Re: [asterisk-users] OT - Gigaset C470IP - How to access SMS settings
2010/9/15 asterisk asterisk aster...@ck-lee.com Olivier, You should find out the SMS tab in the handset but not in the web service. Did you IP pone work? CK Hi, My phone is working OK but there is no SMS menu showing, though you can see this menu all around the user manual. How did you set yours ? Did you use VOIP setup assistant ? Did you go through a menu where you select a country and a provider ? I think SMS settings are country dependent so it should make sense that going through a country menu is necessary. Cheers On Tue, Sep 14, 2010 at 2:27 PM, Olivier oza_4...@yahoo.fr wrote: Hi, With my Gigaset C470IP (with latest 02223 firmware), I can't find a way to access SMS settings from web configuration app or using a handset. Has someone been more successful without using auto-configuration mode ? (For instance, manual says an SMS entry is showing on handset screen but as I plugged my base station into a private LAN, I skipped the whole auto-configuration process ). Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi playback to execute say.conf settings
Hi danny, U r the one responding for me. Thanks a lot. How do we make it visible to asterisk developers. Thanks, Ashik On Tue, Sep 14, 2010 at 7:30 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ashik Ali *Sent:* Tuesday, September 14, 2010 2:27 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] agi playback to execute say.conf settings Hi danny, Shall we take it as agi bug ? Thanks, Ashik snip For lack of a better description, yes. In truth, it is more of an “inconsistency” than an actual “bug” -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11
Hello, I've had the problem again, but there is no core.pid file in my /etc/asterisk... I have : dumpcore = yes ; Dump core on crash (same as -g at startup) in asterisk.conf This time my CLI was open, and I was suddenly disconnected... just like that. There is nothing in my debug file concerning core : bash-3.2# less /var/log/asterisk/debug.vps.hosting.net | grep core bash-3.2# Please help me find this problem. Jonas. On 09/14/2010 09:36 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, September 14, 2010 2:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11 On 09/14/2010 09:12 PM, Carlos Chavez wrote: On Tue, 2010-09-14 at 20:27 +0200, Jonas Kellens wrote: And again !! Without me doing anything !! PBX Core settings - Version: 1.6.2.11 Build Options: LOADABLE_MODULES Maximum calls: Not set Maximum open file handles: Not set Verbosity: 25 Debug level: 0 Maximum load average:0.00 Minimum free memory: 0 MB Startup time:20:24:51 Last reload time:20:24:51 Jonas. The most common explanation is that your Asterisk is crashing and that safe_asterisk is restarting the process. Check your system for core files. Thank you for your reaction. Can you be more specific ?! What system core files do I need to check and what am I looking for ?! I've recently upgraded from 1.4.30 to 1.6.2.11 and then these problems occurred. I had no problems before. Therefore I would think it has something to do with the version I'm currently using... But of course I'm not sure. Could it be my MySQL database ? In what situation does safe_asterisk restart ?? Jonas. #1. look for core.* in /etc/asterisk (may be elsewhere, but if you did - cd /etc/asterisk - asterisk -vvgc The core.pid (pid being the unix process ID of the process) would be generated there. #2. always a possibility, but it shouldn't have broken going from 1.4 to 1.6 #3. numerous answers, but basically anytime a module fails. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Gigaset C470IP - How to access SMS settings
On the S675IP SMS is here: Messaging - SMS - Settings -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11
The reboot occured a 10:11:11, this my debug log : ... [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Function result is '252227026' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Set' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Macro' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Connect connid localhost username passwd Asterisk [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'connid' to value '1' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Query resultid 1 SELECT klantID, vakantieID, feestdagID, kantoorurenID, routeID, Accountco deIN, backupID from DID where DID=252482233 [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'resultid' to value '2' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Fetch fetchid 2 klantID vakantieID feestdagenID kantoorurenID routeID AccountcodeIN BACKUP ID [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: ast_MYSQL_fetch: numFields=7 [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'fetchid' to value '1' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: NoOp [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Clear 2 [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Disconnect 1 [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MacroExit' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Macro' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Connect connid localhost username passwd Asterisk [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'connid' to value '1' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Query resultid 1 SELECT KNUMMER , vmcontext , accountcode_out , hostedformule , mohclass F ROM AstDB where klantID=50 [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'resultid' to value '2' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Fetch fetchid 2 KNUMMER VMCONTEXT ACCOUT FORMULE MOHCLASS [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: ast_MYSQL_fetch: numFields=5 [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'fetchid' to value '1' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: NoOp [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Clear 2 [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Disconnect 1 [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MacroExit' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Set' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'SetMusicOnHold' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Set' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Macro' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: NoOp [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Goto' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: GoTo [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Set' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: Set [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MacroExit' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Function result is '50' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Function result is '1' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Function result is '1' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Expression result is
Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11
Hello, these are my settings in asterisk.conf : ;maxcalls = 10 ; Maximum amount of calls allowed ;maxload = 0.9 ; Asterisk stops accepting new calls if the load average exceed this limit ;maxfiles = 1000 ; Maximum amount of openfiles ;minmemfree = 1 ; in MBs, Asterisk stops accepting new calls if the amount of free memory falls below this watermark So I don't think that asterisk restarts/reloads because of the above settings ?! Jonas. On 09/15/2010 10:36 AM, Jonas Kellens wrote: The reboot occured a 10:11:11, this my debug log : ... [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Function result is '252227026' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Set' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Macro' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Connect connid localhost username passwd Asterisk [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'connid' to value '1' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Query resultid 1 SELECT klantID, vakantieID, feestdagID, kantoorurenID, routeID, Accountco deIN, backupID from DID where DID=252482233 [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'resultid' to value '2' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Fetch fetchid 2 klantID vakantieID feestdagenID kantoorurenID routeID AccountcodeIN BACKUP ID [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: ast_MYSQL_fetch: numFields=7 [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'fetchid' to value '1' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: NoOp [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Clear 2 [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Disconnect 1 [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MacroExit' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Macro' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Connect connid localhost username passwd Asterisk [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'connid' to value '1' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Query resultid 1 SELECT KNUMMER , vmcontext , accountcode_out , hostedformule , mohclass F ROM AstDB where klantID=50 [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'resultid' to value '2' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Fetch fetchid 2 KNUMMER VMCONTEXT ACCOUT FORMULE MOHCLASS [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: ast_MYSQL_fetch: numFields=5 [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: setting var 'fetchid' to value '1' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: NoOp [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Clear 2 [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MYSQL' [Sep 15 10:11:11] DEBUG[12353] app_addon_sql_mysql.c: MYSQL: data=Disconnect 1 [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: MYSQL [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'MacroExit' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Set' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'SetMusicOnHold' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Set' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Macro' [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'NoOp' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: NoOp [Sep 15 10:11:11] DEBUG[12353] pbx.c: Launching 'Goto' [Sep 15 10:11:11] DEBUG[12353] app_macro.c: Executed application: GoTo [Sep 15 10:11:11] DEBUG[12353] pbx.c:
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx formule) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:2]
Set(SIP/INTERTELin-, CDR(accountcode)=AZURAin) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:3]
Set(SIP/INTERTELin-, BRON=473555006 473555006) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:4]
Goto(SIP/INTERTELin-, vakantie) in new stack
[Sep 15 11:16:32] -- Goto (azura,pbx,5)
[Sep 15 11:16:32] -- Executing [...@azura:5]
Macro(SIP/INTERTELin-, vakantie,58) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:1]
MYSQL(SIP/INTERTELin-, Connect connid localhost username
passwd AsteriskHosted) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:2]
MYSQL(SIP/INTERTELin-, Query resultid 1 SELECT ast1 , ast2 ,
na , naID FROM vakantiedata where ID=58) in new stack
vps2301*CLI
Disconnected from Asterisk server
[Sep 15 11:16:32] Executing last minute cleanups
Dialplan :
[macro-vakantie]
exten = s,1,MYSQL(Connect connid localhost username passwd AsteriskHosted)
exten = s,n,MYSQL(Query resultid ${connid} SELECT ast1 , ast2 , na ,
naID FROM vakantiedata where ID=${ARG1})
exten = s,n,MYSQL(Fetch fetchid ${resultid} AST1 AST2 NA naID )
exten = s,n,NoOp(vakantie-ast1 = ${AST1} vakantie-ast2 = ${AST2} na =
${NA} naID = ${naID})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,NoOp(fetchid = ${fetchid})
exten = s,n,GoToIf($[${fetchid}==0]?exit)
exten = s,n,NoOp()
exten = s,n,GoToIfTime(${AST1}?opvakantie)
exten = s,n,GoToIfTime(${AST2}?opvakantie)
exten = s,n(exit),NoOp()
exten = s,n,Set(vakantieresult=continue)
exten = s,n,MacroExit
exten = s,n(opvakantie),NoOp(op vakantie !)
exten = s,n,GoToIf($[${NA}=hangup]?hangup:route)
Do you guys see why Asterisk has problems with this part of the dialplan ?!
Jonas.
On 09/15/2010 10:58 AM, Jonas Kellens wrote:
Hello,
these are my settings in asterisk.conf :
;maxcalls = 10 ; Maximum amount of calls allowed
;maxload = 0.9 ; Asterisk stops accepting new calls if the load
average exceed this limit
;maxfiles = 1000 ; Maximum amount of openfiles
;minmemfree = 1 ; in MBs, Asterisk stops accepting new calls if the
amount of free memory falls below this watermark
So I don't think that asterisk restarts/reloads because of the above
settings ?!
Jonas.
--
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[asterisk-users] Skip Busy Agents/Channels from Queue
Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory memberdelay=0 timeoutrestart = no joinempty = strict leavewhenempty = yes timeout = 50 member = SIP/1009 member = SIP/1010 sip.conf [1009] username=1009 type=friend secret= mailbox=779000 context=default host=dynamic call-limit=2 [1010] username=1010 type=friend secret= mailbox=779000 context=default host=dynamic call-limit=2 -- Regards, Shariq Khan 0333-3501125 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio when overlapdial is set to yes
Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local --- hipath 300 - isdn pri asterisk -- isdn pri - telco-- dest. Here is my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes overlapdial=yes autofalltrought=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 ; Span 2: WPE1/1 wanpipe2 card 1 HDB3/CCS/CRC4 RED group=1,12 context=from-internal switchtype = euroisdn ;overlapdial = outgoing priindication = inband signalling = pri_net channel = 32-46,48-62 context = default group = 63 Span 4: WPE1/3 wanpipe4 card 3 HDB3/CCS/CRC4 group=4,14 context=outrt-001-PSTN_E1 switchtype=qsig signalling=pri_cpe ;facilityenable=yes ;callprogress=yes pridialplan=unknown prilocaldialplan=unknown ;priindication = outofband ;overlapdial = incoming ;priexclusive = yes ;pritimer = t200,1000 ;pritimer = t313,4000 ;immediate=yes channel = 94-108,110-124 context = default group = 63 Any suggestion will be much appreciated. Regards, Mac -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is busy in the Queue.
I have two channel member in queue and i have set the peer limit 2 for
these members.
I want to skip those member who are currently on the call (answered to
calls) and now their status is busy, if Queue see the busy status caller
will not enter in the Queue and go to the next priority.
[test-queue]
strategy = rrmemory
memberdelay=0
timeoutrestart = no
joinempty = strict
leavewhenempty = yes
timeout = 50
member = SIP/1009
member = SIP/1010
sip.conf
[1009]
username=1009
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
[1010]
username=1010
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
--
Regards,
Shariq Khan
0333-3501125
You could use ${DEVICE_STATE(SIP/1009}. Set a variable to indicate all
extensions are busy and then a couple of ExecIf calls to reset the
variable if either of the extensions state is set to NOT_INUSE. You then
have a variab you can use to decide where to jump to in the dialplan
depending on whether both phones are busy or not.
--
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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Jonas, everyone here supports you in your effort to get a good Asterisk installation going, but could you ... maybe restrain yourself a little bit and reduce the number of hasty postings you are sending to this mailing list? Thank you, Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip Busy Agents/Channels from Queue
You mean, I need to check the DEVICE_STATUS of both (sip) users before
sending the caller into queue, otherwise skip the caller from going into
Queue by using ExecIf.
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 3:16 PM, Gareth Blades
list-aster...@skycomuk.comwrote:
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is busy in the
Queue.
I have two channel member in queue and i have set the peer limit 2 for
these members.
I want to skip those member who are currently on the call (answered to
calls) and now their status is busy, if Queue see the busy status caller
will not enter in the Queue and go to the next priority.
[test-queue]
strategy = rrmemory
memberdelay=0
timeoutrestart = no
joinempty = strict
leavewhenempty = yes
timeout = 50
member = SIP/1009
member = SIP/1010
sip.conf
[1009]
username=1009
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
[1010]
username=1010
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
--
Regards,
Shariq Khan
0333-3501125
You could use ${DEVICE_STATE(SIP/1009}. Set a variable to indicate all
extensions are busy and then a couple of ExecIf calls to reset the
variable if either of the extensions state is set to NOT_INUSE. You then
have a variab you can use to decide where to jump to in the dialplan
depending on whether both phones are busy or not.
--
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Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Yes something like this. Note the Execif syntax I have used is for
asterisk 1.6
exten = s,n,Set(AGENTSBUSY=yes)
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1009} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1010} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[$AGENTSBUSY = no]?QUEUE(xxx))
Shariq Khan wrote:
You mean, I need to check the DEVICE_STATUS of both (sip) users before
sending the caller into queue, otherwise skip the caller from going into
Queue by using ExecIf.
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 3:16 PM, Gareth Blades
list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com wrote:
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is busy in
the Queue.
I have two channel member in queue and i have set the peer limit
2 for
these members.
I want to skip those member who are currently on the call
(answered to
calls) and now their status is busy, if Queue see the busy status
caller
will not enter in the Queue and go to the next priority.
[test-queue]
strategy = rrmemory
memberdelay=0
timeoutrestart = no
joinempty = strict
leavewhenempty = yes
timeout = 50
member = SIP/1009
member = SIP/1010
sip.conf
[1009]
username=1009
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
[1010]
username=1010
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
--
Regards,
Shariq Khan
0333-3501125
You could use ${DEVICE_STATE(SIP/1009}. Set a variable to indicate all
extensions are busy and then a couple of ExecIf calls to reset the
variable if either of the extensions state is set to NOT_INUSE. You then
have a variab you can use to decide where to jump to in the dialplan
depending on whether both phones are busy or not.
--
_
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Re: [asterisk-users] One way audio when overlapdial is set to yes
Hi Group, I was able to resolve the problem by disabling the echo cancellation in a104d and using the same dahdi config. Thanks... - Original Message From: leonimar cape leo_mac...@yahoo.com To: asterisk-users@lists.digium.com Sent: Wednesday, September 15, 2010 6:12:35 PM Subject: [asterisk-users] One way audio when overlapdial is set to yes Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local --- hipath 300 - isdn pri asterisk -- isdn pri - telco-- dest. Here is my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes overlapdial=yes autofalltrought=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 ; Span 2: WPE1/1 wanpipe2 card 1 HDB3/CCS/CRC4 RED group=1,12 context=from-internal switchtype = euroisdn ;overlapdial = outgoing priindication = inband signalling = pri_net channel = 32-46,48-62 context = default group = 63 Span 4: WPE1/3 wanpipe4 card 3 HDB3/CCS/CRC4 group=4,14 context=outrt-001-PSTN_E1 switchtype=qsig signalling=pri_cpe ;facilityenable=yes ;callprogress=yes pridialplan=unknown prilocaldialplan=unknown ;priindication = outofband ;overlapdial = incoming ;priexclusive = yes ;pritimer = t200,1000 ;pritimer = t313,4000 ;immediate=yes channel = 94-108,110-124 context = default group = 63 Any suggestion will be much appreciated. Regards, Mac -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Hello Philipp, I know I post a lot concerning this issue, but this is because this problem occurs on a production system and I feel very hot breathing down my neck. I have tested during several weeks my implementation on a test system which is similar to the production system. The only difference is : test-system on VirtualBox, production system on Xen. Also now I can not reproduce this reboot of asterisk on my test system. While on the production system the part of the dialplan gives the spontaneous reboot every time over and over again. I have now placed this part of the dialplan in comment, and it seems to help. I was wrong yesterday thinking it was a bug in version 1.6.2.11. The problem today occurs when the specific macro needs to be executed... Did not saw this yesterday... I hope I have found the *problem* now. Can you help me with the *solution* please ?! Kind regards, Jonas. On 09/15/2010 12:20 PM, Philipp von Klitzing wrote: Jonas, everyone here supports you in your effort to get a good Asterisk installation going, but could you ... maybe restrain yourself a little bit and reduce the number of hasty postings you are sending to this mailing list? Thank you, Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
I cant help you with fixing the actual cause but have you considered moving the mysql and as much of the associated logic to an AGI running something like a perl or php script. From previous posts that generally seems to me the more reliable way of making mysql queries. Jonas Kellens wrote: Hello Philipp, I know I post a lot concerning this issue, but this is because this problem occurs on a production system and I feel very hot breathing down my neck. I have tested during several weeks my implementation on a test system which is similar to the production system. The only difference is : test-system on VirtualBox, production system on Xen. Also now I can not reproduce this reboot of asterisk on my test system. While on the production system the part of the dialplan gives the spontaneous reboot every time over and over again. I have now placed this part of the dialplan in comment, and it seems to help. I was wrong yesterday thinking it was a bug in version 1.6.2.11. The problem today occurs when the specific macro needs to be executed... Did not saw this yesterday... I hope I have found the *problem* now. Can you help me with the *solution* please ?! Kind regards, Jonas. On 09/15/2010 12:20 PM, Philipp von Klitzing wrote: Jonas, everyone here supports you in your effort to get a good Asterisk installation going, but could you ... maybe restrain yourself a little bit and reduce the number of hasty postings you are sending to this mailing list? Thank you, Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Gareth
Usualy the queue has the ability to know if the agent is INUSE and skip
them.. you can simply use ringinuse=no to the queues.conf under the queue
itself or the general section and that's it .. no need for the whole
dialplan.. as you are using SIP members.
Salam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Wednesday, September 15, 2010 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Yes something like this. Note the Execif syntax I have used is for
asterisk 1.6
exten = s,n,Set(AGENTSBUSY=yes)
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1009} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1010} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[$AGENTSBUSY = no]?QUEUE(xxx))
Shariq Khan wrote:
You mean, I need to check the DEVICE_STATUS of both (sip) users before
sending the caller into queue, otherwise skip the caller from going into
Queue by using ExecIf.
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 3:16 PM, Gareth Blades
list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com wrote:
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is busy in
the Queue.
I have two channel member in queue and i have set the peer limit
2 for
these members.
I want to skip those member who are currently on the call
(answered to
calls) and now their status is busy, if Queue see the busy status
caller
will not enter in the Queue and go to the next priority.
[test-queue]
strategy = rrmemory
memberdelay=0
timeoutrestart = no
joinempty = strict
leavewhenempty = yes
timeout = 50
member = SIP/1009
member = SIP/1010
sip.conf
[1009]
username=1009
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
[1010]
username=1010
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
--
Regards,
Shariq Khan
0333-3501125
You could use ${DEVICE_STATE(SIP/1009}. Set a variable to indicate all
extensions are busy and then a couple of ExecIf calls to reset the
variable if either of the extensions state is set to NOT_INUSE. You
then
have a variab you can use to decide where to jump to in the dialplan
depending on whether both phones are busy or not.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] setting up phones
Hi Ott, Have you made it work with Asterisk and Aastra IP Phone. I am also trying the same thing, in Asterisk it shows registered OK but when I dial from extension to extension, call is failed... Please let me know have you made it work...:( On Mon, Jul 13, 2009 at 11:46 PM, Ott Rose sixfourimp...@hotmail.comwrote: I did set sip debug on from the CLI It doesn't scroll messages like it did on Fri i tried 99# and the screen on the phone changed to an ip of 10.0.0.99 which isn't either one of the ips of the asterisk server. then it hung up i do have a dial tone i just figured something out after reading my post. if i dial 60# it shows the ip 10.0.0.60 of the other phone then switch to the extension and the other phone rings. still can't get the 99 to call the asterisk server to work i put in the ips of the server but it hangs up right away -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 13 Jul 2009 12:57:59 -0500 Subject: Re: [asterisk-users] setting up phones I assume you get a dial tone when you pick up the handset?If you had a good phone-to-asterisk connection, debug would show a connection or rejection when you did 99#. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Monday, July 13, 2009 12:49 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones added that line to the extensions.conf file because i could find a way to add it in the GUI. I put it under the dial plan that i have selected. i just get a busy signal i tried #99 just 99, *99 nothing works. debugging isnt showing anything. -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 13 Jul 2009 12:12:16 -0500 Subject: Re: [asterisk-users] setting up phones Most folks (AFAIK) use TFTP to connect to the Asterisk server. I personally use HTTP, but that took a few days of research to figure out. You’re really only using that protocol for configuration and log transfers. The actual lifting is done on a TCP or UDP connection. Your posts Friday indicated that Asterisk was up and “functional” but that you couldn’t make your phones talk to it. I’m thinking that instead of trying to dial phone-to-phone, that you should first make one phone talk to asterisk using this little snippet. - exten = 99,1,Playback(tt-monkeys) - exten = 99,2,Playback(vm-goodbye) - exten = 99,3,hangup When you get your phone where it can dial 99 and get a message, you will be ready to proceed with P2P talking. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ott Rose *Sent:* Monday, July 13, 2009 12:02 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] setting up phones Ok here is what i did. reinstalled asterisk (i used the make samples option) and asterisk-gui in the gui i did the following created a dial plans using the defaults. no outgoing dial plans just local crated two users logged into the web interface with each phone and pointed them to our asterisk server. Just the Proxy server and Registrar server. Still doesn't work. Should i be able to use the configuration server settings form the phones web gui. it has the options for tftp, ftp, http, https. I don't know how this is supposed to be configured. I still don't know what the problem is and sip set debug off does display any info like it was lastweek. I am just trying to use the gui like you suggestd Date: Fri, 10 Jul 2009 14:22:25 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones On Fri, 10 Jul 2009, Ott Rose wrote: I don't think the GUI is editing the conf files correctly. I am not sure I have configure things right. At this point i think i am going to start from scratch. Yea! -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Windows Live™ Hotmail®: Spread the word when you add celeb photos to your e-mails. Check it out.http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_WL_QA_HM_celebrity_photos1_072009cat=celebrity -- Bing™ brings you health information from trusted sources. Try it
Re: [asterisk-users] Help me Out!!!!
I hope someone has helped poor Rob, I would as I am just over the bridge in Bristol, UK but some evil internet scammer has stolen all my money! ;) Cheers! On 15/09/10 12:14, Rob Fugina wrote: It is with deep sorrow and broken heart that am sending you this mail. Am in deep need and my situation is lamentable. my family and I decide to come visit Wales,United Kingdom for a short vacation. To our greatest dismay we were attacked and ripped apart at the park of the hotel where we were lodging,all cash,credit cards and cell phone were forcefully robbed off us at gun point but we still have our passports with us. We've seek help at embassy and high commission,the Police too, unfortunately they have been unable to help or offer any reasonable support whatsoever. Our flight leaves in couple of hour from now but we are being held to ransom by the hotel management because we cannot settle the hotel bills. It is clear we would not be allowed to leave until pay the bill. Word cannot explain the anguish in my heart now. I am in need of immediate assistance. Rob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me Out!!!!
Rough area. Consider yourself lucky you haven't been ripped apart :P Pete wrote: I hope someone has helped poor Rob, I would as I am just over the bridge in Bristol, UK but some evil internet scammer has stolen all my money! ;) Cheers! On 15/09/10 12:14, Rob Fugina wrote: It is with deep sorrow and broken heart that am sending you this mail. Am in deep need and my situation is lamentable. my family and I decide to come visit Wales,United Kingdom for a short vacation. To our greatest dismay we were attacked and ripped apart at the park of the hotel where we were lodging,all cash,credit cards and cell phone were forcefully robbed off us at gun point but we still have our passports with us. We've seek help at embassy and high commission,the Police too, unfortunately they have been unable to help or offer any reasonable support whatsoever. Our flight leaves in couple of hour from now but we are being held to ransom by the hotel management because we cannot settle the hotel bills. It is clear we would not be allowed to leave until pay the bill. Word cannot explain the anguish in my heart now. I am in need of immediate assistance. Rob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Gigaset C470IP - How to access SMS settings
2010/9/15 Randy R randulo2...@gmail.com On the S675IP SMS is here: Messaging - SMS - Settings No SMS entry is showing on Settings/Messaging page, here. How did you set your S675IP ? Did you use any autoconfiguration or country menu ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me Out!!!!
I really need you to help me out of here. On 2010-09-15, Gareth Blades list-aster...@skycomuk.com wrote: Rough area. Consider yourself lucky you haven't been ripped apart :P Pete wrote: I hope someone has helped poor Rob, I would as I am just over the bridge in Bristol, UK but some evil internet scammer has stolen all my money! ;) Cheers! On 15/09/10 12:14, Rob Fugina wrote: It is with deep sorrow and broken heart that am sending you this mail. Am in deep need and my situation is lamentable. my family and I decide to come visit Wales,United Kingdom for a short vacation. To our greatest dismay we were attacked and ripped apart at the park of the hotel where we were lodging,all cash,credit cards and cell phone were forcefully robbed off us at gun point but we still have our passports with us. We've seek help at embassy and high commission,the Police too, unfortunately they have been unable to help or offer any reasonable support whatsoever. Our flight leaves in couple of hour from now but we are being held to ransom by the hotel management because we cannot settle the hotel bills. It is clear we would not be allowed to leave until pay the bill. Word cannot explain the anguish in my heart now. I am in need of immediate assistance. Rob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me Out!!!!
- Original Message - Rough area. Consider yourself lucky you haven't been ripped apart :P Pete wrote: I hope someone has helped poor Rob, I would as I am just over the bridge in Bristol, UK but some evil internet scammer has stolen all my money! ;) Cheers! On 15/09/10 12:14, Rob Fugina wrote: It is with deep sorrow and broken heart that am sending you this mail. Am in deep need and my situation is lamentable. my family and I decide to come visit Wales,United Kingdom for a short vacation. To our greatest dismay we were attacked and ripped apart at the park of the hotel where we were lodging,all cash,credit cards and cell phone were forcefully robbed off us at gun point but we still have our passports with us. We've seek help at embassy and high commission,the Police too, unfortunately they have been unable to help or offer any reasonable support whatsoever. Our flight leaves in couple of hour from now but we are being held to ransom by the hotel management because we cannot settle the hotel bills. It is clear we would not be allowed to leave until pay the bill. Word cannot explain the anguish in my heart now. I am in need of immediate assistance. Rob Makes me want to jump on a train and head down to London and help ... Unfortunately some unscrupulous person has ran off with my wallet! -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 09/15/2010 12:59 PM, Gareth Blades wrote: I cant help you with fixing the actual cause but have you considered moving the mysql and as much of the associated logic to an AGI running something like a perl or php script. From previous posts that generally seems to me the more reliable way of making mysql queries. I have always used mysql directly in the dialplan. Also, I have several other mysql-statements in other parts of the dialplan that do not make asterisk crash. These 2 things make what I'm experciencing odd : - same setup on TEST-server : no problem - other mysql-statements in same dialplan : no problem I have posted the part of the dialplan that makes asterisk reset on this mailinglist. I have also posted the debug log concerning these mysql-lookups. It is actually very basic. If anyone has feedback, please share. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me Out!!!!
http://blog.tmcnet.com/blog/rich-tehrani/google/new-scam-held-up-at-gunpoint-in-wales.html Can't believe (s)he's tried to convince us (s)he's genuine :) http://www.railroad.net/forums/viewtopic.php?f=127t=74905 Been stuck in that hotel for at least two weeks apparently! Must have missed their flight by now? Mind you, Wales is a beautiful part of the world, extra two weeks holiday? Kewl... Pete On 15/09/10 12:52, --[ UxBoD ]-- wrote: - Original Message - Rough area. Consider yourself lucky you haven't been ripped apart :P Pete wrote: I hope someone has helped poor Rob, I would as I am just over the bridge in Bristol, UK but some evil internet scammer has stolen all my money! ;) Cheers! On 15/09/10 12:14, Rob Fugina wrote: It is with deep sorrow and broken heart that am sending you this mail. Am in deep need and my situation is lamentable. my family and I decide to come visit Wales,United Kingdom for a short vacation. To our greatest dismay we were attacked and ripped apart at the park of the hotel where we were lodging,all cash,credit cards and cell phone were forcefully robbed off us at gun point but we still have our passports with us. We've seek help at embassy and high commission,the Police too, unfortunately they have been unable to help or offer any reasonable support whatsoever. Our flight leaves in couple of hour from now but we are being held to ransom by the hotel management because we cannot settle the hotel bills. It is clear we would not be allowed to leave until pay the bill. Word cannot explain the anguish in my heart now. I am in need of immediate assistance. Rob Makes me want to jump on a train and head down to London and help ... Unfortunately some unscrupulous person has ran off with my wallet! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Dear Tarek,
IN_USE is other then the BUSY status, i want to skip the BUSY agent but not
IN_USE
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 4:07 PM, Tarek Sawah tareksa...@hotmail.com wrote:
Gareth
Usualy the queue has the ability to know if the agent is INUSE and skip
them.. you can simply use ringinuse=no to the queues.conf under the queue
itself or the general section and that's it .. no need for the whole
dialplan.. as you are using SIP members.
Salam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth
Blades
Sent: Wednesday, September 15, 2010 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Yes something like this. Note the Execif syntax I have used is for
asterisk 1.6
exten = s,n,Set(AGENTSBUSY=yes)
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1009} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1010} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[$AGENTSBUSY = no]?QUEUE(xxx))
Shariq Khan wrote:
You mean, I need to check the DEVICE_STATUS of both (sip) users before
sending the caller into queue, otherwise skip the caller from going into
Queue by using ExecIf.
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 3:16 PM, Gareth Blades
list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com wrote:
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is busy in
the Queue.
I have two channel member in queue and i have set the peer limit
2 for
these members.
I want to skip those member who are currently on the call
(answered to
calls) and now their status is busy, if Queue see the busy status
caller
will not enter in the Queue and go to the next priority.
[test-queue]
strategy = rrmemory
memberdelay=0
timeoutrestart = no
joinempty = strict
leavewhenempty = yes
timeout = 50
member = SIP/1009
member = SIP/1010
sip.conf
[1009]
username=1009
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
[1010]
username=1010
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
--
Regards,
Shariq Khan
0333-3501125
You could use ${DEVICE_STATE(SIP/1009}. Set a variable to indicate
all
extensions are busy and then a couple of ExecIf calls to reset the
variable if either of the extensions state is set to NOT_INUSE. You
then
have a variab you can use to decide where to jump to in the dialplan
depending on whether both phones are busy or not.
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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Hi! I know I post a lot concerning this issue, but this is because this problem occurs on a production system and I feel very hot breathing down my neck. Why not reduce the pressure and revert to 1.4.30 for the production system until you have figued out the issue? That will give you more time for (off-hours) testing and a more thorough look at what is going on. Look in /tmp if you can't find the core file elsewhere. http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_informati on.txt Philipp P.S.: reboot usually refers to the OS, whereas Asterisk crashes or restarts. From your messages I understand that your linux box is not rebooting itself regularly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Synway cards
I also want to hear the experience of yours with Synway Cards. -- Regards, Shariq Khan 0333-3501125 On Mon, Sep 13, 2010 at 12:47 AM, Anita Hall anita.h...@simmortel.comwrote: Hi Does anyone have experience with Synway cards like SHD-240D-CT/PCI with asterisk and SynAst driver ? Are they any good ? Do they really run on Asterisk ? Thanks. Anita Hall, Simmortel Voice www.simmortel.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me Out!!!!
On 15/09/10 12:14, Rob Fugina wrote: It is with deep sorrow and broken heart that am sending you this mail. Am in deep need and my situation is lamentable. my family and I decide to come visit Wales,United Kingdom for a short vacation. To our greatest dismay we were attacked and ripped apart at the park of the hotel where we were lodging,all cash,credit cards and cell phone were forcefully robbed off us at gun point but we still have our passports with us. We've seek help at embassy and high commission,the Police too, unfortunately they have been unable to help or offer any reasonable support whatsoever. Our flight leaves in couple of hour from now but we are being held to ransom by the hotel management because we cannot settle the hotel bills. It is clear we would not be allowed to leave until pay the bill. Word cannot explain the anguish in my heart now. I am in need of immediate assistance. Rob Weird... My friend's son Steve sent me the exact same email last week word for word! Maybe they are in it together? :-) Which is odd... because Steve is only 12! Anyone else got a theory? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me Out!!!!
He's fortunate that the hotel insists he stay there until his situation improves. --Don Rough area. Consider yourself lucky you haven't been ripped apart :P Pete wrote: I hope someone has helped poor Rob, I would as I am just over the bridge in Bristol, UK but some evil internet scammer has stolen all my money! ;) Cheers! On 15/09/10 12:14, Rob Fugina wrote: It is with deep sorrow and broken heart that am sending you this mail. Am in deep need and my situation is lamentable. my family and I decide to come visit Wales,United Kingdom for a short vacation. To our greatest dismay we were attacked and ripped apart at the park of the hotel where we were lodging,all cash,credit cards and cell phone were forcefully robbed off us at gun point but we still have our passports with us. We've seek help at embassy and high commission,the Police too, unfortunately they have been unable to help or offer any reasonable support whatsoever. Our flight leaves in couple of hour from now but we are being held to ransom by the hotel management because we cannot settle the hotel bills. It is clear we would not be allowed to leave until pay the bill. Word cannot explain the anguish in my heart now. I am in need of immediate assistance. Rob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech To Text on linux with asterisk
2010/9/15, DHAVAL INDRODIYA dhaval.it01...@gmail.com: Hello i have tried to convert through sphinx as suggested by Nickolay i am not getting convert my simple audio file. i am having following error while i fire following command pocketsphinx_continuous -infile /usr/etc/ask-propertyid.WAV -samprate 8000 \ -hmm /usr/etcSpeechToText/Communicator_semi_40.cd_semi_6000 -lm lm_giga_20k_nvp_3gram.lm.DMP *FATAL_ERROR: continuous.c, line 149: Failed to calibrate voice activity detection* Hi That's a progress already. I suspect this file has wrong format. It must be little-endian 16-bit PCM with sample rate 8kHz. uLaw will not work. It's also nice to have a little period of silence in the beginning of the file. Can you provide the file itself/share it somehow so I could take a look. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Just see what the function returns when the agents are busy. You said in
your first post you want to skip the queue if both agents are already on
a call. The dialplan I gave was just an example. You will need to modify
it to do exactly what you want.
I have asterisk emulating a traditional hunt group but I use the
DIAL_STATUS to avoid calling people if they are already on a call. That
way I can still keep call waiting enabled on the phones without it
frequently bothering end users unless its an urgent internal call.
Shariq Khan wrote:
Dear Tarek,
IN_USE is other then the BUSY status, i want to skip the BUSY agent but
not IN_USE
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 4:07 PM, Tarek Sawah tareksa...@hotmail.com
mailto:tareksa...@hotmail.com wrote:
Gareth
Usualy the queue has the ability to know if the agent is INUSE and
skip
them.. you can simply use ringinuse=no to the queues.conf under the
queue
itself or the general section and that's it .. no need for the whole
dialplan.. as you are using SIP members.
Salam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Gareth Blades
Sent: Wednesday, September 15, 2010 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Yes something like this. Note the Execif syntax I have used is for
asterisk 1.6
exten = s,n,Set(AGENTSBUSY=yes)
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1009} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1010} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[$AGENTSBUSY = no]?QUEUE(xxx))
Shariq Khan wrote:
You mean, I need to check the DEVICE_STATUS of both (sip) users
before
sending the caller into queue, otherwise skip the caller from
going into
Queue by using ExecIf.
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 3:16 PM, Gareth Blades
list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com
mailto:list-aster...@skycomuk.com
mailto:list-aster...@skycomuk.com wrote:
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is
busy in
the Queue.
I have two channel member in queue and i have set the peer
limit
2 for
these members.
I want to skip those member who are currently on the call
(answered to
calls) and now their status is busy, if Queue see the busy
status
caller
will not enter in the Queue and go to the next priority.
[test-queue]
strategy = rrmemory
memberdelay=0
timeoutrestart = no
joinempty = strict
leavewhenempty = yes
timeout = 50
member = SIP/1009
member = SIP/1010
sip.conf
[1009]
username=1009
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
[1010]
username=1010
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
--
Regards,
Shariq Khan
0333-3501125
You could use ${DEVICE_STATE(SIP/1009}. Set a variable to
indicate all
extensions are busy and then a couple of ExecIf calls to
reset the
variable if either of the extensions state is set to
NOT_INUSE. You
then
have a variab you can use to decide where to jump to in the
dialplan
depending on whether both phones are busy or not.
--
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[asterisk-users] incoming call FXO
Hi all, Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)...[Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in context 'default', but no invalid handler I have not this 's' extension. Anybody knows what happen? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 09/15/2010 02:03 PM, Philipp von Klitzing wrote: Hi! I know I post a lot concerning this issue, but this is because this problem occurs on a production system and I feel very hot breathing down my neck. Why not reduce the pressure and revert to 1.4.30 for the production system until you have figued out the issue? That will give you more time for (off-hours) testing and a more thorough look at what is going on. Look in /tmp if you can't find the core file elsewhere. http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_informati on.txt Philipp P.S.: reboot usually refers to the OS, whereas Asterisk crashes or restarts. From your messages I understand that your linux box is not rebooting itself regularly. Hello Philipp, I have indeed found the core file in /tmp (that is where 'locate' does not look huh...) It indicates to be a binary file, however I have not found instructions on dealing with this @ the link you gave me. Can you give me instruction on how to handle the core.pid file ? THANKS ! Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me Out!!!!
Dan Journo wrote: Anyone else got a theory? Same message here in the States. The person here had his Gmail account cracked. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 15 Sep 2010, at 13:22, Jonas Kellens wrote: I have indeed found the core file in /tmp (that is where 'locate' does not look huh...) 'updatedb'? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
On 09/15/2010 07:20 AM, Flavio Miranda wrote: Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)... [Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)... [Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in context 'default', but no invalid handler I have not this 's' extension. Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 800 Origination/Termination - International
On Tue, 14 Sep 2010, Joe Freeman wrote: Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can. Thanks- Joe I use IPComms to do US/Canada/US Virgin Islands. 2.5c/min + a flat rate per channel (I think $15?). j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Hi Jonas! It indicates to be a binary file, however I have not found instructions on dealing with this @ the link you gave me. Can you give me instruction on how to handle the core.pid file ? Could I ask you again to make an effort to reduce your number of daily postings to this list? If you google for Asterisk core - or rather search this list's archive - I am quite sure you will very quickly find additional info. In case that fails (for reasons that I cannot imagine) here is another keyword to look for: gdb So please do put a little bit more effort into attacking the issues yourself before creating a post for every single tiny question that you might come across in the process. That's what everyone else here does - we all have to do our own homework first. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 09/15/2010 02:45 PM, Steve Howes wrote: On 15 Sep 2010, at 13:22, Jonas Kellens wrote: I have indeed found the core file in /tmp (that is where 'locate' does not look huh...) 'updatedb'? S Off course I did that, Steve, before I did a locate on 'core'. But doesn't locate also have some PATH ? Where in my case /tmp is not in it. Meanwhile I have come across this : 1. start Asterisk with safe_asterisk 2. enter gdb asterisk core. 3. enter bt while in gdb (or do a bt full) 4. enter thread apply all bt I have no experience with this, so I post my output : [r...@asterisk ~]# gdb asterisk core.4483 snip This GDB was configured as i386-redhat-linux-gnu... /root/core.4483 is not a core dump: File format not recognized (gdb) bt No stack. (gdb) Quit Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Gigaset C470IP - How to access SMS settings
On Wed, Sep 15, 2010 at 1:43 PM, Olivier oza_4...@yahoo.fr wrote: On the S675IP SMS is here: Messaging - SMS - Settings No SMS entry is showing on Settings/Messaging page, here. How did you set your S675IP ? Did you use any autoconfiguration or country menu ? We don't use SMS on fixed. There is nothing on the web menu, only the handset menus. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 09/15/2010 05:45 AM, Steve Howes wrote: On 15 Sep 2010, at 13:22, Jonas Kellens wrote: I have indeed found the core file in /tmp (that is where 'locate' does not look huh...) 'updatedb'? S off topic, but updatedb deliberately doesn't usually look in /tmp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Hi, I went over your dialplan and though it looks fine at first glance, but because I have no experience with Asterisk 1.6, so I would like to ask if commas in mysql query are ok without escape character? In my asterisk 1.4 I would type it like: SELECT var1\, var2\, var3 FROM ... Other things which come to mind: 1. Is your MySQL up to date? 2. Software versions on your test system are the same as on the production system? 3. Can you post a MySQL query from your dialplan which works fine. Regards, Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 9:20 AM, Jonas Kellens jonas.kell...@telenet.be wrote: On 09/15/2010 02:45 PM, Steve Howes wrote: On 15 Sep 2010, at 13:22, Jonas Kellens wrote: ... Off course I did that, Steve, before I did a locate on 'core'. But doesn't locate also have some PATH ? Where in my case /tmp is not in it. Meanwhile I have come across this : 1. start Asterisk with safe_asterisk 2. enter gdb asterisk core. 3. enter bt while in gdb (or do a bt full) 4. enter thread apply all bt I have no experience with this, so I post my output : [r...@asterisk ~]# gdb asterisk core.4483 snip This GDB was configured as i386-redhat-linux-gnu... /root/core.4483 is not a core dump: File format not recognized (gdb) bt No stack. (gdb) Quit Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g., [default] exten = s,1,Verbose( - - - Call received - - - ) exten = s,n,Playback(hello-world) extent = s,n,HangUp() Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO should play the message 'hello-world' (assuming this sound file exists in the sound folder of asterisk), and you'll see the call activity on the CLI. For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of Telephony' book. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 8:59 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 09/15/2010 07:20 AM, Flavio Miranda wrote: Recently I have instaled one Digium TDM410 on my... Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 09/15/2010 03:47 PM, Zeeshan Zakaria wrote:
Hi,
I went over your dialplan and though it looks fine at first glance,
but because I have no experience with Asterisk 1.6, so I would like to
ask if commas in mysql query are ok without escape character? In my
asterisk 1.4 I would type it like:
SELECT var1\, var2\, var3 FROM ...
Other things which come to mind:
1. Is your MySQL up to date?
2. Software versions on your test system are the same as on the
production system?
3. Can you post a MySQL query from your dialplan which works fine.
Regards,
Zeeshan A Zakaria
Indeed the mysql queries on asterisk 1.4 need escape characters. But
when upgrading to 1.6, I noticed that these were misinterpreted. So I
dropped them...
Point 1
Mysql :
mysql status
--
mysql Ver 14.12 Distrib 5.0.77, for redhat-linux-gnu (x86_64) using
readline 5.1
Connection id:2314
Current database:
Current user:r...@localhost
SSL:Not in use
Current pager:stdout
Using outfile:''
Using delimiter:;
Server version:5.0.77 Source distribution
Protocol version:10
Connection:Localhost via UNIX socket
Server characterset:latin1
Db characterset:latin1
Client characterset:latin1
Conn. characterset:latin1
UNIX socket:/var/lib/mysql/mysql.sock
Uptime:5 hours 51 min 17 sec
Threads: 3 Questions: 238036 Slow queries: 0 Opens: 51 Flush tables:
1 Open tables: 42 Queries per second avg: 11.294
Point 2
Asterisk is the same version (compiled from source)
CentOS is the same version, php is the same version, mysql is the same
version (yum update)
!!! one difference : x64 (x86_64 x86_64 x86_64 GNU/Linux) on production
system, and x86 (i686 i686 i386 GNU/Linux) on test environment !!!
Point 3
Mysql from dialplan that works :
[macro-QueryAstDB]
exten = s,1,MYSQL(Connect connid localhost username passwd AsteriskHosted)
exten = s,n,MYSQL(Query resultid ${connid} SELECT KNUMMER , vmcontext ,
accountcode_out , hostedformule , mohclass FROM AstDB where klantID=${ARG1})
exten = s,n,MYSQL(Fetch fetchid ${resultid} KNUMMER VMCONTEXT ACCOUT
FORMULE MOHCLASS)
exten = s,n,NoOp(knummer = ${KNUMMER} vmcontext = ${VMCONTEXT}
accountcode_out = ${ACCOUT} hostedformule = ${FORMULE} mohclass =
${MOHCLASS})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,MacroExit()
Thanks !
Jonas.
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Re: [asterisk-users] Help me Out!!!!
Clearly, if Word cannot explain the anguish in his heart, Mr. Fugina should be using OpenOffice! Cheers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Gigaset C470IP - How to access SMS settings
Yes, only on the handset. My line does not support SMS so sending out is failed. On Wed, Sep 15, 2010 at 9:28 PM, Randy R randulo2...@gmail.com wrote: On Wed, Sep 15, 2010 at 1:43 PM, Olivier oza_4...@yahoo.fr wrote: On the S675IP SMS is here: Messaging - SMS - Settings No SMS entry is showing on Settings/Messaging page, here. How did you set your S675IP ? Did you use any autoconfiguration or country menu ? We don't use SMS on fixed. There is nothing on the web menu, only the handset menus. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dual WAN with load balancing
I encounter problem in using Dual WAN with load balancing on asterisk 1.6.2.11. My problem is registration of one VOIP provider. I can dial out but not probably answer. It drops. One of the error message is SIP/2.0 404 not found. I am not sure about the problem but note that it may be related to incorrect IP being used. Sometimes, WAN 1 and sometimes WAN 2 Could someone help to point to fix? Thanks. CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with audio
Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check Thanks! -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] changing from zap to DAHDI
I am changing a system from zap to DAHDI. I removed everything zap. when doing the command: sh -x /etc/init.d/dahdi start, I see initlog -q -c 'modprobe wct4xxp' sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct4xxp doing updatedb then, locate zap returns me /var/lib/asterisk/sounds/spy-zap.gsm /usr/share/groff/1.18.1.1/font/devps/zapfdr.pfa /usr/lib/asterisk/modules/app_zapateller.so locate zt return me /lib/modules/2.6.9-42.0.3.EL/kernel/sound/pci/snd-azt3328.ko /lib/modules/2.6.9-42.0.3.ELsmp/kernel/sound/pci/snd-azt3328.ko /lib/modules/2.6.9-42.EL/kernel/sound/pci/snd-azt3328.ko /lib/modules/2.6.9-42.ELsmp/kernel/sound/pci/snd-azt3328.ko /usr/src/kernels/2.6.9-42.0.3.EL-i686/include/config/radio/aztech.h /usr/src/kernels/2.6.9-42.0.3.EL-i686/include/config/snd/azt3328 /usr/src/kernels/2.6.9-42.0.3.EL-i686/include/config/snd/azt3328/module.h /usr/src/kernels/2.6.9-42.0.3.EL-hugemem-i686/include/config/radio/aztech.h /usr/src/kernels/2.6.9-42.0.3.EL-hugemem-i686/include/config/snd/azt3328 /usr/src/kernels/2.6.9-42.0.3.EL-hugemem-i686/include/config/snd/azt3328/module.h /usr/src/kernels/2.6.9-42.EL-i686/include/config/radio/aztech.h /usr/src/kernels/2.6.9-42.EL-i686/include/config/snd/azt3328 /usr/src/kernels/2.6.9-42.EL-i686/include/config/snd/azt3328/module.h /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686/include/config/radio/aztech.h /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686/include/config/snd/azt3328 /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686/include/config/snd/azt3328/module.h /usr/share/terminfo/z/ztx-1-a /usr/share/terminfo/z/ztx /usr/share/terminfo/z/ztx11 /usr/share/terminfo/z/zt-1 So everything is gone. why is it using ztcfg? I removed everything zap or zt then installed dahdi, I have even re-installed dahdi and the same is happening. I installed: asterisk-1.4.35 asterisk-1.4.35.tar.gz dahdi-linux-complete-2.1.0+2.1.0 dahdi-linux-complete-2.1.0+2.1.0.tar.gz libpri-1.4.11.4 libpri-1.4.11.4.tar.gz Any thoughts? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error loading skype_for_asterisk
This suddenly started appearing and I'm not sure why. Any ideas? asterisk*CLI module load chan_skype.so Unable to load module chan_skype.so Command 'module load chan_skype.so' failed. [Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: sfa_send_chat_message [Sep 15 11:08:25] WARNING[12274]: loader.c:797 load_resource: Module 'chan_skype.so' could not be loaded. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Wednesday, September 15, 2010 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] changing from zap to DAHDI I am changing a system from zap to DAHDI. I removed everything zap. when doing the command: sh -x /etc/init.d/dahdi start, I see initlog -q -c 'modprobe wct4xxp' sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct4xxp doing updatedb then, locate zap returns me /var/lib/asterisk/sounds/spy-zap.gsm /usr/share/groff/1.18.1.1/font/devps/zapfdr.pfa /usr/lib/asterisk/modules/app_zapateller.so locate zt return me /lib/modules/2.6.9-42.0.3.EL/kernel/sound/pci/snd-azt3328.ko /lib/modules/2.6.9-42.0.3.ELsmp/kernel/sound/pci/snd-azt3328.ko /lib/modules/2.6.9-42.EL/kernel/sound/pci/snd-azt3328.ko /lib/modules/2.6.9-42.ELsmp/kernel/sound/pci/snd-azt3328.ko /usr/src/kernels/2.6.9-42.0.3.EL-i686/include/config/radio/aztech.h /usr/src/kernels/2.6.9-42.0.3.EL-i686/include/config/snd/azt3328 /usr/src/kernels/2.6.9-42.0.3.EL-i686/include/config/snd/azt3328/module.h /usr/src/kernels/2.6.9-42.0.3.EL-hugemem-i686/include/config/radio/aztech.h /usr/src/kernels/2.6.9-42.0.3.EL-hugemem-i686/include/config/snd/azt3328 /usr/src/kernels/2.6.9-42.0.3.EL-hugemem-i686/include/config/snd/azt3328/mod ule.h /usr/src/kernels/2.6.9-42.EL-i686/include/config/radio/aztech.h /usr/src/kernels/2.6.9-42.EL-i686/include/config/snd/azt3328 /usr/src/kernels/2.6.9-42.EL-i686/include/config/snd/azt3328/module.h /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686/include/config/radio/aztech.h /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686/include/config/snd/azt3328 /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686/include/config/snd/azt3328/module. h /usr/share/terminfo/z/ztx-1-a /usr/share/terminfo/z/ztx /usr/share/terminfo/z/ztx11 /usr/share/terminfo/z/zt-1 So everything is gone. why is it using ztcfg? I removed everything zap or zt then installed dahdi, I have even re-installed dahdi and the same is happening. I installed: asterisk-1.4.35 asterisk-1.4.35.tar.gz dahdi-linux-complete-2.1.0+2.1.0 dahdi-linux-complete-2.1.0+2.1.0.tar.gz libpri-1.4.11.4 libpri-1.4.11.4.tar.gz Any thoughts? Jerry AIR, DAHDI out of the box configures modules that aren't necessarily installed on your computer. Check /etc/dahdi/modules and make sure only the installed modules are active. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
On 09/15/2010 10:06 AM, Jerry Geis wrote: I am changing a system from zap to DAHDI. I removed everything zap. when doing the command: sh -x /etc/init.d/dahdi start, I see initlog -q -c 'modprobe wct4xxp' sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct4xxp You wouldn't have a udev rule set to run ztcfg configured in /etc/modprobe.d by any chance would you? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error loading skype_for_asterisk
On 09/15/2010 10:09 AM, Richard Kenner wrote: This suddenly started appearing and I'm not sure why. Any ideas? asterisk*CLI module load chan_skype.so Unable to load module chan_skype.so Command 'module load chan_skype.so' failed. [Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: sfa_send_chat_message [Sep 15 11:08:25] WARNING[12274]: loader.c:797 load_resource: Module 'chan_skype.so' could not be loaded. You don't have a matching version of res_skypeforasterisk loaded. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.36 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.36. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.36 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release candidate: * Fix issue where DNID does not get cleared on a new call when using immediate=yes with ISDN signaling. (Closes issue #17568. Reported by wuwu. Patched by rmudgett) * Fix issue where SIP promiscuous redirect could fail to dial the redirect (app_queue). * Fixes issue with translator frame not getting freed. This issue prevented G.729 licenses from being freed up. (Closes issue #17630. Reported by manvirr. Patched by dvossel) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Q931 - Sending PROGRESS after sending ALERTING is a protocol error. (Closes issue #17874. Reported, patched by nic_bellamy) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.36 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.12 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.12 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix issue where DNID does not get cleared on a new call when using immediate=yes with ISDN signaling. (Closes issue #17568. Reported by wuwu. Patched by rmudgett) * Several updates to res_config_ldap. (Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer. Tested by suretec) * Prevent loss of Caller ID information set on local channel after masquerade. (Closes issue #17138. Reported by kobaz, patched by jpeeler) * Fix SIP peers memory leak. (Closes issue #17774. Reported, patched by kkm) * Add Danish support to say.conf.sample (Closes issue #17836. Reported, patched by RoadKill) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Only do magic pickup when notifycid is enabled. A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set, but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup. (Closes issue #17633. Reported, patched by urosh. Patched by dvossel. Tested by: dvossel, urosh, okrief, alecdavis) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.12 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
You wouldn't have a udev rule set to run ztcfg configured in /etc/modprobe.d by any chance would you? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org Shaun Yes I did in fact. Thanks for pointing me there. jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with audio
Have you checked that the codec order on the phone matched the order set on the server? On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote: Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check Thanks! -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
Jerry Geis wrote: You wouldn't have a udev rule set to run ztcfg configured in /etc/modprobe.d by any chance would you? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org Shaun Yes I did in fact. Thanks for pointing me there. jerry Shaun, After removing everything in modprobe.conf that was ztcfg related: alias eth0 tg3 alias eth1 tg3 alias scsi_hostadapter ata_piix alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd and rebooting it still happens. same error. is there another place I need to edit? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
On Wed, Sep 15, 2010 at 10:15:03AM -0500, Shaun Ruffell wrote: On 09/15/2010 10:06 AM, Jerry Geis wrote: I am changing a system from zap to DAHDI. I removed everything zap. when doing the command: sh -x /etc/init.d/dahdi start, I see initlog -q -c 'modprobe wct4xxp' sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct4xxp You wouldn't have a udev rule set to run ztcfg configured in /etc/modprobe.d by any chance would you? (Not really udev. Modprobe) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.12 Download
Anybody else notice that the 1.6.2.12 download has a version and changelog for 1.6.2.12-rc1? http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.12.tar.gz Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On Wed, Sep 15, 2010 at 9:14 AM, Jonas Kellens jonas.kell...@telenet.be wrote: I have no experience with this, so I post my output : Read doc/backtrace.txt it will explain how to generate a backtrace from a core dump. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dual WAN with load balancing
I am not sure about the problem but note that it may be related to incorrect IP being used. Sometimes, WAN 1 and sometimes WAN 2 Most likely. Get a provider that uses IP authentication rather than registrations, and enable access from both of your WAN IPs. All set. Luki -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with audio
Yes my friend...CONFIRMED!!! G729 on both sides 2010/9/15 Ishfaq Malik i...@pack-net.co.uk Have you checked that the codec order on the phone matched the order set on the server? On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote: Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check Thanks! -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.12 Download
On Wed, Sep 15, 2010 at 11:54 AM, Ryan Wagoner rswago...@gmail.com wrote: Anybody else notice that the 1.6.2.12 download has a version and changelog for 1.6.2.12-rc1? I can confirm, asterisk-dev notified. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not working with Festival
Hi,
I'm experiencing the same problem, with identical symptoms.
I also noticed that after making a call attempt, I see this stuck TCP
connection pair until I stop and restart the asterisk server process.
# netstat -an | grep 1314
tcp0 0 0.0.0.0:13140.0.0.0:*
LISTEN
tcp 46 0 127.0.0.1:52206 127.0.0.1:1314
CLOSE_WAIT
tcp0 0 127.0.0.1:1314 127.0.0.1:52206
FIN_WAIT2
Mark
On Thu, Aug 12, 2010 at 02:41:50PM +0530, Davinder Kumar Meen wrote:
I tried it but I still cannot hear any sound created from Festival()
function. I can hear only a voice saying one which was working earlier
as well. Here is log of asterisk console:
-- Attempting call on SIP/011xx...@gafachi1a for
s...@connect-to-me:1 (Retry 1)
-- Executing [...@connect-to-me:1] Answer(SIP/gafachi1a-,
) in new stack
-- Executing [...@connect-to-me:2] Wait(SIP/gafachi1a-,
7) in new stack
-- Executing [...@connect-to-me:3]
SayDigits(SIP/gafachi1a-, '1') in new stack
-- SIP/gafachi1a- Playing 'digits/1.slin' (language 'en')
-- Executing [...@connect-to-me:4] Festival(SIP/gafachi1a-,
hello john) in new stack
== Parsing '/usr/local/etc/asterisk/festival.conf': == Found
On 11/08/10 11:22 PM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[[1]mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Davinder Kumar Meen
Subject: Re: [asterisk-users] Asterisk not working with Festival
Can anyone help please on this?
snip
[connect-to-me]
exten = s,1,Answer
Exten = s,n,SayDigits(`1')
exten = s,n,Festival(hello john)
exten = s,n,Hangup
snip
When you call in from your mobile, you are using a DAHDI channel
which introduces a 3-7 second delay into the process, unless you
have one of the blessed phone companies that offers call
supervision. If you put a wait(7) in front of SayDigits, you should
hear the call normally.
This is what I would suggest
[connect-to-me]
exten = s,1,Answer
Exten = s,n,Gotoif($[${EXTEN}:0:3) = SIP]?4:3
Exten = s,n,wait(7)
Exten = s,n,SayDigits(`1')
exten = s,n,Festival(hello john)
exten = s,n,Hangup
Thanks,
Davinder Kumar Meen
Partner Project Manager
Impinge Solutions, F-250, Phase 8B, Mohali (India)
www.impingesolutions.com
We also provide server hosting services. Please checkout our website
www.goforspace.com
References
1. mailto:asterisk-users-boun...@lists.digium.com]
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--
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Tel: +1-215-512-0112 US: 877-512-0112
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Re: [asterisk-users] SIP 800 Origination/Termination - International
We use Excel Telecom (recently purchased by Matrix) for International and toll-free origination and termination. Alex On 09/15/2010 06:04 AM, Jeff LaCoursiere wrote: On Tue, 14 Sep 2010, Joe Freeman wrote: Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can. Thanks- Joe I use IPComms to do US/Canada/US Virgin Islands. 2.5c/min + a flat rate per channel (I think $15?). j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with audio
On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias ing.diasda...@gmail.com wrote: Yes my friend...CONFIRMED!!! G729 on both sides If the problem happen with SIP to SIP calls and with the same codec, the problem is inside the phone. Check if you can pump up the volume inside his configuration. What phones are you using? -- -- Adrià Vidal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
On 09/15/2010 10:35 AM, Tzafrir Cohen wrote: On Wed, Sep 15, 2010 at 10:15:03AM -0500, Shaun Ruffell wrote: On 09/15/2010 10:06 AM, Jerry Geis wrote: I am changing a system from zap to DAHDI. I removed everything zap. when doing the command: sh -x /etc/init.d/dahdi start, I see initlog -q -c 'modprobe wct4xxp' sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct4xxp You wouldn't have a udev rule set to run ztcfg configured in /etc/modprobe.d by any chance would you? (Not really udev. Modprobe) Ehh..good point. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
On 09/15/2010 10:35 AM, Jerry Geis wrote: Jerry Geis wrote: You wouldn't have a udev rule set to run ztcfg configured in /etc/modprobe.d by any chance would you? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org Shaun Yes I did in fact. Thanks for pointing me there. jerry Shaun, After removing everything in modprobe.conf that was ztcfg related: alias eth0 tg3 alias eth1 tg3 alias scsi_hostadapter ata_piix alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd and rebooting it still happens. same error. is there another place I need to edit? /etc/modprobe.conf? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 10-09-15 05:25 AM, Jonas Kellens wrote:
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx formule) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:2]
Set(SIP/INTERTELin-, CDR(accountcode)=AZURAin) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:3]
Set(SIP/INTERTELin-, BRON=473555006 473555006) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:4]
Goto(SIP/INTERTELin-, vakantie) in new stack
[Sep 15 11:16:32] -- Goto (azura,pbx,5)
[Sep 15 11:16:32] -- Executing [...@azura:5]
Macro(SIP/INTERTELin-, vakantie,58) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:1]
MYSQL(SIP/INTERTELin-, Connect connid localhost username
passwd AsteriskHosted) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:2]
MYSQL(SIP/INTERTELin-, Query resultid 1 SELECT ast1 , ast2 ,
na , naID FROM vakantiedata where ID=58) in new stack
vps2301*CLI
Disconnected from Asterisk server
[Sep 15 11:16:32] Executing last minute cleanups
Dialplan :
[macro-vakantie]
exten = s,1,MYSQL(Connect connid localhost username passwd AsteriskHosted)
exten = s,n,MYSQL(Query resultid ${connid} SELECT ast1 , ast2 , na ,
naID FROM vakantiedata where ID=${ARG1})
exten = s,n,MYSQL(Fetch fetchid ${resultid} AST1 AST2 NA naID )
exten = s,n,NoOp(vakantie-ast1 = ${AST1} vakantie-ast2 = ${AST2} na =
${NA} naID = ${naID})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,NoOp(fetchid = ${fetchid})
exten = s,n,GoToIf($[${fetchid}==0]?exit)
exten = s,n,NoOp()
exten = s,n,GoToIfTime(${AST1}?opvakantie)
exten = s,n,GoToIfTime(${AST2}?opvakantie)
exten = s,n(exit),NoOp()
exten = s,n,Set(vakantieresult=continue)
exten = s,n,MacroExit
exten = s,n(opvakantie),NoOp(op vakantie !)
exten = s,n,GoToIf($[${NA}=hangup]?hangup:route)
Do you guys see why Asterisk has problems with this part of the dialplan ?!
I've seen problems with MYSQL() application crashing on customers boxes before.
It is not that well supported, and would greatly recommend you move to
func_odbc
usage for dialplan-database integration.
Not only will it simplify your dialplan, but likely will resolve your crashing
issues as well. I've done this for at least 3 customers who were using MYSQL()
and all crashing issues stopped and their dialplans ended up becoming
significantly more readable.
Leif.
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Re: [asterisk-users] Problems with audio
Hi, On 09/15/2010 04:04 PM, Danny Dias wrote: Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check I had this problem with an Asterisk setup few months ago. People outside the company/setup would hear people on the Asterisk side very faintly/low volume. Even after pushing the volume up on the phones to max. In my case, upgrading the firmware of the Grandstream phones we were using solved the problem. I don't know if this is your case as well though. Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.12 Download
On 10-09-15 12:13 PM, Paul Belanger wrote: On Wed, Sep 15, 2010 at 11:54 AM, Ryan Wagonerrswago...@gmail.com wrote: Anybody else notice that the 1.6.2.12 download has a version and changelog for 1.6.2.12-rc1? I can confirm, asterisk-dev notified. Odd, not sure how this happened, but I'll be rebuilding a new release here shortly. Sorry for the noise. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
/ After removing everything in modprobe.conf that was ztcfg related: // alias eth0 tg3 // alias eth1 tg3 // alias scsi_hostadapter ata_piix // alias usb-controller ehci-hcd // alias usb-controller1 uhci-hcd // // and rebooting it still happens. same error. is there another place I // need to edit? // / /etc/modprobe.conf? Shaun, This is what is in my modprobe.conf file presently. more /etc/modprobe.conf alias eth0 tg3 alias eth1 tg3 alias scsi_hostadapter ata_piix alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with audio
Hello Adriá... We are using Linksys 942, softphones Xlite...it's a macro pbx, with almost 1000 users, we've checked the gain and volume on the phones :( 2010/9/15 Adrià Vidal adriavi...@gmail.com On Wed, Sep 15, 2010 at 6:08 PM, Danny Dias ing.diasda...@gmail.comwrote: Yes my friend...CONFIRMED!!! G729 on both sides If the problem happen with SIP to SIP calls and with the same codec, the problem is inside the phone. Check if you can pump up the volume inside his configuration. What phones are you using? -- -- Adrià Vidal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with audio
Thanks Sebastian, It's the same firmware version for all our linksys phones...and we have hundreds of pbx's runnning this firmwares versions without any problem 2010/9/15 Sebastian s...@open-t.co.uk Hi, On 09/15/2010 04:04 PM, Danny Dias wrote: Hello, I'm having some problems with a total SIP Asterisk scenario, some extensions when make internal and outgoing calls can't hear very well the other party, not echo, not packet lostthe problem is that the volume seems to be very low...what could be happening? i'm not sure what to check I had this problem with an Asterisk setup few months ago. People outside the company/setup would hear people on the Asterisk side very faintly/low volume. Even after pushing the volume up on the phones to max. In my case, upgrading the firmware of the Grandstream phones we were using solved the problem. I don't know if this is your case as well though. Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Salu2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 09/15/2010 12:42 PM, Leif Madsen wrote:
On 10-09-15 05:25 AM, Jonas Kellens wrote:
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx formule) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:2]
Set(SIP/INTERTELin-, CDR(accountcode)=AZURAin) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:3]
Set(SIP/INTERTELin-, BRON=473555006473555006) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:4]
Goto(SIP/INTERTELin-, vakantie) in new stack
[Sep 15 11:16:32] -- Goto (azura,pbx,5)
[Sep 15 11:16:32] -- Executing [...@azura:5]
Macro(SIP/INTERTELin-, vakantie,58) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:1]
MYSQL(SIP/INTERTELin-, Connect connid localhost username
passwd AsteriskHosted) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:2]
MYSQL(SIP/INTERTELin-, Query resultid 1 SELECT ast1 , ast2 ,
na , naID FROM vakantiedata where ID=58) in new stack
vps2301*CLI
Disconnected from Asterisk server
[Sep 15 11:16:32] Executing last minute cleanups
Dialplan :
[macro-vakantie]
exten = s,1,MYSQL(Connect connid localhost username passwd AsteriskHosted)
exten = s,n,MYSQL(Query resultid ${connid} SELECT ast1 , ast2 , na ,
naID FROM vakantiedata where ID=${ARG1})
exten = s,n,MYSQL(Fetch fetchid ${resultid} AST1 AST2 NA naID )
exten = s,n,NoOp(vakantie-ast1 = ${AST1} vakantie-ast2 = ${AST2} na =
${NA} naID = ${naID})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,NoOp(fetchid = ${fetchid})
exten = s,n,GoToIf($[${fetchid}==0]?exit)
exten = s,n,NoOp()
exten = s,n,GoToIfTime(${AST1}?opvakantie)
exten = s,n,GoToIfTime(${AST2}?opvakantie)
exten = s,n(exit),NoOp()
exten = s,n,Set(vakantieresult=continue)
exten = s,n,MacroExit
exten = s,n(opvakantie),NoOp(op vakantie !)
exten = s,n,GoToIf($[${NA}=hangup]?hangup:route)
Do you guys see why Asterisk has problems with this part of the dialplan ?!
I've seen problems with MYSQL() application crashing on customers boxes
before.
It is not that well supported, and would greatly recommend you move to
func_odbc
usage for dialplan-database integration.
Not only will it simplify your dialplan, but likely will resolve your crashing
issues as well. I've done this for at least 3 customers who were using MYSQL()
and all crashing issues stopped and their dialplans ended up becoming
significantly more readable.
This will also fix any real or perceived mysql takeover issues since
odbc can be attached to any backend without changing the code.
Leif.
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Re: [asterisk-users] Digest Username/auth name mismatch
Hi, On 09/15/2010 04:19 AM, t. k wrote: Hi I'm sorry. I mailed the same question again. because, it cannot be yet solved. any ideas with asterisk? [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have, digest has a...@192.168.0.1[aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479 handle_request_register: Registration from 'sip:a...@192.168.0.1' failed for '192.168.0.2' - Username/auth name mismatch [] type=friend username= secret= context= canreinvite=no host=dynamic disallow=allallow=ulaw The error seems that UAC set different username of digest. But UAC cannot send same username of digest and from for specification. *Digest username set a...@192.168.0.1 So I want to know how to solve with Asterisk. I will try to help. But others might know more. What SIP client are you using - a softphone, a hardphone? It looks like the client is sending the full a...@192.168.0.1 instead of just as the username. Sebastian Register From: sip:a...@192.168.0.1;tag=644056924 To: sip:a...@192.168.0.1 Call-ID: 2457796...@192.168.0.2 CSeq: 125 REGISTER Contact:sip:a...@192.168.0.2:5060 Authorization: Digest username=a...@192.168.0.1, realm=asterisk, nonce=3e635209, uri=sip:192.168.0.1, response=ec89ab3c90316e05d83774630488c61a, algorithm=MD5 Max-Forwards: 70 Expires: 3600 thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
On 09/14/2010 06:33 PM, Dan Journo wrote: Hi, It seems ive broken my settings and now, asterisk isnt detecting my DTMF tones. What kind of diagnostics can I do to work this out? I've set the extension in sip.conf to everything listed on this page but no result. I've also played around with the settings on the phone with no help either. Someone once said on here that Asterisk and the SIP phone have to match, but that doesnt seem to work either. As per the following link, you have to set dtmfmode the same in sip.conf and in the SIP client configuration (hardware or software phone). Also, it looks like 'inband' only works with ulaw and alaw. http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode Sebastian Any ideas? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call FXO
Ok. Problem solved . Thank you very much!!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Wed, 15 Sep 2010 09:56:36 -0400 From: zisha...@gmail.com To: kpflem...@digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] incoming call FXO As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g., [default] exten = s,1,Verbose( - - - Call received - - - ) exten = s,n,Playback(hello-world) extent = s,n,HangUp() Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO should play the message 'hello-world' (assuming this sound file exists in the sound folder of asterisk), and you'll see the call activity on the CLI. For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of Telephony' book. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-15 8:59 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 09/15/2010 07:20 AM, Flavio Miranda wrote: Recently I have instaled one Digium TDM410 on my... Right, that's what the message is telling you. For incoming calls on FXO, they can *only* be sent to the 's' extension in the target context, since there is no target number passed over the FXO connection. You'll have to create an 's' extension to handle incoming calls however you like. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
On 09/15/2010 11:44 AM, Jerry Geis wrote: /etc/modprobe.conf? Shaun, This is what is in my modprobe.conf file presently. more /etc/modprobe.conf alias eth0 tg3 alias eth1 tg3 alias scsi_hostadapter ata_piix alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd Sorry about that, that's what you said but I didn't see that. What does 'grep zt /etc/modprobe.d/*' return then? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Dear Gareth,
DEVICE_STATE function is not available in asterisk, even DEVSTATE does not
work for me in asterisk 1.4.35. Any other method function to check the
channel status
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 5:11 PM, Gareth Blades
list-aster...@skycomuk.comwrote:
Just see what the function returns when the agents are busy. You said in
your first post you want to skip the queue if both agents are already on
a call. The dialplan I gave was just an example. You will need to modify
it to do exactly what you want.
I have asterisk emulating a traditional hunt group but I use the
DIAL_STATUS to avoid calling people if they are already on a call. That
way I can still keep call waiting enabled on the phones without it
frequently bothering end users unless its an urgent internal call.
Shariq Khan wrote:
Dear Tarek,
IN_USE is other then the BUSY status, i want to skip the BUSY agent but
not IN_USE
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 4:07 PM, Tarek Sawah tareksa...@hotmail.com
mailto:tareksa...@hotmail.com wrote:
Gareth
Usualy the queue has the ability to know if the agent is INUSE and
skip
them.. you can simply use ringinuse=no to the queues.conf under the
queue
itself or the general section and that's it .. no need for the whole
dialplan.. as you are using SIP members.
Salam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Gareth Blades
Sent: Wednesday, September 15, 2010 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Yes something like this. Note the Execif syntax I have used is for
asterisk 1.6
exten = s,n,Set(AGENTSBUSY=yes)
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1009} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[${DEVICE_STATE(SIP/1010} =
NOT_INUSE]?Set(AGENTSBUSY=no))
exten = s,n,ExecIf($[$AGENTSBUSY = no]?QUEUE(xxx))
Shariq Khan wrote:
You mean, I need to check the DEVICE_STATUS of both (sip) users
before
sending the caller into queue, otherwise skip the caller from
going into
Queue by using ExecIf.
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 3:16 PM, Gareth Blades
list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com
mailto:list-aster...@skycomuk.com
mailto:list-aster...@skycomuk.com wrote:
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is
busy in
the Queue.
I have two channel member in queue and i have set the peer
limit
2 for
these members.
I want to skip those member who are currently on the call
(answered to
calls) and now their status is busy, if Queue see the busy
status
caller
will not enter in the Queue and go to the next priority.
[test-queue]
strategy = rrmemory
memberdelay=0
timeoutrestart = no
joinempty = strict
leavewhenempty = yes
timeout = 50
member = SIP/1009
member = SIP/1010
sip.conf
[1009]
username=1009
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
[1010]
username=1010
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
--
Regards,
Shariq Khan
0333-3501125
You could use ${DEVICE_STATE(SIP/1009}. Set a variable to
indicate all
extensions are busy and then a couple of ExecIf calls to
reset the
variable if either of the extensions state is set to
NOT_INUSE. You
then
have a variab you can use to decide where to jump to in the
dialplan
depending on whether both phones are busy or not.
--
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http://www.api-digital.com --
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Re: [asterisk-users] Skip Busy Agents/Channels from Queue
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shariq Khan Sent: Wednesday, September 15, 2010 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Skip Busy Agents/Channels from Queue Dear Gareth, DEVICE_STATE function is not available in asterisk, even DEVSTATE does not work for me in asterisk 1.4.35. Any other method function to check the channel status -- Regards, Shariq Khan 0333-3501125 snip In 1.4.30 I use hints and an AGI to tell me which channels are in use. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 FAX not working
By somehow I made it work by having T38 passthru in both Asterisk and SPA3102. Thanks for the comments.. On Tue, Sep 14, 2010 at 7:05 PM, Gopalakrishnan A.N sai...@gmail.comwrote: Hi, I tried to send fax from Linksys to Grandstream by configuring openSER account.. that works fineonly when I send fax from Linksys to Asterisk I am not able to send On Thu, Sep 9, 2010 at 8:42 PM, Gopalakrishnan A.N sai...@gmail.comwrote: I am from India and I hope I have to use G711u...If I am not wrong On Thu, Sep 9, 2010 at 8:36 PM, Gergo Csibra csi...@gmail.com wrote: Thursday, September 9, 2010, 4:32:29 PM, Gopalakrishnan wrote: I am sending FAX from one extension to another extension. I am not able to send. Preferred Codec:G711u You forget to mentoin where do you live? In some countries the G711a codec and in onther countries the G711u codec useable. -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N, -- Thank you with regards, Gopalakrishnan A.N, -- Thank you with regards, Gopalakrishnan A.N, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.13 Now Available (Re-Releast of 1.6.2.12)
The Asterisk Development Team has announced the release of Asterisk 1.6.2.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves an issue where the .version and ChangeLog files were not updated for 1.6.2.12. Asterisk 1.6.2.13 has no additional changes from 1.6.2.12 other than the .version, ChangeLog and summary files. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.13 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
Sorry about that, that's what you said but I didn't see that. What does 'grep zt /etc/modprobe.d/*' return then? grep zt /etc/modprobe.d/* /etc/modprobe.d/modprobe.conf.dist:alias block-major-29-* aztcd jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Hi! DEVICE_STATE function is not available in asterisk, even DEVSTATE does not work for me in asterisk 1.4.35. Any other method function to check the channel status There is a backport available for 1.4: http://www.voip-info.org/wiki/view/Asterisk+func+device_State I assume that with does not work for me you meant that in your unpachted Asterisk 1.4 version you do not have that function available. This backport is very straigt forward to use: Just drop the file into the funcs/ directory and re-compile. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 800 Origination/Termination - International
nexVortex (http://bit.ly/9bEw9e) can do this. They use Global for TF. They can support both US and CA origination. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Freeman Sent: Tuesday, September 14, 2010 10:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP 800 Origination/Termination - International Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can. Thanks- Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue member status not changing
I have an Asterisk 1.6.0.28 system, with a queue called 'marketing'. Everything appears normal, but the status of the members never changes from 'not in use', even if they are being rang or are in a call. Members are added like so: queue add member SIP/1406 to marketing penalty 0 as SIP/1406 state_interface SIP/1406 And they are present as a hint: exten = 1406,hint,Custom:1406 (I've done exten = 1406,hint,SIP/1406 but that doesn't seem to make a difference.) DEVICE_STATE is being set all through the rest of the dialplan, and is currently working fine for busy lamp notifications (desksets are Polycom 550s) and the like. I can even call the queue and see the call ringing, but the set being rung doesn't register it... marketing has 1 calls (max unlimited) in 'leastrecent' strategy (4s holdtime), W:0, C:10, A:1, SL:0.0% within 0s Members: SIP/1406 (dynamic) (Not in use) has taken no calls yet Callers: 1. SIP/1405-0482 (wait: 0:04, prio: 0) And after answering it, it will still show no calls happening - this is it, while a call was happening. marketing has 0 calls (max unlimited) in 'leastrecent' strategy (4s holdtime), W:0, C:9, A:1, SL:0.0% within 0s Members: SIP/1406 (dynamic) (Not in use) has taken 1 calls (last was 7 secs ago) No Callers Call count and time from last call does update after the call terminates. Is there some trick to get calls in a Queue to show status that I missed? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug with Realtime?
Hi, I think ive found a bug but need someone to double check. Whenever I issue a reload in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue member status not changing
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Sherrill Sent: Wednesday, September 15, 2010 2:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue member status not changing I have an Asterisk 1.6.0.28 system, with a queue called 'marketing'. Everything appears normal, but the status of the members never changes from 'not in use', even if they are being rang or are in a call. Members are added like so: queue add member SIP/1406 to marketing penalty 0 as SIP/1406 state_interface SIP/1406 And they are present as a hint: exten = 1406,hint,Custom:1406 (I've done exten = 1406,hint,SIP/1406 but that doesn't seem to make a difference.) DEVICE_STATE is being set all through the rest of the dialplan, and is currently working fine for busy lamp notifications (desksets are Polycom 550s) and the like. I can even call the queue and see the call ringing, but the set being rung doesn't register it... marketing has 1 calls (max unlimited) in 'leastrecent' strategy (4s holdtime), W:0, C:10, A:1, SL:0.0% within 0s Members: SIP/1406 (dynamic) (Not in use) has taken no calls yet Callers: 1. SIP/1405-0482 (wait: 0:04, prio: 0) And after answering it, it will still show no calls happening - this is it, while a call was happening. marketing has 0 calls (max unlimited) in 'leastrecent' strategy (4s holdtime), W:0, C:9, A:1, SL:0.0% within 0s Members: SIP/1406 (dynamic) (Not in use) has taken 1 calls (last was 7 secs ago) No Callers Call count and time from last call does update after the call terminates. Is there some trick to get calls in a Queue to show status that I missed? 2 things to check 1. core show hints will tell you If the hint is proper and present 2. call-limit (or something that replaced it). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 800 Origination/Termination - International
On Wed, Sep 15, 2010 at 6:04 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Tue, 14 Sep 2010, Joe Freeman wrote: Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can. Thanks- Joe I use IPComms to do US/Canada/US Virgin Islands. 2.5c/min + a flat rate per channel (I think $15?). j Is that the same rate for calls from US and canada? I ask as these two, good incoming rate for calls from the states, but 7 cents a minute for calls from canada: http://flowroute.com/services/inbound/ http://vitelity.net/index.php?p=retailserv voip.ms has two options for tollfree's $0.99 a month and is mentioned on their website $1.49 a month and 3.2 cents a minute incoming from USA or canada, its only listed in the account manager -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug with Realtime?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, September 15, 2010 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bug with Realtime? Hi, I think ive found a bug but need someone to double check. Whenever I issue a reload in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. Thanks Dan By reload you mean sip reload or just any reload in general? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing from zap to DAHDI
On 09/15/2010 01:48 PM, Jerry Geis wrote: Sorry about that, that's what you said but I didn't see that. What does 'grep zt /etc/modprobe.d/*' return then? grep zt /etc/modprobe.d/* /etc/modprobe.d/modprobe.conf.dist:alias block-major-29-* aztcd jerry Somewhere on your system you have a modprobe install command that's running when the module is loaded. Most likely it was installed on your system by http://svn.asterisk.org/view/zaptel/branches/1.4/build_tools/genmodconf?view=markup when you installed zaptel. Do you have an /etc/conf.modules file? What does 'grep -r ztconfig /etc/.' return? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug with Realtime?
By reload you mean sip reload or just any reload in general? Reload in general. It might be an issue only with the Polycom sip phones. Not been able to test any others. I'll try a software phone tomorrow. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug with Realtime?
On 09/15/2010 09:41 PM, Dan Journo wrote: Hi, I think ive found a bug but need someone to double check. Whenever I issue a reload in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. Thanks Dan Yes you loose all SIP registrations and they need to re-register to be reachable again. Don't know if this is a bug, but it's like that in 1.4 and 1.6.2. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
Hi,
I'm using the CallTime and a few other variables to name a recording so that I
can then take the wav file name and see when it was recorded, and what the
recording contains.
However, since ${CDR(start)} contains a space in part of the date, the filename
becomes corrupted when I use samba and share the file over a network.
Therefore I need to replace the spaces with another valid character.
Any ideas how I can do this (simply)?
Here is the macro that i'm using to trigger call recording when the user
presses *1.
[macro-mixmon]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} = ]?startrec:donothing)
exten = s,n(startrec),Playback(beep)
exten = s,n,Set(XAD=1)
exten =
s,n,MixMonitor(/var/lib/asterisk/clientsounds/${CDR(start)}~${CALLFROM}~${CDR(channel):4}.wav,b)
exten = s,n(donothing),MacroExit
Thanks
Dan
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Re: [asterisk-users] Bug with Realtime?
On 10-09-15 03:41 PM, Dan Journo wrote: I think ive found a bug but need someone to double check. Whenever I issue a reload in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. That's not a bug. Only when the phone registers or performs some sort of action (such as placing a call, etc...) does Asterisk query the database. If your phones have a short re-registration time this becomes less of a problem. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording Questions
Hi,
On 09/15/2010 09:02 PM, Dan Journo wrote:
Hi,
I'm using the CallTime and a few other variables to name a recording so that
I can then take the wav file name and see when it was recorded, and what the
recording contains.
However, since ${CDR(start)} contains a space in part of the date, the
filename becomes corrupted when I use samba and share the file over a network.
Therefore I need to replace the spaces with another valid character.
Any ideas how I can do this (simply)?
Here is the macro that i'm using to trigger call recording when the user
presses *1.
[macro-mixmon]
exten = s,1,GotoIf($[${XAD} = 0 | ${XAD} = ]?startrec:donothing)
exten = s,n(startrec),Playback(beep)
exten = s,n,Set(XAD=1)
exten =
s,n,MixMonitor(/var/lib/asterisk/clientsounds/${CDR(start)}~${CALLFROM}~${CDR(channel):4}.wav,b)
Are you sure it is the space which is corrupting it? The space is not
incompatible with either Samba or Linux filesystem. However, is the ~
character part of the filename you are creating? If it is, that is
definitely an illegal/reserved character in the Linux file systems.
Sebastian
exten = s,n(donothing),MacroExit
Thanks
Dan
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Re: [asterisk-users] SIP 800 Origination/Termination - International
On Wed, 15 Sep 2010, Kyle Kienapfel wrote: On Wed, Sep 15, 2010 at 6:04 AM, Jeff LaCoursiere j...@sunfone.com wrote: On Tue, 14 Sep 2010, Joe Freeman wrote: Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can. Thanks- Joe I use IPComms to do US/Canada/US Virgin Islands. 2.5c/min + a flat rate per channel (I think $15?). j Is that the same rate for calls from US and canada? I ask as these two, good incoming rate for calls from the states, but 7 cents a minute for calls from canada: http://flowroute.com/services/inbound/ http://vitelity.net/index.php?p=retailserv voip.ms has two options for tollfree's $0.99 a month and is mentioned on their website $1.49 a month and 3.2 cents a minute incoming from USA or canada, its only listed in the account manager Yes, same rate all around, which is why I settled on them. Very competitive - especially for the Caribbean. Thanks, j-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
