...ody on pulver.com than it works. When I do a count <one, two, three, four, five, six,...> the person receives: <one, two, ..., ..., fi..e,...ten> . Same other way around. The connection will still be alive. Here my * output with (asterisk -vvvvvvvgc): -- Executing SetCallerID("SIP/2201-f8d6", "password") in new stack -- Executing SetCIDName("SIP/2201-f8d6", ""username"") in new stack -- Executing Dial("SIP/2201-f8d6", "SIP/612@fwd.pulver.com") in new stack -- Called 612@fwd.pulver.com -- SIP/fwd.pulver.com-29d8 is r...

...o_ks callwaiting=yes callwaitingcallerid=yes cancallforward=yes callreturn=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=0 immediate=no musiconhold=yes usecallerid=yes callerid="Analog Phone" <2201> mailbox=2201 channel => 2 Does anyone have any suggestions on where to start looking? Scott

I have Centro 5.4 it named starts at bootup but fails to restart when I give command service named restart and it there is no any error in syslog messages. With reload command I get the following message in syslog :- Nov 25 14:03:30 unitedinfotechs named[2201]: using default UDP/IPv4 port range: [1024, 65535] Nov 25 14:03:30 unitedinfotechs named[2201]: using default UDP/IPv6 port range: [1024, 65535] Nov 25 14:16:03 unitedinfotechs named[2201]: loading configuration from '/etc/named.conf' Nov 25 14:16:03 unitedinfotechs named[2201]: using defau...

...roxy-out] ;type=peer ; we only want to call out, not be called ;secret=guessit ;username=yourusername ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! ;host=box.provider.com ;------------------------------------------------ ; Test Ext 2201 ; <extension use> - <users name> - <extension number> ;------------------------------------------------ [2201] type=friend host=192.192.192.220 context=home secret=xxxxxx callerid="Paul" <2201> mailbox=2201 dtmfmode=rfc2833 nat=no EXTENSIONS.CONF writep...

Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (Context Extension Pri ) State Appl. Data IAX2[iaxtel]/1 ( s 1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 18888476626 1 ) Ring Dial IAX2/XXX:YYYY@iaxtel.com/18888476626@iaxtel But I never hear a ringing on the actual phone, and it seems to stay in this state (i.e. never gets to bridge mode) for a long time..to a point that ijust hang up. Thanks, Mark

...; Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf: [incoming] exten => 2933975,1,Goto(fax,2201,1) [fax] exten => 2201,1,Macro(faxreceive) [macro-faxreceive] exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) exten => s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN}) exten => s,3,rxfax(${FAXFILE}) exten => s,4,Congestion exten => s,103,...

...roxy-out] ;type=peer ; we only want to call out, not be called ;secret=guessit ;username=yourusername ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! ;host=box.provider.com ;------------------------------------------------ ; Test Ext 2201 ; <extension use> - <users name> - <extension number> ;------------------------------------------------ [2201] type=friend host=192.192.192.220 context=home secret=xxxxxx callerid="Paul" <2201> mailbox=2201 dtmfmode=rfc2833 nat=no EXTENSIONS.CONF writep...

Dear all, I am looking for software sip phone and hardware sip phone for our network with great quality. Need your suggestion. Thank you. Best regards IT Department Director of Information Technology Albert Chong 562-695-8823 Ext. 2201 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040604/787bfc7f/attachment.htm

...LLERIDNUM}) lets assume I called extension 403 from my extension 2204. then a caller (extension 2203) enters into the techsupport queue I am able to receive the support call on my phone (extension 2204 rings). I take the call and am talking with the customer (extension 2203). Now if extension 2201 calls into the techsupport queue, The problems rears its ugly head. One would think that the caller (extension 2203) would wait in the que until I am finished with the current caller (2203). However, My Phone (2204) rings a second time while I am on the phone with the first caller. I then re...

...every package needed. I've done everything on this page (altough, some bash-scripting problems): http://www.voip-info.org/tiki-index.php?page=Asterisk+Fax+to+email anyway, when i try to send an fax, i get theese messages in asterisk: -- Executing Goto("SIP/5060-08148520", "fax|2201|1") in new stack -- Goto (fax,2201,1) -- Executing Macro("SIP/5060-08148520", "faxreceive") in new stack -- Executing SetVar("SIP/5060-08148520", "FAXFILE=/var/spool/asterisk/fax/1126714845.5.tif") in new stack -- Executing DBget("SI...

The IAX2 channel is to IAXmodem. The SIP extension is an ATA with a fax attached. Nov 28 15:30:20 DEBUG[2997] chan_sip.c: build_route: Contact hop: Nov 28 15:30:20 VERBOSE[3276] logger.c: -- SIP/2201-090995f0 answered IAX2/24729-2 Nov 28 15:30:20 DEBUG[2995] chan_iax2.c: Ooh, voice format changed to 8 So what does this mean? The fax works just fine. I am just trying to tune up my dialplan.