Re: [Asterisk-Users] SNOM and 1.0.9
This feature has worked for us since ver 1.0 (not cvs) Alvaro Parres wrote: Josheph: I had have that problem, and it get solve when i take out the incominglimit from my sip.cfg Also if you send you sip.cfg and extensions.cfg will be easier to help you Tray it. Alvaro Parres On 11/28/05, *BJ Weschke* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 11/28/05, Kevin Hanson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Joseph Rothstein wrote: Greetings to all, I am trying to get the line lights on a SNOM 320 to work using 'hint' in extensions.conf. Unfortunately I have not been able to get it to work properly. Does anyone know for sure if the hint function works properly in 1.0.9? If anyone has gotten this to work properly under 1.0.9 please post a sample. This is definitely a 1.2 only feature. It is not in 1.0.9. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Return of experience : Asterisk more stablewith 2.6 or 2.4
I also have no trouble on production systems 2.6.9/10 Gentoo-dev-sources On Sun, 2005-01-16 at 15:14 +1300, Matt Riddell wrote: Brian West wrote: I have never had an issue with 2.6.9 with asterisk. I second that. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 - incomming call is answered but if i hang up asterisk never detects it
as the subject states I have a TDM400 that when a call is answered asterisk runs the dialplan even if i hang up it NEVER detects the hangup and I am also having a hard time with CID info I don't get that either. most of our production machines are PRI based and I have little experience with the TDM400 cards, ANY help would be appreciated. Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 answers the line all the time!
)
;exten = s,2,GotoIf($[${CALLREDIR} = 0]?5:8)
;exten = s,3,GotoIf($[${CALLREDIR} = 1]?5:8)
exten =
s,1,Dial(SIP/${ARG1},20,t,T) ; Ring
the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Goto(routing,s,1) ; If they press
#, return to start
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED]) ; If busy,
send to voicemail w/ busy announce
exten = s-BUSY,2,Goto(routing,s,1) ; If
they press #, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1})
[macro-operator];
exten = s,1,DBget(OPERATOR=single/${ARG1})
exten = s,2,Dial(SIP/${ARG1},20) ;
Ring the interface, 20 seconds maximum
exten = s,3,Goto(s-${DIALSTATUS},1); Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Goto(routing,s,1) ; If they press
#, return to start
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED]) ;
If busy, send to voicemail w/ busy announce
exten = s-BUSY,2,Goto(routing,s,1) ; If
they press #, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${GENERALMAIL})
On Sat, 2005-01-15 at 18:29 +1300, Matt Riddell wrote:
Michael George wrote:
On Tue, Jan 18, 2005 at 04:16:08AM -0600, Justin Carlson wrote:
no i was using line 1 for testing /w fxs module and i never changed it
back
Also, could you show us the contents of your [routing] context in
extensions.conf?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] is it possible to use a sp2000 for intercom/paging?
I need to know if this works and if so does anyone have a sipura config to post? I have looked and not found anything conclusive. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom hint for ZAP channels?
is the hint 99,hint,ZAP/1 supposed to work or how do I get the lights on the phones to display channels in use in addition to extensions in use? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom hint for ZAP channels?
no we have a tdm400 at this site does this still apply? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 answers the line all the time!
no i was using line 1 for testing /w fxs module and i never changed it back On Fri, 2005-01-14 at 07:43 -0500, Michael George wrote: On Mon, Jan 17, 2005 at 08:12:24AM -0600, Justin Carlson wrote: hi all, We have a TDM400 card with 4 wfo modules. now the modules load fine and when i start asterisk with on phone line connected it just starts spewing these messages: -- Starting simple switch on 'Zap/4-1' Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:51 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... but no one is calling. i have plugged in a analog phone and dialed out on this line before i used it for *. any help would be great. zapata.conf [trunkgroups] [channels] language=en context=routing group=1 immediate=no signalling=fxs_ks channel = 1-4 zaptel.conf fxsks=2-4 loadzone = us Is there a reason you have fxsks=2-4 in zaptel.conf rather than 1-4? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 answers the line all the time!
hi all, We have a TDM400 card with 4 wfo modules. now the modules load fine and when i start asterisk with on phone line connected it just starts spewing these messages: -- Starting simple switch on 'Zap/4-1' Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:51 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... but no one is calling. i have plugged in a analog phone and dialed out on this line before i used it for *. any help would be great. zapata.conf [trunkgroups] [channels] language=en context=routing group=1 immediate=no signalling=fxs_ks channel = 1-4 zaptel.conf fxsks=2-4 loadzone = us ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CPU priorities (nice?)
what is wrong with running asterisk with the -pg flags at startup? On Mon, 2005-01-03 at 19:13 +0200, Gilad Ben-Yossef wrote: Matt Schulte wrote: Had a good question for the list, it seems whenever I work in an Asterisk console or on the machine normally I get jitters on any audio going through it. Especially if you did file copies or a 'ps ax' for example. I was wondering if there was a proper way to 'nice' the asterisk proc's? Cisco does this for example to it's EXEC and icmp processes, I tried reniceing the asterisk processes with very bad results, especially when I/O (voicemail, etc) comes into play. I'm not swapping out or anything, ideas? Since VoIP is a real time activity, simple nice really isn't enough. What you should do is mark the Asterisk proccess as a real time task for the Linux kernel to schedule accordingly. You can do this with Asterisk by passing the -p option to the Asterisk command line. A warning is due here: real time priority scheduled tasks are not something to be toyed with. You need to be root to be able to turn on this feature (meaning you have to be running Asterisk as root). A bug in Asterisk, a problem with mpg123 or a red alert on a FXO card can very well leave your system completly non responsive - so use with care. Having said that, I've been running an Asterisk server on a machine which is also used as SOHO firewall and file server for year now and it works great. Hope this helps, Gilad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ????
double post ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is asterisk that unstable ????
what was wrong with logrotate? On Thu, 2004-12-30 at 10:57 -0500, Matt Gibson wrote: Hi Randy, Randy MacKay wrote: I do about 500 calls per day on average volume and about 750 on heavy volume and find it necessary to run a logger rotate every other day... other then that I can go on for a couple weeks until I need a full reboot. How do you rotate your logs? I have made a script to rotate mine, it's a little over complicated, but it works. asterisk is run as user, and logs are kept in /var/log/asterisk old logs are kept in /var/log/asterisk/old_logs crontab for root: # this is to rotate asterisk logs daily at 11:58 pm 58 23 * * * /etc/asterisk_logr.sh | mail - -s [asterisk] daily log rotate root asterisk_logr.sh: #!/bin/sh #Rotates log files for asterisk #variables today=`/bin/date +%m%d%Y` chown=/bin/chown mv=/bin/mv ls='/bin/ls -sh' #tell asterisk to do its thing echo echo --- echo # MESSAGES # echo --- /usr/sbin/asterisk -rx logger rotate echo # sleepy sleepy #sleep 2 #set shit up sourcef1=/var/log/asterisk/queue_log.0 sourcef2=/var/log/asterisk/event_log.0 sourcef3=/var/log/asterisk/asterisk_norm.log.0 sourcef4=/var/log/asterisk/asterisk_debug.log.0 sourcef5=/var/log/asterisk/screenlog.0 destf1=/var/log/asterisk/old_logs/queue_log.$today destf2=/var/log/asterisk/old_logs/event_log.$today destf3=/var/log/asterisk/old_logs/asterisk_norm.log.$today destf4=/var/log/asterisk/old_logs/asterisk_debug.log.$today destf5=/var/log/asterisk/old_logs/screenlog.0.$today #moveem to dest dir echo --- echo # QUEUE LOG # echo --- if [ -f $sourcef1 ]; then $mv $sourcef1 $destf1 echo - rotated $sourcef1 to $destf1 $chown root:wheel $destf1 echo - $destf1 file attributes set echo - file size: `$ls $destf1` echo else echo - no queue log to rotate echo - no queue log to give permissions to echo fi echo --- echo # EVENT LOG # echo --- if [ -f $sourcef2 ]; then $mv $sourcef2 $destf2 echo - rotated $sourcef2 to $destf2 $chown root:wheel $destf2 echo - $destf2 file attributes set echo - file size: `$ls $destf2` echo else echo - no event log to rotate echo - no event log to give permissions to echo fi echo --- echo # NORM LOG # echo --- if [ -f $sourcef3 ]; then $mv $sourcef3 $destf3 echo - rotated $sourcef3 to $destf3 $chown root:wheel $destf3 echo - $destf3 file attributes set echo - file size: `$ls $destf3` echo else echo no normal log to rotate echo no normal log to give permissions to echo fi echo --- echo # DEBUG LOG # echo --- if [ -f $sourcef4 ]; then $mv $sourcef4 $destf4 echo - rotated $sourcef4 to $destf4 $chown root:wheel $destf4 echo - $destf4 file attributes set echo - file size: `$ls $destf4` echo else echo no debug logfile to rotate echo no debug log to give permissions to echo fi echo --- echo # SCREEN LOG # echo --- if [ -f $sourcef5 ]; then $mv $sourcef5 $destf5 echo - rotated $sourcef5 to $destf5 $chown root:wheel $destf5 echo - $destf5 file attributes set echo - file size: `$ls $destf5` echo else echo no screen logfile to rotate echo no screen log to give permissions to echo fi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot transfer with Cisco or Snom
Toggle the break key in the web config on your snom and then the break/transfer key will actually be the transfer key. On Sun, 2004-12-26 at 09:09 -0500, steve szmidt wrote: On Tuesday 21 December 2004 10:36 pm, Tracy R Reed wrote: I am having a hell of a time with transfers. First the Snom issues: The transfer button on the Snom 220 does not work. I have read about Use the soft button! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disabling ! command
you could comment that portion out and rebuild? On Fri, 2004-12-17 at 13:15 +0100, Alessio Focardi wrote: Hi, since I run asterisk as root with a CLI open on TTY12 I was wondering if the ! (shell) command can be disabled from the config, for safety reasons it seems me usefully. Tnx for any help ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 190, led and shared lines with asterisk
we also have observed that you must reboot the snom phones EVERY time you reload the dial plan or restart the server we are using *Asterisk 1.0.0 On Fri, 2004-12-17 at 21:47 +0100, Joris Trooster / Interstroom wrote: In your extensions.conf create a hint: exten = 215,hint,SIP/215 On the snom phone(s) subscribe the button to: destination: sip:[EMAIL PROTECTED];user=phone Where 192,168.0.200 is the ip of your asterisk server. When extension 215 is called, the light on the subscribed button on the snom phones is light up. Regards, Joris Netherlands. On Dec 17, 2004, at 8:37 PM, Dee Lowndes wrote: Hi All, I am trying to setup my snom 190 so that the LED's light up when one of my shared lines are in use. e.g. Extension 2 should ring on the snom and the phone associated with extension 2 and I should be able to see if the phone associated with extension 2 is making a call on the Snom. I think this is achieved with hints but don't seem to be getting on very well with it, does anyone have an example or pointers. Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX client behind a NAT
I wonder if you can put can reinvite=yes in the iax2.conf file like we use in our sip.conf file to do what you are requesting. I believe it should tell the phones to do what you wish On Thu, 2004-12-16 at 17:40 +0200, CuPoTKa wrote: Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, it goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uniden UIP200
Thank you! I will try again tomarow On Tue, 2004-12-14 at 14:05 -0500, Leif Madsen wrote: On Tue, 14 Dec 2004 03:22:47 -0600, Justin Carlson [EMAIL PROTECTED] wrote: If anyone has a working unidencomm.txt and unidenMACOFPHONE.txt file Could you please post it. Hi Justin, I am using the UIP200 here at home. Find pasted my configuration files. Note that I haven't tested everything in the file, but basic functionality (inbound and outbound calling) definately works. As does the MWI. This is based on the 4.59a firmware. Different firmwares may have different configurations? unidenMACOFPHONE.txt -- # UIP200 Mass Configuration System Mac-based File # Notes: Lines start with '#' are comments # To leave a field value unchanged (as saved on local phone), leave value to blank. # To disable a field, use '-' as value # MAXIMUM FILE SIZE IS 10KB # Current Limitation: No spaces allowed for a setting's value # Version: BS.459a # Firmware. The items listed in this Firmware section must be in this order. # FirmwareVersion and FirmwareFileName only used if AutoFirmwareUpdate is YES # FimrwareFileName only used if FirmwareVersion differ from firmware ver in Flash AutoFirmwareUpdateYES #choices are YES and NO FirmwareFileName uip200_459aenc.pac FirmwareVersion BS4.59a # Sip Settings MyLcdDisplay 1001 MyDialNumber 1001 DisplayNameApartment 1406 UserNameForProxy 1001 PasswordForProxy 1001 UserNameForRegistrar 1001 PasswordForRegistrar 1001 # Programmable Keys. Key functionality must go before key values. ProgrammableKey1 OneTouchDial ProgrammableKey2 OneTouchDial ProgrammableKey3 OneTouchDial ProgrammableKey4 OneTouchDial ProgrammableKey5 TwoTouchDial ProgrammableKey6 DoNotDisturb ProgrammableKey7 VMA ProgrammableKey8 Mute # One and Two-touch keys. Must go after Programmable keys functionality definitions. # Refer to Programmable and Fixed Function Keys for usage guide # OneTouchKeyX value is used ONLY when ProgrammableKeyX is OneTouchDial OneTouchKey1 1000 OneTouchKey2 OneTouchKey3 OneTouchKey4 1601 OneTouchKey5 2001 OneTouchKey6 OneTouchKey7 8500 OneTouchKey8 TwoTouchDigit0 TwoTouchDigit1 TwoTouchDigit2 TwoTouchDigit3 TwoTouchDigit4 TwoTouchDigit5 TwoTouchDigit6 TwoTouchDigit7 TwoTouchDigit8 TwoTouchDigit9 # Hotline and vmwi numbers --Must be placed after OneTouchDial's HotLineNumber- VmaDirectCallNo 8500#value associating with VMA Programmable key. VmwiLampIndicatorEnable TimeDisplay Enable #end of file unidencom.txt - # UIP200 Mass Configuration System Generic File # Notes: # 1. Lines start with '#' are comments # 2. To leave a field value unchanged (as saved on local phone), leave value to blank. # 3. To set a field's value to empty, use '-' as value. # 4. To NOT overwrite user local settings of: programmable key, one/two touch keys, VMA #number, VMWILampIndicator, set OverwriteLocalSetting = NO. Default is YES. This #key will ALSO affect whether or not THESE settings in unidenMAC.txt be used. # 5. Any duplicate parameters exist in both unidencom.txt and unidenMAC.txt, MAC settings #will be used. # MAXIMUM FILE SIZE IS 10KB # Current Limitation: No spaces allowed for a setting's value # Version: 4.59a #Overwrite user local settings of programmable keys, one/two touch keys, vma settings #If set to no, these current settings on the phone will not be overwritten. OverwriteLocalSettingsYES # must be placed on top of config file # Sip Settings --If only ProxyServer needed, set OutboundProxy1/Port same as ProxyServer/Port ProxyServer 192.168.1.1# can be an IP address or FDQN ProxyServerPort 0 # 0 to use default port OutboundProxy1192.168.1.1# can be an IP address or FQDN OutboundProxy1Port0 # enter a port number or 0 for default (5060) Registrar1192.168.1.1# can be an IP address or FQDN Registrar1Port0 # enter a port number or 0 for default (5060) RegisterExpireSec 3600 RegisterRetrySec 90 Q_Param 50 RegisterExpireLimitPercent10 FailoverRetrySec 8 SipPort 5060 SRVRecordName - #_sip._udp.unisip.com # options are ON or OFF SessionTimerSupport ON # options are ON or OFF SessionTimerRefresher ON SessionTimerMin 60 TimerInterval0300 TimerInterval1150 # Audio Settings G711MuTxPacketLength
Re: [Asterisk-Users] Multiline / Console / Receptionist phone
we also use the 220 but with the additional button panel with great results! On Tue, 2004-12-14 at 07:25 -0600, Gerald J. Puhl wrote: Does this phone have LEDs showing lines in-use? Thanx! Gary P. Tracy R Reed wrote: On Mon, Dec 13, 2004 at 12:50:54PM -0600, Gerald J. Puhl spake thusly: I have been looking to see if this type of phone can be implimented in *. I have found nothing conclusive. Is any out there using a multiline / mutlifunction phone typically used by a receptionist for transfering / routing calls? I need to know how this is accomplished or what alternative exists for this. I am using the Snom 220 with the hint extension priority with success. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __ Prototypes Patterns Models Dies Fixtures __ Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Uniden UIP200
Hello all, I have a uip200 for testing and I can't seem to get the phone to register to my * server. I have configured the unidencomm.txt and the unidenMACOFPHONE.txt files and the phone tries to register but * comes back with a 403 Forbidden message in sip debug, the phone simply displays #3 Register error. I have snom 200/190's and grandstreams, these phones we hoped could replace the (less than quality) grandstreams. If anyone has a working unidencomm.txt and unidenMACOFPHONE.txt file Could you please post it. Cheers! Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Transfer on Snom 190
make sure that the extension dialed has the ability to transfer calls and that the break key function is set correctly in the snom web admin. otherwise when you hit the transfer key it simply drops the call On Tue, 2004-12-07 at 18:44 +, Asterisk wrote : I cannot get the transfer button to work on a Snom 190, I cannot get the # to work either. I have both working fine here. What version of firmware are you running on the Snom? What's your dtmfmode set to in your sip.conf? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Justin Roy Carlson [EMAIL PROTECTED] Lachnet Technical Support Services 1.651.405.4780 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Iaxy issue
Doesn't the IAXy device have a power adjustment like the sipuras? On Thu, 2004-08-05 at 18:16, Glen Hinkle wrote: For anyone interested, the banshee screen I was experiencing was due to my cordless phone. I used a normal corded phone without separate power it was fine. I suppose there was some type of power overload that the iaxy couldn't handle. -g On Tue, 2004-06-22 at 17:20, Andre Gironda wrote: I've had an IAXy for about 3 weeks and have not heard a banshee scream. I use it constantly, and have had cofigurations using a local Asterisk machine in my house as well as Gafachi and VoicePulse. VoicePulse was pretty awful. I am going to setup NuFone just so I can compare the 3. -Amdre ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and RT
Has anyone integrated asterisk with current version of rt. I followed the Wiki but I only get as far as hold on while i create a ticket then it hangs up. I don't see it connect to the rt-soap-server.pl script running on the console of my rt machine. any help would be greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Flah Operator Panel show iax2 trunk
We use an IAX2 trunk to our remote office and would like for the receptionist to be able to transfer incoming calls from this trunk. but all calls come in as one user, Is there a way to get a breakout on the flash GUI of the incoming calls? Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Flah Operator Panel show iax2 trunk
Thank you for the prompt reply but when I add 7;8;9, in my button number field the iax2 button goes away. i just got .10 today . On Mon, 2004-06-28 at 11:51, Nicolas Gudino wrote: Hi Justin, Justin Carlson wrote: We use an IAX2 trunk to our remote office and would like for the receptionist to be able to transfer incoming calls from this trunk. but all calls come in as one user, Is there a way to get a breakout on the flash GUI of the incoming calls? I'm working exactly on it right now. The way I am handling the IAX or any other VOIP trunk is maybe limited, but I couldn't find a better aproach. Basically, you can have one line in op_buttons.cfg for IAX users, like IAX2[guest] for Iaxtel. In the button number, you can add as many as you like, eg: 1;2;3;4;5;6. The server then populates the buttons as they are being used. If you have only one call, it will show it in button 1, if you have more, it will use the remaining buttons. If you exceed the number of buttons, the rest of the calls will not show up. This is working now, but only for showing info (in the online demo there are three iaxtel buttons, you can call 17005011506 to see it working). I have to work now on transfers and hangups. If time permits I will finish later today or maybe tomorrow. For anyone interested in Flash Operator Panel, there is a mailing list to discuss about it. You can subscribe sending a mail to [EMAIL PROTECTED] Best regards, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Module nonsense (zaptel, wcfxs and wxfxo)
I have had similar troubles and doing a modprobe -r zaptel then re-loading the zaptel modules seems to cure it. ( if you unload the wcfxo ztdummy wct1xxp etc it leaves the zaptel module loaded.) On Tue, 2004-06-08 at 07:25, Rich Adamson wrote: I've been playing with two pieces of hardware: a X100P and a TDM400P with an FXO and two FXS modules. I had been using just the TDM card; however, the TDM FXO module seems to hear things and answer the telephone for no reason, and I wanted to compare the results with an X100P card. Yes, same issue here. I'm not a programmer, so my comments are based on observations only. For whatever reason, the TDM card is far more sensitive to analog line activities then was the x100p. Simply taking a bridged analog phone off hook and back on hook causes the TDM card to assume the phone is ringing. Also, female voices on the analog side tend to be interpreted as ringing as well (that's with callprogress=no). If you want further details, I can give them to you, but suffice it to say that trying to work with both cards and both modules has been incredibly frustrating. Modules that won't load, or that load but don't work when you run Asterisk, or Asterisk segfaulting even though the modules *seem* to load properly... Observed the same here on RH v9. System has been mostly stable for over six months in terms of processing calls, but doing 'service zaptel stop' and start (after stopping *) leads to unpredictable results. Usually can get by with one or two, but anything after that leads to failures that require a system reboot. The exact number varies. Other observations tend to suggest echo cancellation is not any better on the TDM compared to the x100p, and the amount of echo seems to change from time to time with no noticable correlation to other system events. CallerID on the TDM seems to be less reliable then the x100p (more ID's showing up as 'asterisk' when a bridged analog phone receives the ID's just fine). I'm using three pstn lines from two different central offices and two different ring cadences. Inbound and outbound calls are processed correctly, and echo (and other unusual * activities) is the same across all three lines. (The TDM card runs on a dedicated interrupt, and CVS-HEAD-05/28/04.) Unplugging a pstn line from the TDM card (and reconnecting) creates some rather unusual events too. Guess its time to open a bug report even though we don't have any significant technical data that would help debug the problems. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF and SIP
have you tried commenting out the dtmf lines in your sip.conf we had similar problems with our snom 200's and after commenting out the dtmf lines in sip.conf asterisk reload they worked great :-) On Wed, 2004-06-02 at 11:36, Lee Norvall wrote: Hi I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also tried inband) and I get the following error: june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? This means that I cannot get access to voicemail from the handsets !!! Any clues??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI indicator on SNOM200 doesn't disappear
we have the same problem could you please send me the chan_sip2 info. Thanks! On Sat, 2004-04-24 at 14:23, Geert Nijpels wrote: Ian White wrote: On Apr 22, 2004, at 23:48, Olle E. Johansson wrote: Geert Nijpels wrote: Ian White wrote: On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report with snom, but they're claiming it is an asterisk issue and that it should have been resolved. They suggested that I ask on the list. Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to turn off the MWI. The message doesn't contain the line so the phone doesn't know which line to apply the messages to. Basically the NOTIFY message should contain something like the following: NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0 There was a bugfix for this in Asterisk for this problem, do you have that applied? I am running the current CVS version, and don't see anything in the code that looks like this has been touched, and I haven't seen reference to it on this list. They are right in that the line information isn't being sent, looking at the SIP debugs on both ends. Anybody have ideas? Ian This is a problem I have been digging into a bit. In my case asterisk did not send out the NOTIFY with the header Content-Type: application/simple-message-summary, but with Content-Type: text/plain, so the NOTIFY is treated as a txt message. In result, when I pressed the MWI button, I saw the text from asterisk stating the amount of messages I have. I changed it to work, and now asterisk calls the extension the message is sent from ([EMAIL PROTECTED]). After calling this the MWI indication disappears, I'm not sure if it also disappears after calling from another phone. I'm using chan_sip2 and I changed some stuff, so I'm not sure if this is also a problem with standard chan_sip (the txt vs vm issue). Chan_sip2 handles Contact: differently than chan_sip and works better with Snom phones. It's actually where the whole chan_sip2 project started... :-) Any idea what sort of time frame before chan_sip2 becomes usable in a production environment, or at least becomes part of the CVS tree? I see your note saying that you are using it in production. I'm using it with some changes with -stable. It's developed by oej for -devel. Works great with my SNOM's and Cisco 9760. You can get chan_sip2 through the bugtracker: http://bugs.digium.com/bug_view_page.php?bug_id=759 I can also send you my -stable version, but you can backport it with some minor trouble yourself. Geert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2.05a firmware
where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) also anyone got a fix for the horrible speaker phone on the 200's ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded problem...
I don't think your DTMF is set right look in sip.conf for the dtmf
directive for your phones.
cheers!
On Tue, 2004-05-04 at 13:41, Michael Picher wrote:
Searched the archives thoroughly...
Can't find this specific problem...
Simple setup with Asterisk on RedHat. No voice cards in the box, 2
SNOM 200 phones...
Phones seem to work well, can leave VM, Message Waiting Indicator
lights up but when I try to retrieve messages the call terminates and
the following happens:
-- Executing VoiceMailMain(SIP/520-a25e, Mike) in new stack
-- Playing 'vm-login' (language 'en')
May 4 07:58:07 WARNING[1125329600]: chan_sip.c:497 retrans_pkt:
Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 2
(Response
)
May 4 07:58:07 WARNING[1217602880]: app_voicemail.c:2748 vm_execmain:
Couldn't
read username
== Spawn extension (default, asterisk, 1) exited non-zero on
'SIP/520-a25e'
asterisk*CLI
Pertinent section of extensions.conf
exten = 504,1,Dial,sip/${EXTEN}|10
exten = 504,2,Voicemail(u504)
exten = 504,102,Voicemail(b504)
exten = 504,103,Hangup
exten = 520,1,Dial,sip/${EXTEN}|10
exten = 520,2,Voicemail(u520)
exten = 520,102,Voicemail(b520)
exten = 520,103,Hangup
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
Pertinent section of voicemail.conf
504 = 504,Tech Desk,[EMAIL PROTECTED]
520 = 520,Mike Picher,[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2
we are getting these errors too which cvs was it fixed in ? we just upgraded to cvs-stable from friday to see if that would help. On Sun, 2004-05-02 at 21:45, brian k. west wrote: I think this was fixed in CVS-HEAD because I do not see that message in the src at all while looking to see if t was fixed. bkw - Original Message - From: Serge Oleinikov To: [EMAIL PROTECTED] Sent: Sunday, May 02, 2004 2:40 PM Subject: [Asterisk-Users] IAX2 What does it mean ? May 2 20:37:21 WARNING[1205250992]: chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16386, dropping Asterisk CVS-05/02/04-23:04:14 built by [EMAIL PROTECTED] on a i686 running Linux from cvs checkout -r v1-0_stable asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2
no I usually have 2 to 3 calls going down a full data T1(only voice data) and I get this message and 2 sec later calls are dropped. we look at our bandwidth for that time and we were no where near full utilization. On Mon, 2004-05-03 at 13:58, brian wrote: 1. Its not an error. 2. It's a warning. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Justin Carlson Sent: Monday, May 03, 2004 3:21 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 we are getting these errors too which cvs was it fixed in ? we just upgraded to cvs-stable from friday to see if that would help. On Sun, 2004-05-02 at 21:45, brian k. west wrote: I think this was fixed in CVS-HEAD because I do not see that message in the src at all while looking to see if t was fixed. bkw - Original Message - From: Serge Oleinikov To: [EMAIL PROTECTED] Sent: Sunday, May 02, 2004 2:40 PM Subject: [Asterisk-Users] IAX2 What does it mean ? May 2 20:37:21 WARNING[1205250992]: chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16386, dropping Asterisk CVS-05/02/04-23:04:14 built by [EMAIL PROTECTED] on a i686 running Linux from cvs checkout -r v1-0_stable asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX firmware for snom 200s?
is there a firmware for IAX for the snom 200's. or are there any other hard phones that use iax(2)? Thanks in advance! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dropped calls
We also are having randomly dropped calls with our IAX2 connections, we have tried IAX2 with and without trunking enabled. the phones are snom 200's with SIP and there is an asterisk box at each site so no sip nat problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Thursday, April 15, 2004 11:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dropped calls Hi! I see this very same effect rather often in the following setup: SIP (GS101) -- *1 -- IAX2 -- *2 -- MGCP (ip10) In fact I think I've seen it also with SIP instead of MGCP at the end. The first client is behind NAT, by the way. That must be it. I have seen this happening with sip -- * -- IAX as well. I take it you don't know a cure? Unfortunately not, no. By the way I am not on latest CVS as that would disable my MGCP phones. And so far I didn't even get a chance to debug this since it happens approx 1 out of 10 calls only. By the way, I can now conirm that it can be both MGCP or SIP at the end, it doesn't matter. So to me it looks like IAX2 is involved as well, not just SIP. *1: CVS-02/10/04-16:49:37 *2: CVS-03/05/04-00:50:56 Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Out of trunk data space on call number 16386, dropping
how did you guys go about diableing it. Is it the threwaycalling directive in zapata.conf ? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Warren H. PrinceSent: Thursday, April 08, 2004 8:01 AMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Out of trunk data space on call number 16386, droppingI work with Tony, so I'm responding for him. Yes, it appears only during a conference call. So, if we disable conferencing, we do not receive the error.Justin Carlson wrote: if you disable conferencing does the problem go away? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Tony Buser Sent: Wednesday, April 07, 2004 2:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Out of trunk data space on call number 16386, dropping I'm having the same kind of issues. We get the out of trunk data space error consistently during conference calls between asterisk servers. And occasionally on regular iax calls. Also while we're on a conference call it seems to cause other calls going out through iax to fail and also give this error. (weather its to another asterisk server or through say oneunified) If you figure this out, please let us know here. I'm pretty much at a loss as to what could be causing it. Justin Carlson wrote: Hi all, We keep getting these and all the calls between these two asterisk boxes get dropped. what is going on here, I have been trying to solve this problem on my own but maybe I don't have the trunk setup right. also I have posed the output of my full log of the machine with the zap interface, the other is using ztdummy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Out of trunk data space on call number 16386, dropping
Hi all, We keep getting these and all the calls between these two asterisk boxes get dropped. what is going on here, I have been trying to solve this problem on my own but maybe I don't have the trunk setup right. also I have posed the output of my full log of the machine with the zap interface, the other is using ztdummy. IAX.conf on machine 1: [general] port=5036 ;iaxcompat=yes bandwidth=low disallow=ilbc disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=ulaw ;allow=gsm ; Always allow GSM, it's cool :) jitterbuffer=no trunkfreq=20 ;dropcount=3 ;maxjitterbuffer=500 ;maxexcessbuffer=100 ; tos=lowdelay register = [EMAIL PROTECTED] register = [EMAIL PROTECTED] ; [woodlane] allow=ulaw ;allow=gsm type=friend jitterbuffer=no username=woodlane context=dialout host=dynamic trunk=yes trunkfreq=20 IAX.conf on machine2: [general] port=5036 bindaddr = XXX.XXX.XXX.XXX iaxcompat=yes ;amaflags=default ;accountcode=lss0101 bandwidth=low disallow=ilbc disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=ulaw disallow=gsm; Always allow GSM, it's cool :) jitterbuffer=no ;dropcount=3 ;maxjitterbuffer=500 maxexcessbuffer=100 trunkfreq=20; How frequently to send trunk msgs (in ms) register = [EMAIL PROTECTED] authdebug=yes tos=lowdelay [lachnet] allow=ulaw disallow=ilbc disallow=lpc10 disallow=gsm jitterbuffer=no username=lachnet type=friend trunk=yes trunkfreq=20 host=dynamic ;secret=telco context=default include = dialout [woodlane] allow=ulaw ;allow=gsm type=friend jitterbuffer=no username=woodlane context=dialout host=dynamic trunk=yes trunkfreq=20 Full.log: Apr 7 09:41:21 DEBUG[704531]: Bridge stops bridging channels [EMAIL PROTECTED]/16385 and Zap/1-1 Apr 7 09:41:21 DEBUG[704531]: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Apr 7 09:41:21 DEBUG[704531]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Apr 7 09:41:21 DEBUG[704531]: disabled echo cancellation on channel 1 Apr 7 09:41:21 DEBUG[704531]: Set option TDD MODE, value: OFF(0) on Zap/1-1 Apr 7 09:41:21 DEBUG[704531]: Updated conferencing on 1, with 0 conference users Apr 7 09:41:21 DEBUG[704531]: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Apr 7 09:41:21 DEBUG[704531]: disabled echo cancellation on channel 1 Apr 7 09:41:21 VERBOSE[704531]: -- Hungup 'Zap/1-1' Apr 7 09:41:21 VERBOSE[704531]: == Spawn extension (dialout, 5522307, 1) exited non-zero on '[EMAIL PROTECTED]/16385' Apr 7 09:41:21 DEBUG[704531]: We're hanging up [EMAIL PROTECTED]/16385 now... Apr 7 09:41:21 VERBOSE[704531]: -- Hungup '[EMAIL PROTECTED]/16385' Apr 7 09:41:29 DEBUG[163851]: Made call 5 into trunk call 16386 Apr 7 09:41:29 VERBOSE[163851]: -- Accepting unauthenticated call from 65.113.15.19, requested format = 4, actual format = 4 Apr 7 09:41:29 VERBOSE[737299]: -- Executing Dial([EMAIL PROTECTED]/16386, Zap/g1/BYEXTENSION) in new stack Apr 7 09:41:29 VERBOSE[737299]: -- Called g1/5522307 Apr 7 09:41:29 DEBUG[163851]: Ooh, voice format changed to 4 Apr 7 09:41:30 DEBUG[114696]: Enabled echo cancellation on channel 1 Apr 7 09:41:30 VERBOSE[737299]: -- Zap/1-1 is ringing Apr 7 09:41:35 DEBUG[114696]: Echo cancellation already on Apr 7 09:41:35 VERBOSE[737299]: -- Zap/1-1 answered [EMAIL PROTECTED]/16386 Apr 7 09:41:35 WARNING[737299]: Out of trunk data space on call number 16386, dropping Apr 7 09:41:44 DEBUG[163851]: Made call 8 into trunk call 16387 Apr 7 09:41:44 VERBOSE[163851]: -- Accepting unauthenticated call from 65.113.15.19, requested format = 4, actual format = 4 Apr 7 09:41:44 VERBOSE[753684]: -- Executing Dial([EMAIL PROTECTED]/16387, Zap/g1/BYEXTENSION) in new stack Apr 7 09:41:44 VERBOSE[753684]: -- Called g1/5540408 Apr 7 09:41:44 DEBUG[163851]: Ooh, voice format changed to 4 Apr 7 09:41:46 DEBUG[114696]: Enabled echo cancellation on channel 2 Apr 7 09:41:46 VERBOSE[753684]: -- Zap/2-1 is ringing Apr 7 09:41:49 DEBUG[114696]: Echo cancellation already on Apr 7 09:41:49 VERBOSE[753684]: -- Zap/2-1 answered [EMAIL PROTECTED]/16387 Apr 7 09:42:04 VERBOSE[114696]: -- Channel 1, span 1 got hangup Apr 7 09:42:04 DEBUG[737299]: Bridge stops because we're zombie or need a soft hangup: [EMAIL PROTECTED]/16386, c1=Zap/1-1, flags: No,No,No,Yes Apr 7 09:42:04 DEBUG[737299]: Bridge stops bridging channels [EMAIL PROTECTED]/16386 and Zap/1-1 Apr 7 09:42:04 DEBUG[737299]: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Apr 7 09:42:04 DEBUG[737299]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Apr 7 09:42:04 DEBUG[737299]: disabled echo cancellation on channel 1 Apr 7 09:42:04 DEBUG[737299]: Set option TDD MODE, value: OFF(0) on Zap/1-1 Apr 7 09:42:04 DEBUG[737299]: Updated conferencing on 1, with 0 conference users Apr 7 09:42:04 DEBUG[737299]: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Apr 7 09:42:04
RE: [Asterisk-Users] Problems with IAX2?
dido -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Wednesday, April 07, 2004 2:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with IAX2? Hi I am also having jitter trouble on IAX2, and I can vouch that the jitter buffer is busted. On Wed, 07 Apr 2004 09:56:01 -0400 Steve Kann [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: Are there open problems/issues with iax2 and jitter (quality)? Just upgraded to today's dev cvs about an hour ago, and it seems the iax conversations are lower quality then a month or two ago. iax2 show firmware says version 13. (Test call originated from C7960 with g711.) I noticed the same thing. Jitter buffer apparently is broken, and has always been. I was advised to say jitterbuffer=no in iax.conf, but I swear it's better with it set to yes and then executing iax2 set jitter 250 in the CLI. At least it was before I cvs up'd. :-) I found a jitter buffer bug in IAX2 a short while ago. It could potentially lead to misordered frames in conversations, and does so quite often when the sender of frames is using iaxclient under win9x. I compensated for this with a change in iaxclient, but the problem could also happen in asterisk-generated frames. See : http://sourceforge.net/mailarchive/forum.php?thread_id=4096021forum_id=2938 0 I don't know if this is the bug people are hitting, or not, though. Jeremy (of NuFone fame) has his jitterbuffer=no on his servers and since he's my VOIP provider I tend to just try and match his setup in terms of IAX2 anyway. I dunno, I do agree with you that it seemed better a while ago. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Herbalife Independent Distributor http://www.healthiest.co.za ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need a list of asterisk built-in variables
I need to be able to use a variable that has the calling extension number rather than the called. thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need a list of asterisk built-in variables
Title: RE: [Asterisk-Users] Need a list of asterisk built-in variables
it
puts the callerid number I have in the sip.conf instead.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Pedro Bessa
GoncalvesSent: Tuesday, April 06, 2004 11:37 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Need a
list of asterisk built-in variables
The ${CALLERIDNUM} variable has the calling extens
number.
Regards, Pedro Goncalves
-Original Message- From:
Justin Carlson [mailto:[EMAIL PROTECTED]]
Sent: terça-feira, 6 de Abril de 2004 17:32
To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Need a list of asterisk built-in
variables
I need to be able to use a variable that has the calling
extension number rather than the called.
thanks.
___
Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mpg123 issue and solution
I have a suse 8.2 installation of mpg123 and I have no problems with the id3 tags -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Tuesday, April 06, 2004 11:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] mpg123 issue and solution Hi! If you put mp3 files into your mohmp3 directory and these files have ID3v2 tags, mpg123 will throw an error message Found new ID3 Header, regardless of the -q flag. This, in turn, will cause Asterisk to crash (yes), although it's a soft crash (exits cleanly). It took me forever to figure this out, since the default mp3 and everything else was working fine. And the lack of any meaningfull error messages made diagnosis even more difficult My work around was to open the file in WinAmp and remove the ID3 tags entirely. mpg123 and Asterisk were both happy and there was much rejoycing. See also: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf (as well as the Wiki Asterisk FAQ section concerning variable bit rate) It might be a good idea to move away from mpg123 as it is no longer supported and there are bound to be more problems like this. MAD seems to be what everyone is migrating to... Indeed mpg123 is known to be the cause for many problems. Cheers, Philipp Finally, if anyone has any ideas about how to improve IAX voice quality, I'd be happy to hear them. Everything is hearable, but there are an awfull lot of clicks and pops in the background. This is probably due to the IAX software phone that you are using (and its underlying library). On * server to * server connections this shouldn't be the case (if yes: try to enable/disable the jitter buffer, see other mails here). -Philipp- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need a list of asterisk built-in variables
The wiki did not seem to have the exact variable I need so I have got it
working now but it would have been nice to be able to have exten =
asterisk,1,Voicemailmain(s{callingexten}).
.and please read the previous post next time.
cheers!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Tuesday, April 06, 2004 12:12 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Need a list of asterisk built-in variables
Justin Carlson wrote:
I need to be able to use a variable that has the calling extension number
rather than the called.
It's on the WIki and in your source code tree.
Check the Wiki page Asterisk variables
http://www.voip-info.org
...and please try to read the docs before posting to the list. Thank you.
/Olle
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need a list of asterisk built-in variables
Title: RE: [Asterisk-Users] Need a list of asterisk built-in variables
yes
but this gives the entire phone number and there phone numbers do not match
their extension numbers for more reasons that I want to explain. (they wanted
the new system and the old one to behave the same) so I needed to get the
extension of the calling extension for the auto-voicemail login to
work.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Pedro Bessa
GoncalvesSent: Tuesday, April 06, 2004 12:27 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Need a
list of asterisk built-in variables
Suppose EXT1 makes
call to EXT2. Then the ${CALLERIDNUM} is the number of EXT1 while ${EXT} is
the number of EXT2.
Any
doubts?
Regards,
Pedro
Goncalves
From: Justin
Carlson [mailto:[EMAIL PROTECTED] Sent: terça-feira, 6 de Abril de 2004
18:01To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Need a list
of asterisk built-in variables
it puts the callerid
number I have in the sip.conf instead.
-Original
Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Pedro Bessa
GoncalvesSent: Tuesday,
April 06, 2004 11:37 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Need a
list of asterisk built-in variables
The
${CALLERIDNUM} variable has the calling extens number.
Regards, Pedro Goncalves
-Original Message- From: Justin Carlson [mailto:[EMAIL PROTECTED]]
Sent:
terça-feira, 6 de Abril de 2004 17:32 To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Need a list of asterisk built-in variables
I need
to be able to use a variable that has the calling extension
number rather
than the called.
thanks.
___
Asterisk-Users mailing
list [EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
if you don't give them the pass code they can't hang-up or transfer calls -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adams, Gavin Sent: Friday, April 02, 2004 7:30 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel -Original Message- http://sip.house.com.ar/operator I love these types of applications that show off the capabilities of *. This was easy to get up and running for my SIP channels, but for some reason my PRI (ZAP/1 through ZAP/6) aren't showing up. Has anyone else got this working for SIP - Trunk lines? You can also perform some actions. Hang-up channels and Transfers via drag and drop. How hard would it be to disable these functions. We have the need to show station status to our users, but would like to remove the ability to hang up other peoples calls. The difference with other similar tools is that it displays status in real time (no refreshing necessary), and its graphically appealing. Looking good too! Not being a Flash developer, can the .swf file be decoded? I'm thinking of changing some colors, making the buttons smaller, etc. to allow for more channels to be displayed. Keep up the great work! Regards, --- Gavin attachment: winmail.dat
RE: [Asterisk-Users] UNSUBSCRIBE
this is not where to send your unsubscribe to ! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Altus Snyman Sent: Friday, April 02, 2004 7:20 AM To: asterisk Subject: [Asterisk-Users] UNSUBSCRIBE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
just type it in it will remain until you restart your browser. ( it does not disappear and you do not have to hit enter or anything like that) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Capouch Sent: Friday, April 02, 2004 3:45 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel I downloaded the app and for the most part have it going. I have not yet managed to get it to accept the password in the flash widget that appears as if it wants to accept it. I wonder about browser-related problems in that respect: I'm running fairly recent Mozilla. I have also hacked the thing to watch my IAX phones and incoming lines. . I need to test a bit and will post my changes. Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
we also would require more buttons, at least 40, can we get a multipage view. right know I run multiple servers on the same page to get the effect of having 3 pages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nicolas Gudino Sent: Friday, April 02, 2004 8:26 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel Hi Eric, - Original Message - From: Eric Wieling [EMAIL PROTECTED] Sent: Friday, April 02, 2004 11:17 AM Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel Being able to have more buttons as well as changing the button size would be useful. What screen resolutions do you use, how many buttons do you need? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail
vmail.cgi seems to be written in perl so modifying it should require knowledge of perl and vi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Clifton Sent: Friday, April 02, 2004 10:51 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voicemail How would one hack the voicemail app to play saved vm messages back in a 'most recent first' fashion ? What source file is this defined in ? Thanks, Chris Clifton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo's and dropped calls
Hi all, I have a problem with echo and silence in the middle of calls. the echo problem is that in the first 5 to 10 seconds of a call there is echo on the sip side but not on the PSTN side, also the echo will randomly come back in the call sometimes, I'd say 3 out of 10 calls. the other problem I have is that sometimes ( like maybe 4 times a day ) we will be talking to PSTN calls and one side of the call will go silent and then 5 to 10 seconds later come back. we have this setup. PSTN -- Asterisk /w T100p -IAX2- Asterisk2 /w zapdummy -sip snom 200's, and locally in our office it looks like this PSTN -T100p- Asterisk -sip- snom 200's (the echo is on this setup as well) any help would be appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap Channels Hang
our cvs is 02/25/04 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Juan J. Sierralta P. Sent: Thursday, April 01, 2004 11:56 AM To: Asterisk Users Subject: Re: [Asterisk-Users] Zap Channels Hang On Thu, 2004-04-01 at 10:37, Sergi Gabunia wrote: Hi, I have same problem with zap channels. I have E100P installed on my asterisk box and I worked with CVS-02/22/04-16:30:20 and everything worked well (with Zap channels). I update asterisk to new cvs 2 days ago and incoming zap calls starts hanging. I have mgcp extensions defined in my extensions.conf and I see that if voicemail is enabled for extension and there are two concurent call (from Zap) to this extension, second call to voicemail are hanging in asterisk after user from Zap side hangs up. If there are no voicemail for extension the call are not hanging at all. May be these information will be helpfull to fix this bug. I noted the same problems with CVS from 03/30/2004 when incoming calls were sent to voicemail. Anyway I had to roll back to 03/05 since last Zaptel was giving me yellow alarms con my TE410P on a E1 PRI. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo's and dropped calls
on both the box with the zap interface and the remote office. it helped some but the problem remains -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Vogel Sent: Thursday, April 01, 2004 12:10 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Echo's and dropped calls Do you have echocancel=yes echocancelwhenbridged=yes echotraining=yes In your zapata.conf file? Wiki is good for this - John V. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson Sent: Thursday, April 01, 2004 9:45 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo's and dropped calls Hi all, I have a problem with echo and silence in the middle of calls. the echo problem is that in the first 5 to 10 seconds of a call there is echo on the sip side but not on the PSTN side, also the echo will randomly come back in the call sometimes, I'd say 3 out of 10 calls. the other problem I have is that sometimes ( like maybe 4 times a day ) we will be talking to PSTN calls and one side of the call will go silent and then 5 to 10 seconds later come back. we have this setup. PSTN -- Asterisk /w T100p -IAX2- Asterisk2 /w zapdummy -sip snom 200's, and locally in our office it looks like this PSTN -T100p- Asterisk -sip- snom 200's (the echo is on this setup as well) any help would be appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo's and dropped calls
how do you adjust ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Vogel Sent: Thursday, April 01, 2004 12:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Echo's and dropped calls Did you play with txgain and rxgain? Reduces echo but also volume - -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson Sent: Thursday, April 01, 2004 10:21 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Echo's and dropped calls on both the box with the zap interface and the remote office. it helped some but the problem remains -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Vogel Sent: Thursday, April 01, 2004 12:10 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Echo's and dropped calls Do you have echocancel=yes echocancelwhenbridged=yes echotraining=yes In your zapata.conf file? Wiki is good for this - John V. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson Sent: Thursday, April 01, 2004 9:45 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo's and dropped calls Hi all, I have a problem with echo and silence in the middle of calls. the echo problem is that in the first 5 to 10 seconds of a call there is echo on the sip side but not on the PSTN side, also the echo will randomly come back in the call sometimes, I'd say 3 out of 10 calls. the other problem I have is that sometimes ( like maybe 4 times a day ) we will be talking to PSTN calls and one side of the call will go silent and then 5 to 10 seconds later come back. we have this setup. PSTN -- Asterisk /w T100p -IAX2- Asterisk2 /w zapdummy -sip snom 200's, and locally in our office it looks like this PSTN -T100p- Asterisk -sip- snom 200's (the echo is on this setup as well) any help would be appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
This I one of the things we have been looking for!!! I just installed it in about 5mins and works great!!!. Excellent work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nicolas Gudino Sent: Thursday, April 01, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ANNOUNCE: Flash Operator Panel http://sip.house.com.ar/operator Its a server/client combo that displays the status of your Asterisk PBX in a web browser in real time. You can also perform some actions. Hang-up channels and Transfers via drag and drop. The difference with other similar tools is that it displays status in real time (no refreshing necessary), and its graphically appealing. It's a work in progress... so expect some bugs. I appreciate any feedback you can give me. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip problems
you probably need to add a correct host entry in your /etc/hosts file for your machine it goes ip namealias 192.168.1.1 asterisk.goober.org asterisk so 192.168.1.1 asteriskasterisk.googber.org is not the same thing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Shawn Sent: Thursday, April 01, 2004 11:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sip problems chan_sip.c6524 reload_config= unable to get ip address from asterisk, sip disabled The ip address is working fine, Internet works great. Can anyone help...Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival
I am sorry if this is a silly question but I can not seem to locate the festival binaries. does this come with asterisk or is it another project? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival
it's been a long day. I appreciate your help -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Heison Chak Sent: Friday, March 19, 2004 4:16 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Festival It's not a silly question, but you could have done a google search before asking... google 'festival asterisk install' http://www.voip-info.org/tiki-index.php?page=Asterisk+festival+installation -Heison On Fri, Mar 19, 2004 at 04:10:46PM -0600, Justin Carlson wrote: I am sorry if this is a silly question but I can not seem to locate the festival binaries. does this come with asterisk or is it another project? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conference call?
do you know if there is a way to get the conference button on the snom 200's to work? -Original Message- From: Greg Retkowski [mailto:[EMAIL PROTECTED] Sent: Monday, March 15, 2004 1:31 PM To: [EMAIL PROTECTED]; Justin Carlson Subject: Re: [Asterisk-Users] Conference call? On Mon, 15 Mar 2004, Justin Carlson wrote: how do I setup cal conferencing? and get three-way calling going? I am just looking for some direction as I am having difficulty deciding where to start. Look at the meetme application, you have to configure meetme.conf, and have some meetme() commands in your extensions.conf. You will also need a timing device, such as a digium card. If you don't have any hardware device you'll want to install a software timing device, such as kernel module 'ztdummy'. Hope this helps!! -- Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conference call?
thanks, I will look into the phones further :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Greg Retkowski Sent: Monday, March 15, 2004 2:15 PM To: [EMAIL PROTECTED] Cc: 'Greg Retkowski' Subject: RE: [Asterisk-Users] Conference call? Not familiar with the snom's, but most IP phones implement 'conference' internally.. It's a function of the phone independent of the PBX. However if you set up conferencing on asterisk you can transfer someone into the conference then dial it up yourself.. transfer is '#' followed by your conference extension. -- Greg Greg Retkowski / I.T. Infrastructure Consultant /)/|//` [EMAIL PROTECTED] http://www.rage.net/~greg/ C:408-455-3913 /|/ /_/ On Mon, 15 Mar 2004, Justin Carlson wrote: do you know if there is a way to get the conference button on the snom 200's to work? -Original Message- From: Greg Retkowski [mailto:[EMAIL PROTECTED] Sent: Monday, March 15, 2004 1:31 PM To: [EMAIL PROTECTED]; Justin Carlson Subject: Re: [Asterisk-Users] Conference call? On Mon, 15 Mar 2004, Justin Carlson wrote: how do I setup cal conferencing? and get three-way calling going? I am just looking for some direction as I am having difficulty deciding where to start. Look at the meetme application, you have to configure meetme.conf, and have some meetme() commands in your extensions.conf. You will also need a timing device, such as a digium card. If you don't have any hardware device you'll want to install a software timing device, such as kernel module 'ztdummy'. Hope this helps!! -- Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Night menu not working
adding the day / month augments fixed the issue. I like the suggestion
about breaking up the current config.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Thursday, March 11, 2004 3:42 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Night menu not working
These suggestions may not help get your daytime stuff working, but it
should make life easier later.
On Thu, 2004-03-11 at 14:53, Justin Carlson wrote:
[general]
static=yes
writeprotect=no
[globals]
MARYKAY = 21
RECEPTIONIST = 20
KATHY = 22
[daytime]
include = parkedcalls
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
exten = i,1,Playback(invalid)
switch = IAX2/[EMAIL PROTECTED]/dialout
;sip extentions
Make the following section it's own context and have it included in the
above section.
exten = ${MARYKAY},1,Dial,SIP/21|20
exten = ${MARYKAY},2,Voicemail,u21
exten = ${RECEPTIONIST},1,Dial,SIP/20|20
exten = ${RECEPTIONIST},2,Dial,SIP/20SIP/21SIP/22|20
exten = ${RECEPTIONIST},3,Voicemail,u20
exten = ${KATHY},1,Dial,SIP/22|20
exten = ${KATHY},2,Voicemail,u22
The below section probably needs to be defined in a section that can be
included in both daytime and nighttime. You may want to call afterhours
to access your voicemail.
; for Local Voicemail access
exten = *98,1,VoicemailMain
exten = asterisk,1,VoicemailMain
exten = 25,1,Dial,SIP/fax
Maybe this should be included in the above mentioned newly needed
section for your extensions.
; voicemail extentions
exten = 621,1,Voicemail,u21
exten = 620,1,Voicemail,u20
exten = 622,1,Voicemail,u22
exten = 679,1,VoicemailMain
; direct extentions
exten = 201,1,Dial,IAX/[EMAIL PROTECTED]/6515526201
exten = 307,1,Dial,IAX/[EMAIL PROTECTED]/6515522307
exten = 309,1,Dial,IAX/[EMAIL PROTECTED]/6515522309
exten = 313,1,Dial,IAX/[EMAIL PROTECTED]/6515522313
exten = 317,1,Dial,IAX/[EMAIL PROTECTED]/6515522317
exten = 601,1,Dial,IAX/[EMAIL PROTECTED]/6515523601
exten = 603,1,Dial,IAX/[EMAIL PROTECTED]/6515523603
exten = 609,1,Dial,IAX/[EMAIL PROTECTED]/6515523609
exten = 664,1,Dial,IAX/[EMAIL PROTECTED]/6515523664
exten = 694,1,Dial,IAX/[EMAIL PROTECTED]/6515523694
exten = 816,1,Dial,IAX/[EMAIL PROTECTED]/6515526816
exten = 817,1,Dial,IAX/[EMAIL PROTECTED]/6515526817
exten = 821,1,Dial,IAX/[EMAIL PROTECTED]/6515526821
exten = 842,1,Dial,IAX/[EMAIL PROTECTED]/6515526842
[faxmachine]
switch = IAX2/[EMAIL PROTECTED]/faxmachine
[nighttime]
exten = 21,1,Playback(tt-monkeys)
[default]
include = daytime|8:00-14:48|mon-fri
include = nighttime
Seems you are missing the days of month and months arguments there.
Also, you would probably want to conditionally include nighttime also.
--
Steven Critchfield [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Night menu not working
Hi all,
I am trying to get day and nighttime menus to work in * and no matter what
time I specify the first include entry that matches the number dialed is
used. I have included my extentions.conf and my sip phones have a default
context of default.
[general]
static=yes
writeprotect=no
[globals]
MARYKAY = 21
RECEPTIONIST = 20
KATHY = 22
[daytime]
include = parkedcalls
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
exten = i,1,Playback(invalid)
switch = IAX2/[EMAIL PROTECTED]/dialout
;sip extentions
exten = ${MARYKAY},1,Dial,SIP/21|20
exten = ${MARYKAY},2,Voicemail,u21
exten = ${RECEPTIONIST},1,Dial,SIP/20|20
exten = ${RECEPTIONIST},2,Dial,SIP/20SIP/21SIP/22|20
exten = ${RECEPTIONIST},3,Voicemail,u20
exten = ${KATHY},1,Dial,SIP/22|20
exten = ${KATHY},2,Voicemail,u22
; for Local Voicemail access
exten = *98,1,VoicemailMain
exten = asterisk,1,VoicemailMain
exten = 25,1,Dial,SIP/fax
; voicemail extentions
exten = 621,1,Voicemail,u21
exten = 620,1,Voicemail,u20
exten = 622,1,Voicemail,u22
exten = 679,1,VoicemailMain
; direct extentions
exten = 201,1,Dial,IAX/[EMAIL PROTECTED]/6515526201
exten = 307,1,Dial,IAX/[EMAIL PROTECTED]/6515522307
exten = 309,1,Dial,IAX/[EMAIL PROTECTED]/6515522309
exten = 313,1,Dial,IAX/[EMAIL PROTECTED]/6515522313
exten = 317,1,Dial,IAX/[EMAIL PROTECTED]/6515522317
exten = 601,1,Dial,IAX/[EMAIL PROTECTED]/6515523601
exten = 603,1,Dial,IAX/[EMAIL PROTECTED]/6515523603
exten = 609,1,Dial,IAX/[EMAIL PROTECTED]/6515523609
exten = 664,1,Dial,IAX/[EMAIL PROTECTED]/6515523664
exten = 694,1,Dial,IAX/[EMAIL PROTECTED]/6515523694
exten = 816,1,Dial,IAX/[EMAIL PROTECTED]/6515526816
exten = 817,1,Dial,IAX/[EMAIL PROTECTED]/6515526817
exten = 821,1,Dial,IAX/[EMAIL PROTECTED]/6515526821
exten = 842,1,Dial,IAX/[EMAIL PROTECTED]/6515526842
[faxmachine]
switch = IAX2/[EMAIL PROTECTED]/faxmachine
[nighttime]
exten = 21,1,Playback(tt-monkeys)
[default]
include = daytime|8:00-14:48|mon-fri
include = nighttime
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA 200 Fax
yes be sure you are using ULAW and I found that 9600 was the baud rate to go with. 14400 seemed to be unreliable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Friday, March 05, 2004 2:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipura SPA 200 Fax Is anyone presently using the Sipura SPA 2000 for faxing? I was about to look into it and just figured that I would ask to see if anyone ran into any snags, problems, etc. Thanks. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
