RE: [asterisk-users] Blind transfer 3/4 digits
On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re:sip giving problems, please help.
Yes, I also get these problems occasionally Sep 4 17:44:49 WARNING[1365]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:44:49 WARNING[1364]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 60 ^ Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 120 ^ Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get the channel lock for SIP/gw-442744f0! Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD! Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get the channel lock for SIP/gw-442744f0! Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD!___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with asterisk?
[EMAIL PROTECTED] wrote on 31.08.2006 05:41:52 PM: Matthias Fechner wrote: Hello Roger, * Roger Schreiter [EMAIL PROTECTED] [31-08-06 14:19]: did google for asterisk and fax show no results? yes I found spandsp but it will do everything in software. Is it not a good idea to use my modem for the fax stuff? Why would it not be a good idea to do things in software? Hi all, software-solution would be a good idea... eg. spandsp/iaxmodem hylafax but is where a app_rxfax/app_txfax planned/available for spandsp-0.0.3 ? this would myke things much easier as handling with hylafax, at least to receive fax. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No more linux/compiler.h in Fedora Core 6.
Hi all, When the glibc-kernheaders package disappeared from FC6 test2, builds of Asterisk failed with reference to a missing linux/compiler.h, included from channels/chan_phone.c. I applied the following patch diff -urN asterisk-1.2.11.orig/channels/chan_phone.c asterisk-1.2.11/channels/chan_phone.c --- asterisk-1.2.11.orig/channels/chan_phone.c 2006-08-05 05:08:50.0 + +++ asterisk-1.2.11/channels/chan_phone.c 2006-09-04 07:23:40.0 + @@ -37,9 +37,6 @@ #include linux/telephony.h /* Still use some IXJ specific stuff */ #include linux/version.h -#if LINUX_VERSION_CODE = KERNEL_VERSION(2,6,0) -# include linux/compiler.h -#endif #include linux/ixjuser.h #include asterisk.h Everything builds fine, but I haven't a clue how to test it. I'm inclined to think that if it builds, it should be OK. What y'all think? TIA. -- Bill in Denver ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Prompts playback changing tempo w/ SMP kernel
Hi all, (2nd attempt) this is probably a weird question and something I'm not doing right but I got this bizarre thing going on here. When I boot the system with the SMP kernel and compile (*) with the smp kernel source (actually even if I don't compile, but as long as I boot into the SMP kernel), I get this problem where calling into the system, say to check my voicemail, the prompt playback continously changes tempo. The prompts are played in slow-motion, and then it speeds up to its normal speed, then goes back in slow-mo and so on. It happens (I think) at constant periods. Only the tempo changes, not the pitch of the prompt. Does anyone have any idea what could be happening? I have watched topconstantly but haven't noticed anything bizarre in terms of CPU or Mem usage. This is on a 100mbps LAN with nothing much else on it. And it only happens when it's booted into the smp kernel. So it's something to do with smp, thread scheduling, or some buffer BUT I don't know what exactly. All you champs out there, esp. the asterisk-dev people, any light you can shed on this? Thanks much \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel
Check the timer frequency, it might have a different setting on the two kernels. RR wrote: Hi all, (2nd attempt) this is probably a weird question and something I'm not doing right but I got this bizarre thing going on here. When I boot the system with the SMP kernel and compile (*) with the smp kernel source (actually even if I don't compile, but as long as I boot into the SMP kernel), I get this problem where calling into the system, say to check my voicemail, the prompt playback continously changes tempo. The prompts are played in slow-motion, and then it speeds up to its normal speed, then goes back in slow-mo and so on. It happens (I think) at constant periods. Only the tempo changes, not the pitch of the prompt. Does anyone have any idea what could be happening? I have watched topconstantly but haven't noticed anything bizarre in terms of CPU or Mem usage. This is on a 100mbps LAN with nothing much else on it. And it only happens when it's booted into the smp kernel. So it's something to do with smp, thread scheduling, or some buffer BUT I don't know what exactly. All you champs out there, esp. the asterisk-dev people, any light you can shed on this? Thanks much \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Developers Mailing List asterisk-dev@lists.digium.com,
Hi, I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are 1.2.11 and 1.2.3. Please refer the attached txt files for Linux version information and output of zaptel compile. I will be highly appreciated that any one canhelp meon this regard.-- Thanks Regards,Vidura B. Senadeera. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)
Hi Tzafrir,I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax...Thks, On 9/2/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Sep 01, 2006 at 03:03:39PM +0100, Marco Mouta wrote: Hi all, I've just installed vim70, looking for vim syntax highlighting( for Asterisk.conf files) , http://voip-info.org/tiki-index.php?page=vim+syntax+highlighting, and i notice that both: asterisk.vim and filetype.vimalready refer asterisk configurations. But unfortunately i couldn't get yet the highlight syntax working fine for my asterisk.conf files. Any one can help me?What happens if you run manually::set syntax=asterisk--Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755iax:[EMAIL PROTECTED]+972-50-7952406jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel-1.2.8 compile problem
Hi, I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are 1.2.11 and 1.2.3. Please refer the attached txt files for Linux version information and output of zaptel compile. I will be highly appreciated that any one canhelp meon this regard.-- Thanks Regards,Vidura B. Senadeera. -- Thanks Regards,Vidura B. Senadeera. cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits tones.h cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREmakefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h ./makefw pciradio.rbt radfw radfw.h ZAPTELVERSION=1.2.8 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp cc fw2h.c -o fw2h ./fw2h OCT6114-128D.ima vpm450m_fw.h cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREzttest.c -o zttest cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.EL/build make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/home/vidura/zaptel-1.2.8 modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /home/vidura/zaptel-1.2.8/zaptel.o make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686' Linux version 2.6.9-34.EL ([EMAIL PROTECTED]) (gcc version 3.4.5 20051201 (Red Hat 3.4.5-2)) #1 Wed Mar 8 00:07:35 CST 2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Architecture:MainOffice(AstServer)-SmallOffices(ATA.-LegacyPBX)
Hi,I'm planning a solution to establish a connection between Main office company, and let's say 15 small offices:Plan:1- Asterisk Server @ Main Office (connected to main office legacy pbx)2- Linksys SPA 3000 to install on every small office. Idea behind SPA 3000, the main goal is to keep every user with their traditional phone and just connect SPA3000 to the legacy pbx of the small office and then route the calls from lecagy PBX to MainOffice Asterisk Server via voIP. Then SPA3000 would be used as a low cost solution to allow any user from the small office to call Main office Company. This way with only one or two ATA per small office i would be able to connected every one with main office with very lowcost price I would like to hear from you any suggestions or ideas, is this acceptable for a productions system?-- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.2.8 compile problem
On Mon, Sep 04, 2006 at 03:14:50PM +0600, Vidura Senadeera wrote: Hi, I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are 1.2.11 and 1.2.3. Please refer the attached txt files for Linux version information and output of zaptel compile. Please incluude the errors as well: makelog 21 I will be highly appreciated that any one can help me on this regard. So please don't cross-post. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.2.8 compile problem
Vidura Senadeera wrote: Hi, I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are 1.2.11 and 1.2.3. Please refer the attached txt files for Linux version information and output of zaptel compile. I will be highly appreciated that any one can help me on this regard. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- This message has been scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/, and is believed to be clean. cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits tones.h cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREmakefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h ./makefw pciradio.rbt radfw radfw.h ZAPTELVERSION=1.2.8 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp cc fw2h.c -o fw2h ./fw2h OCT6114-128D.ima vpm450m_fw.h cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREzttest.c -o zttest cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.EL/build make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/home/vidura/zaptel-1.2.8 modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /home/vidura/zaptel-1.2.8/zaptel.o make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686' Linux version 2.6.9-34.EL ([EMAIL PROTECTED]) (gcc version 3.4.5 20051201 (Red Hat 3.4.5-2)) #1 Wed Mar 8 00:07:35 CST 2006 most probably: http://bugs.digium.com/view.php?id=6425 -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any Hardphone with VPNClient embedded?
Hi all,Does any of you knows an Hardphone with VPN client embedded? -- Best regards,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] External calls from Asterisk over a Siemens(legacy) RDSI PBX
Hello,The PBX Siemens BusinessPhone 250 uses the prefix 0 to make an external call(nationals, internationals,...) and I need to make external calls from asterisk to this PBX, for if the IP provider falls.With the internal calls I don't have problems, the PBX make all without problems, but when the call coming from Asterisk, has to be external it doesn't call, although I indicate the prefix in the call, for example 0971539230, and it doesn't call to the number.
An bit of my dialplan to make the call across the PBXexten = _09.,1,Dial(Zap/g1/${EXTEN},20,tTr)exten = _09.,2,CongestionAny suggestions, to how I can make that the PBX receives correctly the call, PREFIX+number, to make the external call.
Thanks in advance,RegardsLlorenç
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Re: [asterisk-users] What I always get asked in SME * deployments
This can be done with the 'hint' priority in the dialplan with the right hardware.Check out this link for an example:http://www.voip-info.org/wiki-Asterisk+phone+snom look at section- SNOM SUBSCRIBE/NOTIFY support for monitoring extension statesOn 9/3/06, Dovid Bender [EMAIL PROTECTED] wrote: Some phones have the BLF feature. You can see on the phone who is and who is not on the phone. With the polycom's you need to get a side car. With the snom's you can use the buttons on the phone itself. When ever we do a roll out of Asterisk in a small business environment replacing an old key system or legacy PBX the receptionist always asks us, How do I know if someone is on a call before transferring them?. My typical answer is why do you need to know, just do an attended transfer and if they can take the call they will, if they can't just tell the caller the person is busy. If the receptionist insists on knowing we give them FOP.Has anyone out there devised a better way to let a receptionist know if someone is on a call? ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX
Hello,The PBX Siemens BusinessPhone 250 uses the prefix 0
to make an external call(nationals, internationals,...) and I need to
make external calls from asterisk to this PBX, for if the IP provider
falls.With the internal calls I don't have problems, the PBX
make all without problems, but when the call coming from Asterisk, has
to be external it doesn't call, although I indicate the prefix in the
call, for example 0971539230, and it doesn't call to the number.
An bit of my dialplan to make the call across the PBXexten = _09.,1,Dial(Zap/g1/${EXTEN},20,tTr)exten = _09.,2,CongestionAny suggestions, to how I can make that the PBX receives correctly the call, PREFIX+number, to make the external call.
Thanks in advance,RegardsLlorenç
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[asterisk-users] includes in realtime ??
Hello ppl, Is it possible to include contexts in the RealTime scenario?? If not, wots the work around?? Thanks in advance. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfer 3/4 digits
Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... What's happen to you guys? I am not yelling, just asking. It is sure not a dialplan question! If it would be a dialplan question, than it would be for each dialing, but it isn't. You mentioned SIP message and that makes me wonder! Are we not using here dtmf ?? that is in my opinion not a sip message, isn't it? If it is a sequence of tones, than why is it different if it is in a string (like snom) or another phone, with single tones? If we understand this part, than is the question, where can I turn on the system to take a longer break between tones still as a string? Back to the dialplan: A Voip number can have different length of digits. Each number is seen as a complete picture, and so a three digit and a four digit number is something different. While in the legacy telephony the digits are worked down one by one and if there is no more use of the digits, they are just garbage and will be not used. Unlike in VoIP, where you can have a three digit number and if you dial four digit, than it is a WRONG number I just verified that: I dialed from 601 to 61522, however, 61522 does not exist, but 615 exists. Guess what? I get a busy tone! That should proof my thoughts (and that without yelling, ... hehehehe) bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] usereqphone=yes seems to don't work
Hi all, I'm looking at a function to add user=phone into sip's trame. So I include usereqphone=yes into the [general] of my sip.conf. But it seems to don't work; so is there an other way to add this user=phone through * ? Cheers, -- Alexandre VERNIOL Technicien VoIP Revendeur Directcentrex Hotline : 0892 46 05 12 Ticket : http://ticket.directcentrex.com www.directcentrex.com www.frontier.fr www.directnom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)
On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote: I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax... Hi!! I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and :set filetype=asterisk :syntax on (optionally) works fine for me. -- Víctor Toofic ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)
Just Great!What was missing is:syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others if is written in the tutorials. I'll look for wiki to post this there. On 9/4/06, Victor Toofic [EMAIL PROTECTED] wrote: On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote: I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax... Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax on (optionally)works fine for me.--Víctor Toofic___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.11 and # key
Hello, Does anybody have problems with recognition of the hash (#) key with * 1.2.11? It seams that after pressing # the call is in a progress but no data is sent.Thanks in advance,Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)
BTW Could you tell me how to i make it load this option by default everytime?On 9/4/06, Marco Mouta [EMAIL PROTECTED] wrote:Just Great!What was missing is :syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others if is written in the tutorials. I'll look for wiki to post this there. On 9/4/06, Victor Toofic [EMAIL PROTECTED] wrote: On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote: I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax... Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax on (optionally)works fine for me.--Víctor Toofic___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)
Done,I've created ~/.vimrc file and inside this file:syntax onthks once moreOn 9/4/06, Marco Mouta [EMAIL PROTECTED] wrote:BTW Could you tell me how to i make it load this option by default everytime? On 9/4/06, Marco Mouta [EMAIL PROTECTED] wrote:Just Great!What was missing is :syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others if is written in the tutorials. I'll look for wiki to post this there. On 9/4/06, Victor Toofic [EMAIL PROTECTED] wrote: On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote: I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax... Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax on (optionally)works fine for me.--Víctor Toofic___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta -- Com os melhores cumprimentos,Marco Mouta -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FAX handling
Hi all,I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8 and we are trying to have FAX receiving working in one of the BRI lines. No problem with FAX transmissions but we can not receive. I have configured in zapata.conf faxdetect=both (tx and rx). FAX machine is connected to one FXS port on a PAP2 with G711a and no echo cancelation configured. When the FAX arrives at the FAX machine, they start negotiating but then it stops as if the format is not recognized by the Fax machine as a valid fax. Does anyone have a similar configuration working?Bests,Jose Limeres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any Hardphone with VPNClient embedded?
Marco Mouta wrote: Hi all, Does any of you knows an Hardphone with VPN client embedded? Take a look at Zultys SIP phones. VPN enabled. www.zultys.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel
Hi Zoa, thanks for responding. Ok, now where do I find this? I'm running 2.6.9-34.0.1 kernel. I tried doing a bit of search and it seems like that the ability to change the frequency doesn't appear till 2.6.13. Am I looking at the right thing? Any hints? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any Hardphone with VPNClient embedded?
Please be aware that from a future support standpoint, you may be a bit limited with Zultys. Their future seems very uncertain they have recently just about ceased operations and let the majority of their employees go. Cory J Andrews voice - 800.398.VoIP X3402 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 04, 2006 10:35 AM Subject: Re: [asterisk-users] Any Hardphone with VPNClient embedded? Marco Mouta wrote: Hi all, Does any of you knows an Hardphone with VPN client embedded? Take a look at Zultys SIP phones. VPN enabled. www.zultys.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PAP2-NA + Asterisk
Lists, Hi! good day, i have been task to install/replace our legacy pbx with asterisk, most of the people here in our office was amaze on how asterisk really works, except for i'm having a problem with dtmf detection on PAP2 ATA converterhere is the call flowATA - SIP - TDM400 - PSTN dtmf detection not working as expected, the other asterisk pbx on the other company decode it with.whenever i press extension 103 the other asterisk server decode it with 110 not 103it doubles the 1 then omit the 3 :(sip.conf snippet[100]type=friend host=dynamic disallow=all allow=ulaw allow=gsm dtmfmode=rfc2833 username=100 secret=100 mailbox=100 nat=yes canreinvite=nothanks in advanceJoy All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Handling Disconnection Causes
hi I am sending all my 'not available' prefixes from h323 gnugk to an asterisk box listening h323 in port 1721 (using oh323 module) to handle disconnection causes based in this document: Example macro for handling hangupcause: http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause so in my extension.conf i put this: ... [voip-h323] ;; All calls to Cause Code 34!! exten = _.,1,Macro(dial-result|34) [macro-dial-result] ; Handles Disconnect Cause Codes (see link above for example) ... It works but I dont know why i'am getting cause 42 instead of 34 (No circuit available) in gnugk, I think my termination parter can re-route with cause 42 because its almost the same (switch congested or overloaded) but i would like to understand this anyway and force to get 34 cause code. thank you rafael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Blind transfer 3/4 digits
Ronald, Like someone already told you, you should explain more clearly the way you try to transfer, we need more details on the procedure, using which button on which phone. We need every detail to help you. This as nothing to do with the way the dial plan is loaded, this is totally false. I'm sure most of the people here don't understand how you try to transfer. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 4 septembre 2006 09:22 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... What's happen to you guys? I am not yelling, just asking. It is sure not a dialplan question! If it would be a dialplan question, than it would be for each dialing, but it isn't. You mentioned SIP message and that makes me wonder! Are we not using here dtmf ?? that is in my opinion not a sip message, isn't it? If it is a sequence of tones, than why is it different if it is in a string (like snom) or another phone, with single tones? If we understand this part, than is the question, where can I turn on the system to take a longer break between tones still as a string? Back to the dialplan: A Voip number can have different length of digits. Each number is seen as a complete picture, and so a three digit and a four digit number is something different. While in the legacy telephony the digits are worked down one by one and if there is no more use of the digits, they are just garbage and will be not used. Unlike in VoIP, where you can have a three digit number and if you dial four digit, than it is a WRONG number I just verified that: I dialed from 601 to 61522, however, 61522 does not exist, but 615 exists. Guess what? I get a busy tone! That should proof my thoughts (and that without yelling, ... hehehehe) bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.2.11 and # key
Are you sure this is not because of the dynamic features in features.conf ? By default, # is defined for the transfer feature. David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Michael Strelnikov Envoyé: 4 septembre 2006 09:53 À: asterisk-users@lists.digium.com Objet: [asterisk-users] Asterisk 1.2.11 and # key Hello, Does anybody have problems with recognition of the hash (#) key with * 1.2.11? It seams that after pressing # the call is in a progress but no data is sent. Thanks in advance, Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Submenus
Hello I'm doing an IVR-service, where pilot can check metar (airport weather information), they enter the 4 letter airport code on their phone, and get the metar read back by text-to-speech. [Metar] exten = 1,1,answer exten = 1,2,Background(Met_welcome) exten = 1,3,set(airport=)2,Background(Met_welcome) exten = 1,3,set(airport=)3,set(airport=) exten = 1,n,Background(Met_Instructions) (When they press an airportcode, I set a variable) exten = 3575,1,set(airport=ekrk) exten = 3575,n,goto(Metar,s,1) exten = 3598,1,set(airport=ekyt) and so on exten = 3575,n,goto(Metar,s,1) exten = 3598,1,set(airport=ekyt) and so on exten = 3598,1,set(airport=ekyt) and so on In the S extension, I do all of the processing . My problem is that some of the airports has the same code, for instance EKCH wich is entered by pressing 3524, but EKAH has the same digits, so I need to make a sub-menu, where I can ask the caller to press 1 for EKCH or 2 for EKAH. Right now, I do like this: exten = 3524,1,background(Met_ch_bi_ah)background(Met_ch_bi_ah) exten = 4,1,set(airport=ekch) exten = 4,n,goto(Metar,s,1) exten = 6,1,set(airport=ekbi) exten = 6,n,goto(Metar,s,1)1,set(airport=ekch) exten = 4,n,goto(Metar,s,1) exten = 6,1,set(airport=ekbi) exten = 6,n,goto(Metar,s,1)6,1,set(airport=ekbi) exten = 6,n,goto(Metar,s,1)6,n,goto(Metar,s,1) This is not a very good solution, if a user by mistake press 4 in the main loop, it goes right to EKCH metar, instead of informing the user that it is an invalid code. So what I need is a way to make a submenu, that is only visible when needed, and can set the airport variable. How do I do this? Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropping extra frame of G.729 ?
Hi anyone know where i can solve this problems ? : Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Sep 4 17:54:11 NOTICE[2774]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX handling
I read on this list not so long ago that you should only enable alaw. I've never tested this. Phil. Jose Limeres [EMAIL PROTECTED] omTo Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 04/09/2006 15:35 [asterisk-users] FAX handling Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Hi all, I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8 and we are trying to have FAX receiving working in one of the BRI lines. No problem with FAX transmissions but we can not receive. I have configured in zapata.conf faxdetect=both (tx and rx). FAX machine is connected to one FXS port on a PAP2 with G711a and no echo cancelation configured. When the FAX arrives at the FAX machine, they start negotiating but then it stops as if the format is not recognized by the Fax machine as a valid fax. Does anyone have a similar configuration working? Bests, Jose Limeres___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX handling
Pls Post your Asterisk CLI when Fax is incoming.On 9/4/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:I read on this list not so long ago that you should only enable alaw.I've never tested this.Phil. Jose Limeres [EMAIL PROTECTED] omTo Sent by:Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.comcc Subje ct 04/09/2006 15:35[asterisk-users] FAX handling Please respond toAsterisk UsersMailing List -Non-CommercialDiscussion [EMAIL PROTECTED] ists.digium.comHi all,I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8 andwe are trying to have FAX receiving working inone of the BRI lines.No problem with FAX transmissions but we can not receive. I have configuredin zapata.conf faxdetect=both (tx and rx).FAX machine is connected to one FXS port on a PAP2 with G711a and no echo cancelation configured. When the FAX arrives at theFAX machine, they start negotiating but then it stops as if the format isnot recognized by the Fax machine as a valid fax.Does anyone have a similar configuration working? Bests,Jose Limeres___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping extra frame of G.729 ?
Noc Phibee wrote: anyone know where i can solve this problems ? : 1) By doing a quick google search; 2) By reading previous posts regarding the same issue; 3) By disabling VAD (Voice activity detection) in your device. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux:My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC)I got the ports open in fedora core 3 (5060 and 1 thru 3) for both interfaces.I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP. When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop.This is my sip.conf file:[general]context=ramosoft allowguest=no realm=ramosoft.com bindaddr=0.0.0.0bindport=5060srvlookup=yespedantic=yestos=184tos=lowdelaymaxexpirey=3600defaultexpirey=120disallow=allallow=ulawallow=ilbcallow=gsmmusicclass=defaultlanguage=esrelaxdtmf=yesrtptimeout=60rtpholdtimeout=300useragent=RamoSoftPBXregcontext=ramosoftlocalnet=10.10.10.0/255.255.255.0rtcachefriends=yes[authentication][311]type=friendregexten=311username=311secret=311callerid="Elpidio Ramos" 311host=dynamicnat=yescanreinvite=no Is there anything I am missing here to get two way voice?Thank you in advance all___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
On 9/4/06, Elpidio Ramos [EMAIL PROTECTED] wrote: When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop. ARE YOU SURE IT ISN'T A DTMF PROBLEM!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
On Mon, 2006-09-04 at 09:49 -0700, Elpidio Ramos wrote: This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux: My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC) I got the ports open in fedora core 3 (5060 and 1 thru 3) for both interfaces. I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP. You do not have either the externip or externhost directives in your sip.conf. If you are connecting from the outside you need to tell Asterisk the IP address or hostname of the outside connection. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
Tzafrir Cohen wrote: On Thu, Aug 31, 2006 at 03:52:00PM -0500, Jay Moore wrote: I have a question on how I can better organize my .conf files. I have 3 different groups of people who use my VoIP service. Let's call them 'Office', 'Factory' and 'Public'. In my Asterisk directory, I have created three folders: 'office', 'factory' and 'public', inside each of which has a sip.conf and an extensions.conf file with appropriate account and extension information. Say, for example, I need to limit some users of the 'Public' group so they cannot make calls outside the building. Obviously I would create two separate contexts. One for users who can make calls outside the build, and one for users who cannot. I would then assign the appropriate context to each user. Right now, I have each appropriate context defined in the main extensions.conf. What I'd like to do is reduce the clutter in extensions.conf and move each context into the extensions.conf in the appropriate subfolder. How do I tell the main extensions.conf file to include the other extensions.conf files without putting an #include file in a context of its own? I hope what I've explained makes sense. If not, please ask questions and I'll try to answer. #include is a verbatim text include. if extensions.conf has: [main] exten = aaa,1,Line1 #include otherfile.conf exten = aaa,2,Line2 and othererfile.conf has: exten = aaa,2,OtherLine1 [other] exten = aaa,1,OtherLine2 You'll eventually get: main: aaa: 1. Line1 2. OtherLine1 other: aaa: 1. OtherLine2 2. Line2 Right, I guess I was wondering if it's possible to include a file without it being in a context. The goal I wanted to achieve was to have as few contexts in the main extensions.conf file as possible. Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
Elpidio Ramos wrote: This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux: My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC) I got the ports open in fedora core 3 (5060 and 1 thru 3) for both interfaces. I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP. When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop. This is my sip.conf file: [general] context=ramosoft allowguest=no realm=ramosoft.com bindaddr=0.0.0.0 bindport=5060 srvlookup=yes pedantic=yes tos=184 tos=lowdelay maxexpirey=3600 defaultexpirey=120 disallow=all allow=ulaw allow=ilbc allow=gsm musicclass=default language=es relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 useragent=RamoSoftPBX regcontext=ramosoft localnet=10.10.10.0/255.255.255.0 rtcachefriends=yes [authentication] [311] type=friend regexten=311 username=311 secret=311 callerid=Elpidio Ramos 311 host=dynamic nat=yes canreinvite=no Is there anything I am missing here to get two way voice? Thank you in advance all If you have two working nic's, then when the soft phone is on the inside of the network, it should register with the IP address of the inside nic. When the soft phone is on the outside (eg Internet), then it should be registering with the IP address of the outside nic. Any other combination is going to give you problems and particularly if you are using a firewall. The problems will be associated with basic layer-3 stuff and nating. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] missing pri connect (wwomera to pri)
I am using my asterisk box with an application linked through the WOOMERA channel. Asterisk bridge the woomera chennel to zap (sangoma aft104d) and vicecersa. The strange think is that after some hours of hevy load asterisk miss some time to relay the connect message from woomera to pri. The debug from woomera and pri is pretty easy, just no connect passed to pri.. If I run asterisk without priority I get the problem early, using priority I have some hours of regular work. This is very wired. Is someone experiencing some issue? Does someone that has woomera knowledg help me to fix the issue? Regards Rosario ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
Not sure if that is the case. I don't get to hear callers from the outside but the can dial my extension within my lan.Does it sound like a DTMF problem?I would think they could not dial my extension if DTMF was involved.Justin Tunney [EMAIL PROTECTED] wrote: On 9/4/06, Elpidio Ramos <[EMAIL PROTECTED]>wrote: When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop.ARE YOU SURE IT ISN'T A DTMFPROBLEM!! Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740Direct ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
But the soft phones have dynamic ip addresses. I have read this is why we use host=dynamic and nat=yes.Carlos Chavez [EMAIL PROTECTED] wrote: On Mon, 2006-09-04 at 09:49 -0700, Elpidio Ramos wrote: This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux: My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC) I got the ports open in fedora core 3 (5060 and 1 thru 3) for both interfaces. I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP. You do not have either the externip or externhost directives in yoursip.conf. If you are connecting from the outside you need to tellAsterisk the IP address or hostname of the outside connection. -- Carlos Chavez PratsDirector de TecnologíaTelecomunicaciones Abiertas de México S.A. de C.V.Tel: +52-55-91169161 Ext 2001 Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740Direct ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Right, I guess I was wondering if it's possible to include a file without it being in a context. The goal I wanted to achieve was to have as few contexts in the main extensions.conf file as possible. Did you try it? It would take... perhaps 30 seconds? A minute if you're a slow typist... Yes, you can do this. #include is a literal text include, as the last poster said. -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
So the #include could be made just after the [general] section o extensions.conf? outside of any specific context, i think this was the question.On 9/4/06, Peter Bowyer [EMAIL PROTECTED] wrote: On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Right, I guess I was wondering if it's possible to include a file without it being in a context.The goal I wanted to achieve was to have as few contexts in the main extensions.conf file as possible.Did you try it? It would take... perhaps 30 seconds? A minute ifyou're a slow typist...Yes, you can do this. #include is a literal text include, as the last poster said.--Peter BowyerEmail: [EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] blf aastra 9133i working but can't pickup calls
Hi,I'm trying to get the blf / pickup working properly on the aastra 9133i,I read the wiki voip-info.org for the setup,setup is working fine for the snom, it works also for the aastra ( the light is flashing when a call comes in on another phone ) but I can't pickup the call ... when I press the prog key corresponding the extension I want to pickup, it just dial the extensions like a new call instead of the picking up any idea ?jean-louis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Roundrobin not working on PRI
Zeeshan Zakaria wrote:
Can somebody send me a sample from their extension.conf to do the
above mentioned thing, i.e. handling DIDs on PRI. This is the first
time I am dealing with PRI, previously I always used SIP DIDs and had
no problem at all.
There is nothing fancy or misterious about it. If you are receiving the
last 4 digits of the DID then this will dial a SIP Phone with the same 4
digits.
exten = _,1,Dial(SIP/${EXTEN})
or if you prefer an auto-attendant then:
exten = _,1,Background(welcome_message)
exten = _,2,Background(more messages...)
--
Andres
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[asterisk-users] Astbill DIALSTRING doesn't work
Hi all,
Im trying to setup Astbill in my Asterisk box, but I'm having some problems.
First I obtain the following when I want to make a call:
-- Executing AGI(SIP/71423-081b9010,
agiastar.agi|called_number) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agiastar.agi
-- SIP Seeding peer from astdb: '71423' at
[EMAIL PROTECTED]:5060 for 3600
-- AGI Script agiastar.agi completed, returning 0
-- Executing Dial(SIP/71423-081b9010, ) in new stack
Sep 4 15:23:25 WARNING[4224]: app_dial.c:781 dial_exec_full: Dial
requires an argument (technology/number)
== Spawn extension (mycontext, called_number, 3) exited non-zero on
'SIP/71423-081b9010'
-- Executing DeadAGI(SIP/71423-081b9010, agistardead.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agistardead.agi
-- AGI Script agistardead.agi completed, returning 0
-- SIP Seeding peer from astdb: '71423' at
[EMAIL PROTECTED]:5060 for 3600
In extensions.conf I have this:
exten = _XX,1,AGI(agiastar.agi,${EXTEN})
exten = _XX,2,Dial(${DIALSTRING})
I think I'm not getting nothing from DIALSTRING. How can I check that,
how can I resolve it?.
I tryed changing the second line bye for this:
exten = _XX,3,Dial(SIP/[EMAIL PROTECTED],90,Ttr)
and it starting work. At this point, I can make calls, but the billing
seems doesn't work. So, I think it is because I supressed the
DIALSTRING line.
Can anybody helpme with that? I cant find a manual or technical
information about astbill, except a few lines y Wiki. Is there
something like that?
Thanks very much in advance,
Sebastian
e-mail:[EMAIL PROTECTED]
msn:[EMAIL PROTECTED]
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[asterisk-users] playback some digits to the caller from the callee (involves DTMF) prob
how come can i get or read DTMF digits from a callee(agent) then playback them on the caller's channel i am aware of the read() application, but the problem is that where shud i place it or how shuld i use it, because if i place it below the Dial() application then the moment Dial app terminates, it detroys the channels(i.e the call gets hungup) leaving no callee channel for read() to read DTMF then i tried another way using features.conf i.e. defining a feature to read DTMF into a variable then playing it back on either both of the channels or anyone of them, herez my features.conf portion + dialplan -features.conf-- ... readFeature = #6,peer/callee,Read,var1 playbackFeature = #9,peer/callee,SayDigits,var1 --- extensions.conf ... exten = 500 , 1 , Answer() exten = 500, 2 , SetVar(DYNAMIC_FEATURES=readFeature#playbackFeature) exten = 500 , 3 , Playback(connecting) exten = 500 , 4 , Dial(SIP/100) --- but whenever someone dials 500 gets connected with the agent(100) the agent(callee) presses #6 or #9, simply nothing happens. and on CLI the command show features results this: Builtin Feature..Default Current ===...=...= Pickup...*8.*8 Blind Transfer#..#1 Attended Transfer.*2 One Touch Monitor Disconnect Call..*..*0 Dynamic Feature...Default Current ===...= (none) Call parking Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-750 it seems that the features that i defined are not even getting registered somebody plz suggest me if i am wrong somewhere doing that, or plz try giving me any newer solution to get DTMF digits durring a call or a way to acheive the overall objective i.e. playback some digits to the caller from the callee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream and H.264 !
hi, I´ need some help to implement the Grandstream GXV-3000 in my * platform. Someone know the state of H.264 Video Codec for Asterrisk?? Thanks!!! p.D.: appreciate any help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream and H.264 !
On Mon, 2006-09-04 at 16:26 -0300, Sergio (Red) wrote: hi, I´ need some help to implement the Grandstream GXV-3000 in my * platform. Someone know the state of H.264 Video Codec for Asterrisk?? Thanks!!! This has been answered multiple times in the last month. Search the list before posting. You have to use the SVN version of Asterisk if you wish to use H264. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
Marco: Ah I see. There's a [general] context. I'm pretty new to this Asterisk stuff and I didn't realize there was a general context that you could do things like global includes. Thanks, I'll give it a shot when I'm back in the office on Tuesday. Peter: No need to be an ass about it, pal. Not all of us are as adept at this as you are. Jay Marco Mouta wrote: So the #include could be made just after the [general] section o extensions.conf? outside of any specific context, i think this was the question. On 9/4/06, Peter Bowyer [EMAIL PROTECTED] wrote: On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Right, I guess I was wondering if it's possible to include a file without it being in a context. The goal I wanted to achieve was to have as few contexts in the main extensions.conf file as possible. Did you try it? It would take... perhaps 30 seconds? A minute if you're a slow typist... Yes, you can do this. #include is a literal text include, as the last poster said. -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looks like Nufone is changing around...
Looks like Nufone is making some positive changes... --- NuFone Announces the Creation of a New Executive Management, Support Team NuFone Inc., a premium-service provider specializing in hosted SIP and IAX VoIP solutions, is proud to announce the creation of a new executive management and support team. Eugene, OR (PRWEB) September 2, 2006 -- NuFone, the world's first commercial provider of IAX-based VoIP services, announced today the creation of a new management and support team to further solidify its dedication to providing reliable VoIP solutions to carrier, enterprise and residential environments. Composed of 5 executive team members, who are highly experienced in the areas of business, sales and support, will provide the necessary leadership to properly manage NuFone. In the past, NuFone always had trouble properly managing and supporting our customer's needs. It has always been my goal to form a proper team to deliver the support our customers demand, Jeremy McNamara, founder and CTO of NuFone, said. By listening to our customers, we were able to determine our weaknesses and have formed a proper team to bring NuFone to the next level. The following are the executive members of the NuFone management and support team: Allan Noorda, President and CEO. Noorda has been engulfed in the advancement of the Telecommunications Industry for over the past 10 years. Noorda recently comes from Newman Telecom, where he was the VP of Sales and Marketing. He brings energy and an understanding of customer needs as well as technology to direct the daily operations of NuFone. Jeremy McNamara, Founder and CTO. Over the past 10 years, McNamara has assisted in the development and deployment of several ISPs, ITSPs and Application Service Providers around the United States. McNamara also has extensive development, testing and deployment expertise with Asterisk PBX-based solutions. Greg Merriweather, Support Specialist. Merriweather has been providing operating system, hardware and application support for the past 10 years including working as a support engineer for Ford and Global-Crossing before assisting in the operation of NuFone beginning in early 2003. Leon Salisbury, Senior Engineer. Salisbury has over 20 years of programming and engineering experience with a wide assortment of programming languages including Assembler, Perl, HTML, C and hardware including PICs, 68xx Series, various DSP and embedded x86 platforms. Krystina Patterson, Customer Relations. Patterson has been working with the public for the past 4 years in marketing and customer relations. Patterson is well versed in problem solving and determining customer needs. About NuFone NuFone was originally deployed by Jeremy McNamara in January 2002, as an IAX-based solution for Asterisk PBX based users. NuFone has since grown into a leading provider of SIP and IAX based VoIP solutions for thousands of customers in the United states and more than 60 countries world-wide. NuFone is a privately held corporation based in Eugene, OR. ### ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any Hardphone with VPNClient embedded?
On Mon, September 4, 2006 16:55, Cory Andrews said: Please be aware that from a future support standpoint, you may be a bit limited with Zultys. Their future seems very uncertain they have recently just about ceased operations and let the majority of their employees go. Cory J Andrews voice - 800.398.VoIP X3402 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 04, 2006 10:35 AM Subject: Re: [asterisk-users] Any Hardphone with VPNClient embedded? Marco Mouta wrote: Hi all, Does any of you knows an Hardphone with VPN client embedded? Take a look at Zultys SIP phones. VPN enabled. www.zultys.com As I too am interested in IPsec capable hardphones (or ATA's), do you have a suggestion what to look at instead? I mean: It's nice to say the company may not be around for long, but if there's no alternative, what choice does one have? TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Marco: Ah I see. There's a [general] context. I'm pretty new to this Asterisk stuff and I didn't realize there was a general context that you could do things like global includes. Thanks, I'll give it a shot when I'm back in the office on Tuesday. Peter: No need to be an ass about it, pal. Not all of us are as adept at this as you are. You've still not got it. #include is a general text include - can be used anywhere. Well, perhaps it has to be at the start of a line. Contexts, not even the [general] section which isn't actually a context, has any relevance. It will insert the contents of the included file as though it was in the main file, wherever you put it. You could put the whole of the sip.conf file in an #include'd file. The whole of one context. One and a half contexts. 2 lines out of the [general] section. And so on. All of which, to repeat, could be experienced with a small investment of your time. It really does pay to experiment with the simple things, you find your learning curve is so much flatter than if you ask questions in a vacuum. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_conference not working for me
I'm having trouble getting app_conference to work and I'm feeling pretty clueless right know. With no flags, it doesn't exit when I press '#.' With flags passed as d, it just ignores '#.' With flags passed as MTV, it crashes Asterisk when I press '#.' Any clues would be appreciated :) Here's how I'm invoking conference: exten = *,n,conference(test) or exten = *,n,conference(test|d) or exten = *,n,conference(test|MTV) Here's the console output with no flags: -- Accepting AUTHENTICATED call from a.b.c.d: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw), priority = mine -- Executing Conference(IAX2/a.b.c.d:1030-4, test) in new stack Sep 4 12:53:41 ERROR[10454]: frame.c:386 convert_frame: unable to translate frame Here's what gets syslogged: Sep 4 12:53:35 dt-ext asterisk[10222]: VERBOSE[10227]: -- Accepting AUTHENTICATED call from a.b.c.d: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw), priority = mine Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10225]: chan_iax2.c:9434 in iax2_devicestate: Checking device state for device a.b.c.d Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10225]: devicestate.c:187 in do_state_change: Changing state for IAX2/a.b.c.d:1030 - state 4 (Invalid) Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: pbx.c:1677 in pbx_extension_helper: Launching 'Conference' Sep 4 12:53:35 dt-ext asterisk[10222]: VERBOSE[10453]: -- Executing Conference(IAX2/a.b.c.d:1030-4, test) in new stack Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:415 in member_exec: [ $Revision: 1.9 $ ] begin processing member thread, channel = IAX2/a.b.c.d:1030-4 Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: chan_iax2.c:3370 in iax2_answer: Answering IAX2 call Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:742 in create_member: attempting to parse passed params, stringp = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:793 in create_member: parsed data params, id = test, flags = , priority = 0, vad_prob_start = 0.05, vad_prob_continue = 0.02 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:1077 in create_member: created member, type = S, priority = 0, readformat = 4 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:451 in member_exec: CHANNEL INFO, CHANNEL = IAX2/a.b.c.d:1030-4, DNID = *, CALLER_ID = 21012006, ANI = 21012006 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:454 in member_exec: CHANNEL CODECS, CHANNEL = IAX2/a.b.c.d:1030-4, NATIVE = 4, READ = 4, WRITE = 4 Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: channel.c:2376 in set_format: Set channel IAX2/a.b.c.d:1030-4 to read format ulaw Sep 4 12:53:35 dt-ext asterisk[10222]: DEBUG[10453]: channel.c:2376 in set_format: Set channel IAX2/a.b.c.d:1030-4 to write format ulaw Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:504 in start_conference: attempting to find requested conference Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:548 in find_conf: conflist has not yet been initialized, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:511 in start_conference: attempting to create requested conference Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:583 in create_conf: entered create_conf, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: WARNING[10453]: translate.c:116 in ast_translator_build_path: No translator path from unknown to unknown Sep 4 12:53:35 dt-ext asterisk[10222]: WARNING[10453]: translate.c:116 in ast_translator_build_path: No translator path from unknown to alaw Sep 4 12:53:35 dt-ext asterisk[10222]: WARNING[10453]: translate.c:116 in ast_translator_build_path: No translator path from unknown to unknown Sep 4 12:53:35 dt-ext last message repeated 5 times Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:796 in add_member: member added to conference, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:646 in create_conf: added new conference to conflist, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: conference.c:663 in create_conf: started conference thread for conference, name = test Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:514 in member_exec: begin member event loop, channel = IAX2/a.b.c.d:1030-4 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:532 in member_exec: Conference Members: 1 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:538 in member_exec: Quiet debug 0 - 0 Sep 4 12:53:35 dt-ext asterisk[10222]: NOTICE[10453]: member.c:546 in member_exec: skipping entry message on IAX2/a.b.c.d:1030-4 Sep 4
Re: [asterisk-users] Looks like Nufone is changing around...
Justin Newman wrote: Looks like Nufone is making some positive changes... Thanks Justin, but Asterisk-users is for Asterisk discussion only. Perhaps a more appropriate list would be asterisk-biz. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] includes in realtime ??
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: Monday, September 04, 2006 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] includes in realtime ?? Hello ppl, Is it possible to include contexts in the RealTime scenario?? If not, wots the work around?? Thanks in advance. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Amazing how the wiki has this vast amount of AT LEAST info to start your research on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call center reports
Can someone point me to call center reports available from Asterisk? We setup a small call center with agents, and will now be looking at reports. Ideas? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX handling
On Mon, Sep 04, 2006 at 04:35:43PM +0200, Jose Limeres wrote: Hi all, I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8 and we are trying to have FAX receiving working in one of the BRI lines. No problem with FAX transmissions but we can not receive. I have configured in zapata.conf faxdetect=both (tx and rx). FAX machine is connected to one FXS port on a PAP2 with G711a and no echo cancelation configured. When the FAX arrives at the FAX machine, they start negotiating but then it stops as if the format is not recognized by the Fax machine as a valid fax. Does anyone have a similar configuration working? Hmmm How can asterisk detect that the call is a fax? Is it by answering it? If you get rid of the fax detection and send everything to the fax extension, will faxes get through? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missing pri connect (wwomera to pri)
On Mon, Sep 04, 2006 at 01:27:59PM -0400, Rosario Pingaro wrote: I am using my asterisk box with an application linked through the WOOMERA channel. Asterisk bridge the woomera chennel to zap (sangoma aft104d) and vicecersa. The strange think is that after some hours of hevy load asterisk miss some time to relay the connect message from woomera to pri. The debug from woomera and pri is pretty easy, just no connect passed to pri.. So what does happen to such a bad call? Could you be more specific? (trace, please) Versions of asterisk, zaptel and woomera may also help a bit. If I run asterisk without priority I get the problem early, using priority I have some hours of regular work. This is very wired. Is someone experiencing some issue? Does someone that has woomera knowledg help me to fix the issue? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call center reports
Technical Support wrote: Can someone point me to call center reports available from Asterisk? http://queuemetrics.loway.it/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FAX handling
Look into NVDETECT, and fax2mail script on www.generationd.com Fax detection is automatic MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, September 04, 2006 5:05 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FAX handling On Mon, Sep 04, 2006 at 04:35:43PM +0200, Jose Limeres wrote: Hi all, I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8 and we are trying to have FAX receiving working in one of the BRI lines. No problem with FAX transmissions but we can not receive. I have configured in zapata.conf faxdetect=both (tx and rx). FAX machine is connected to one FXS port on a PAP2 with G711a and no echo cancelation configured. When the FAX arrives at the FAX machine, they start negotiating but then it stops as if the format is not recognized by the Fax machine as a valid fax. Does anyone have a similar configuration working? Hmmm How can asterisk detect that the call is a fax? Is it by answering it? If you get rid of the fax detection and send everything to the fax extension, will faxes get through? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX handling
On Mon, Sep 04, 2006 at 05:16:00PM -0400, Technical Support wrote: Look into NVDETECT, and fax2mail script on www.generationd.com Fax detection is automatic B ut not needed, as calls already come from chan_zap that has its own fax detection. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_conference not working for me
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Edwards wrote: I'm having trouble getting app_conference to work and I'm feeling pretty clueless right know. Probably the iaxclient list would be the better forum to discuss this as its not in the Asterisk codebase. To sign up for the iaxclient mailing list go to: https://lists.sourceforge.net/lists/listinfo/iaxclient-devel - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE/J6cS6d5vy0jeVcRArIzAJ9GxywjnYuC8k/bOOFsqDqaE6VF/wCbBD10 oXJfOkjFL/MUxpbz+4bDVNE= =W+n6 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Nufone making changes
Justin Newman wrote: Looks like Nufone is making some positive changes... Thanks Justin, but Asterisk-users is for Asterisk discussion only. Perhaps a more appropriate list would be asterisk-biz. Jeremy McNamara Noted. Nonetheless, looks like you guys are making progress. :p ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk calling through FWD?
I thought maybe my configs would have been a good idea to post:
iax.conf:
[general]
bindport=4569
bindaddr=10.0.0.20
bandwidth=medium
disallow=lpc10
allow=gsm
jitterbuffer=no
forcejitterbuffer=no
register = 776754:snipped@iax2.fwdnet.net
allow=ulaw
tos=lowdelay
autokill=yes
[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
Extensions.conf
[globals]
FWDNUMBER=776754
FWDCIDNAME=Nick Ellson
FWDPASSWORD=snipped
FWDRINGS=SIP/4003
FWDVMBOX=4003
[default]
include = mainmenu
exten = _393.,1,SetCIDNum(${FWDNUMBER})
exten = _393.,2,SetCallerID,${FWDCIDNAME}
exten =
_393.,3,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
exten = _393.,4,Congestion
[fromiaxfwd]
exten = ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten = ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten = ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
-
And I am not sure if something changed but now I get:
-- Executing SetCIDNum(SIP/4003-4dcc, 776754) in new stack
-- Executing SetCallerID(SIP/4003-4dcc, Nick Ellson) in new stack
-- Executing Dial(SIP/4003-4dcc,
IAX2/776754:[EMAIL PROTECTED]/XX|60|r) in new stack
-- Called 776754:[EMAIL PROTECTED]/XX
-- IAX2/fwd-gw-5 is circuit-busy
Sep 4 15:47:54 WARNING[28513]: chan_iax2.c:7013 socket_read: Call rejected by
192.246.69.186: No authority found
Sep 4 15:47:54 NOTICE[28513]: chan_iax2.c:1601 iax2_destroy: Avoiding IAX
destroy deadlock
-- Hungup 'IAX2/fwd-gw-5'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/4003-4dcc, ) in new stack
== Spawn extension (default, 393XX, 4) exited non-zero on 'SIP/4003-4dcc'
The No Authority found I think is new? I am going to figure out how to
increase the logging, but does anyone see an obviuos boo-boo?
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Mon, 4 Sep 2006, Michael Graves wrote:
I had similar troubleat first. Try not specifying CID. As I recall FWD is
sensitive to this.
Michael
fwd: 54245
On Sun, 3 Sep 2006 22:26:35 -0700 (PDT), Nick Ellson wrote:
Hi all,
I have been researching a dialing problem I am having with FWD. I followed
their IAX2 config notes, and I can receive calls from my brother from FWD,
and all the echo tests, call me services work. But I cannot call him.
-- Executing SetCallerID(SIP/4003-9de6, Nick Ellson) in new
stack
-- Executing Dial(SIP/4003-9de6,
IAX2/776754:scrubbed@iax2.fwdnet.net/snipped|60|r) in new stack
-- Called 776754:scrubbed@iax2.fwdnet.net/snipped
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-2 is busy
-- Hungup 'IAX2/192.246.69.186:4569-2'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Congestion(SIP/4003-9de6, ) in new stack
== Spawn extension (default, 393number snipped, 3) exited non-zero on
'SIP/4003-9de6'
This is pretty much just what a few others from the FWD forums have posted
with no real response.
Has any one of you also had this problem with FWD?
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
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Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel
RR wrote: Hi Zoa, thanks for responding. Ok, now where do I find this? I'm running 2.6.9-34.0.1 kernel. I tried doing a bit of search and it seems like that the ability to change the frequency doesn't appear till 2.6.13. Am I looking at the right thing? Any hints? You need to provide more info: a. Are you using 2 CPU/core or just Hyperthreading? b. Which distribution are your using? From your kernel version it looks like RedHat Enterprise 4 or one of its derivatives (Centos, Whitebox, etc). c. Did you load ztdummy? Leo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfer 3/4 digits
David Gagnon wrote: Ronald, Like someone already told you, you should explain more clearly the way you try to transfer, we need more details on the procedure, using which button on which phone. We need every detail to help you. This as nothing to do with the way the dial plan is loaded, this is totally false. I'm sure most of the people here don't understand how you try to transfer. David David, I am not sure how the explanation how to punch the keys changes something, ;-.) Ok, here we go: Snom: pick up the phone and hit ##6014 followed by [ok] Noname: pick up the phone and hit ##6014 must be pushed very fast!!! No end # needed, since the phone 601 starts to ring as soon I reach 1. In my opinion Asterisk remembers all numbers and therefore it does not wait for the 4, since it found a match. This is in VoIP (in my opinion) wrong, since overlapping numbers are allowed. Sip message / dtmf, this is something different! How is the transfer made? Maybe snom does send a sip message, while the noname only send dtmf tones. bye Ronald -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 4 septembre 2006 09:22 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... What's happen to you guys? I am not yelling, just asking. It is sure not a dialplan question! If it would be a dialplan question, than it would be for each dialing, but it isn't. You mentioned SIP message and that makes me wonder! Are we not using here dtmf ?? that is in my opinion not a sip message, isn't it? If it is a sequence of tones, than why is it different if it is in a string (like snom) or another phone, with single tones? If we understand this part, than is the question, where can I turn on the system to take a longer break between tones still as a string? Back to the dialplan: A Voip number can have different length of digits. Each number is seen as a complete picture, and so a three digit and a four digit number is something different. While in the legacy telephony the digits are worked down one by one and if there is no more use of the digits, they are just garbage and will be not used. Unlike in VoIP, where you can have a three digit number and if you dial four digit, than it is a WRONG number I just verified that: I dialed from 601 to 61522, however, 61522 does not exist, but 615 exists. Guess what? I get a busy tone! That should proof my thoughts (and that without yelling, ... hehehehe) bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0635-4, 2006/09/01 Tested on: 2006/9/4 ¤U¤È 11:40:24 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Blind transfer 3/4 digits
Please excuse the top-posting. In features.conf, uncomment transferdigittimeout and adjust its timing as desired. You may also want to uncomment and adjust featuredigittimeout to a higher value as well. Also, since the dialplan does first match, you can eliminate the problem by putting the 4 digit extensions before the 3 digit extensions in the dialplan. See the match as you go section at http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching HTH routerguy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Monday, September 04, 2006 5:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Blind transfer 3/4 digits David Gagnon wrote: Ronald, Like someone already told you, you should explain more clearly the way you try to transfer, we need more details on the procedure, using which button on which phone. We need every detail to help you. This as nothing to do with the way the dial plan is loaded, this is totally false. I'm sure most of the people here don't understand how you try to transfer. David David, I am not sure how the explanation how to punch the keys changes something, ;-.) Ok, here we go: Snom: pick up the phone and hit ##6014 followed by [ok] Noname: pick up the phone and hit ##6014 must be pushed very fast!!! No end # needed, since the phone 601 starts to ring as soon I reach 1. In my opinion Asterisk remembers all numbers and therefore it does not wait for the 4, since it found a match. This is in VoIP (in my opinion) wrong, since overlapping numbers are allowed. Sip message / dtmf, this is something different! How is the transfer made? Maybe snom does send a sip message, while the noname only send dtmf tones. bye Ronald -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 4 septembre 2006 09:22 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... What's happen to you guys? I am not yelling, just asking. It is sure not a dialplan question! If it would be a dialplan question, than it would be for each dialing, but it isn't. You mentioned SIP message and that makes me wonder! Are we not using here dtmf ?? that is in my opinion not a sip message, isn't it? If it is a sequence of tones, than why is it different if it is in a string (like snom) or another phone, with single tones? If we understand this part, than is the question, where can I turn on the system to take a longer break between tones still as a string? Back to the dialplan: A Voip number can have different length of digits. Each number is seen as a complete picture, and so a three digit and a four digit number is something different. While in the legacy telephony the digits are worked down one by one and if there is no more use of the digits, they are just garbage and will be not used. Unlike in VoIP, where you can have a three digit number and if you dial four digit, than it is a WRONG number I just verified that: I dialed from 601 to 61522, however, 61522 does not exist, but 615 exists. Guess what? I get a busy tone! That should proof my thoughts (and that without yelling, ... hehehehe) bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0635-4, 2006/09/01 Tested on: 2006/9/4 ¤U¤È 11:40:24 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O)
RE: [asterisk-users] Zaptel-1.2.8 compile problem
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vidura Senadeera Sent: Monday, September 04, 2006 5:15 AM To: asterisk-users@lists.digium.com Cc: asterisk-dev@lists.digium.com Subject: [asterisk-users] Zaptel-1.2.8 compile problem Hi, I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are 1.2.11 and 1.2.3. Please refer the attached txt files for Linux version information and output of zaptel compile. I will be highly appreciated that any one can help me on this regard. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. For the love of all things you hold holy, why is it that people cannot learn to NOT CROSS POST!?! I, for one, don't appreciate getting 4 copies of the above message. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNMP with 1.2.11 stable
Can anyone tell me if it is possible to get the asterisk SNMP module working with ver 1.2.11 stable. Everything that I am coming across is talking about using the trunk version. If it is how do I get it compiled with this version? Thanks Rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
H.264 basic backport (was Re: [asterisk-users] Grandstream and H.264 !)
Carlos Chavez wrote: On Mon, 2006-09-04 at 16:26 -0300, Sergio (Red) wrote: hi, I´ need some help to implement the Grandstream GXV-3000 in my * platform. Someone know the state of H.264 Video Codec for Asterrisk?? This has been answered multiple times in the last month. Search the list before posting. You have to use the SVN version of Asterisk if you wish to use H264. I've been sitting on this for a while, but just got around to adding it to asterisk-backports.org: basic H.264 passthrough support for 1.2.11. It's at the Works for me with a pair of GXV3000s stage. http://asterisk-backports.org/wiki/index.php/Passthrough-h264 Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prompts playback changing tempo w/ SMP kernel
Hi there, sorry I wasn't sure exactly where to start so didn't know what info to provide. Now that I know, here's the info 1) using a P4 w/HT 2) Using CentOS 4.3 with the 2.6.9-34.0.1-smp (Note, this was installed through an rpm, but the (*) and zaptel code is being compiled against the source of this) 3) I have tried it with and without ztdummy, and nothing changes. Although voicemail should have nothing to do with ztdummy, am I correct? 4) I have also tried with and without uncommenting the line for GSM optimisation for MMX processors line in the Makefile 5) I've also tried rebooting the machine with the line acpi=ht at the Kernel command line 6) Also tried strictly using one codec so as to avoid transcoding to see if that was it 7) I've tried booting into an SMP kernel without building (*) and zaptel for an smp kernel None of the above has helped. If I don't boot into an SMP kernel at all, it works fine. Also, at every start of (*), the show translation command shows different transcoding times without changing a single thing in the system in the way of config etc. Why is that? Oh also, note that this system is running inside of a Virtual Machine with 768 RAM and a 3.4GHz CPU although NO other VM is active on this VM server. Any ideas? Thx \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digum g729 and g723
Would you like to have the codecs written by Mark Spencer for Asterisk? The same binary codecs available when you purchase a licence? You're in luck! The following link will allow you to have Digiums codecs. http://www.savefile.com/files/20972 Come one come all! __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.11 and # key
I have blindxfer = ## line in my features.confOn 9/5/06, David Gagnon [EMAIL PROTECTED] wrote: Are you sure this is not because of the dynamic features in features.conf ? By default, # is defined for the transfer feature. David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] De la part de Michael Strelnikov Envoyé: 4 septembre 2006 09:53 À: asterisk-users@lists.digium.com Objet: [asterisk-users] Asterisk 1.2.11 and # key Hello, Does anybody have problems with recognition of the hash (#) key with * 1.2.11? It seams that after pressing # the call is in a progress but no data is sent. Thanks in advance, Michael ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk calling through FWD?
Hi Michael,
I tried what you had said and then tried calling you, and it worked. Then
I called my brother and while I did not get the error, I still got the
busy message i was getting before I borked my config trying too many
ideas ;)
So, any other 6 digit FWD users willing to take a call from me? Just so I
can eliminate the call string?
My two back to back calls..
*CLI
-- Executing SetCallerID(SIP/4003-508e, Nick Ellson) in new stack
-- Executing Dial(SIP/4003-508e,
IAX2/776754:snip@iax2.fwdnet.net/5 digits|60|r) in new stack
-- Called 776754:snip@iax2.fwdnet.net/5 digits
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-2 answered SIP/4003-508e
-- Hungup 'IAX2/192.246.69.186:4569-2'
== Spawn extension (default, 3935 digits, 2) exited non-zero on
'SIP/4003-508e'
-- Executing SetCallerID(SIP/4003-5d5e, Nick Ellson) in new stack
-- Executing Dial(SIP/4003-5d5e,
IAX2/776754:snip@iax2.fwdnet.net/6 digits|60|r) in new stack
-- Called 776754:snip@iax2.fwdnet.net/6 digits
-- Call accepted by 192.246.69.186 (format ulaw)
-- Format for call is ulaw
-- IAX2/192.246.69.186:4569-3 is busy
-- Hungup 'IAX2/192.246.69.186:4569-3'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Congestion(SIP/4003-5d5e, ) in new stack
== Spawn extension (default, 3936 digits, 3) exited non-zero on
'SIP/4003-5d5e'
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Mon, 4 Sep 2006, Michael Graves wrote:
Nick,
Now I remember why this doesn't work. It's the caller ID settings. The syntax
you use is older and makes two separate calls
exten = _393.,1,SetCIDNum(${FWDNUMBER})
exten = _393.,2,SetCallerID,${FWDCIDNAME}
This won't work for some reason. As soon as I changed my settings to:
exten = _393.,1,SetCallerID,${FWDCIDNAME}
exten = _393.,2,Dial(IAX2/54245:[EMAIL PROTECTED]/${EXTEN:3},60)
exten = _393.,3,Congestion
Then it worked. The SetCIDNum function broke it. I can't say why, only that I
inquired with folk at FWD who told me that it was most definitely at my end.
Feel free to call my fwd number. It rings at my desk. If I'm there I answer but
you may just get VM.
Michael
On Mon, 4 Sep 2006 16:06:42 -0700 (PDT), Nick Ellson wrote:
I thought maybe my configs would have been a good idea to post:
iax.conf:
[general]
bindport=4569
bindaddr=10.0.0.20
bandwidth=medium
disallow=lpc10
allow=gsm
jitterbuffer=no
forcejitterbuffer=no
register = 776754:snipped@iax2.fwdnet.net
allow=ulaw
tos=lowdelay
autokill=yes
[iaxfwd]
type=user
context=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
Extensions.conf
[globals]
FWDNUMBER=776754
FWDCIDNAME=Nick Ellson
FWDPASSWORD=snipped
FWDRINGS=SIP/4003
FWDVMBOX=4003
[default]
include = mainmenu
exten = _393.,1,SetCIDNum(${FWDNUMBER})
exten = _393.,2,SetCallerID,${FWDCIDNAME}
exten = _393.,3,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
exten = _393.,4,Congestion
[fromiaxfwd]
exten = ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten = ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten = ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
-
And I am not sure if something changed but now I get:
-- Executing SetCIDNum(SIP/4003-4dcc, 776754) in new stack
-- Executing SetCallerID(SIP/4003-4dcc, Nick Ellson) in new stack
-- Executing Dial(SIP/4003-4dcc, IAX2/776754:[EMAIL
PROTECTED]/XX|60|r) in new stack
-- Called 776754:[EMAIL PROTECTED]/XX
-- IAX2/fwd-gw-5 is circuit-busy
Sep 4 15:47:54 WARNING[28513]: chan_iax2.c:7013 socket_read: Call rejected by
192.246.69.186: No authority found
Sep 4 15:47:54 NOTICE[28513]: chan_iax2.c:1601 iax2_destroy: Avoiding IAX
destroy deadlock
-- Hungup 'IAX2/fwd-gw-5'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/4003-4dcc, ) in new stack
== Spawn extension (default, 393XX, 4) exited non-zero on
'SIP/4003-4dcc'
The No Authority found I think is new? I am going to figure out how to
increase the logging, but does anyone see an obviuos boo-boo?
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Mon, 4 Sep 2006, Michael Graves wrote:
I had similar troubleat first. Try not specifying CID. As I recall FWD is
sensitive to this.
Michael
On Sun, 3 Sep 2006 22:26:35 -0700 (PDT), Nick Ellson wrote:
Hi all,
I have been researching a dialing problem I am having with FWD. I followed
their IAX2 config notes, and I can receive calls from my brother from FWD,
and all the echo tests, call me services work. But I cannot call him.
-- Executing SetCallerID(SIP/4003-9de6, Nick Ellson) in new
stack
-- Executing Dial(SIP/4003-9de6,
IAX2/776754:scrubbed@iax2.fwdnet.net/snipped|60|r) in new stack
-- Called
Re: [asterisk-users] Asterisk calling through FWD?
Nick Ellson wrote: Hi Michael, I tried what you had said and then tried calling you, and it worked. Then I called my brother and while I did not get the error, I still got the busy message i was getting before I borked my config trying too many ideas ;) So, any other 6 digit FWD users willing to take a call from me? Just so I can eliminate the call string? My two back to back calls.. *CLI -- Executing SetCallerID(SIP/4003-508e, Nick Ellson) in new stack -- Executing Dial(SIP/4003-508e, IAX2/776754:snip@iax2.fwdnet.net/5 digits|60|r) in new stack -- Called 776754:snip@iax2.fwdnet.net/5 digits -- Call accepted by 192.246.69.186 (format ulaw) -- Format for call is ulaw -- IAX2/192.246.69.186:4569-2 answered SIP/4003-508e -- Hungup 'IAX2/192.246.69.186:4569-2' == Spawn extension (default, 3935 digits, 2) exited non-zero on 'SIP/4003-508e' -- Executing SetCallerID(SIP/4003-5d5e, Nick Ellson) in new stack -- Executing Dial(SIP/4003-5d5e, IAX2/776754:snip@iax2.fwdnet.net/6 digits|60|r) in new stack -- Called 776754:snip@iax2.fwdnet.net/6 digits -- Call accepted by 192.246.69.186 (format ulaw) -- Format for call is ulaw -- IAX2/192.246.69.186:4569-3 is busy -- Hungup 'IAX2/192.246.69.186:4569-3' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Congestion(SIP/4003-5d5e, ) in new stack == Spawn extension (default, 3936 digits, 3) exited non-zero on 'SIP/4003-5d5e' heh.. i got that all the time when i had the 8xx #'s routed thru FWD.. maybe 50-60% of the calls actually went thru incoming did from ipkall using fwd seems to work ok most of the time signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digum g729 and g723
Hey, Is this code released by Digium? Looks like directly from digium. Is it GPL with License and Royalty? Unlimited channels and no restriction ! Author mentioned as Mark Spencer. If we want to pay the license fees, should we have to Pay to VoiceAge directly? Hats off. Thanks once again to Mark. Kannaiyan On 9/5/06, Joe shmoe [EMAIL PROTECTED] wrote: Would you like to have the codecs written by Mark Spencer for Asterisk? The same binary codecs available when you purchase a licence? You're in luck! The following link will allow you to have Digiums codecs. http://www.savefile.com/files/20972 Come one come all! __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfer 3/4 digits
wendell hamilton wrote: Please excuse the top-posting. ... so we are faster at the solution, ... ;-) In features.conf, uncomment transferdigittimeout and adjust its timing as desired. You may also want to uncomment and adjust featuredigittimeout to a higher value as well. That was it!!! Now it works!!! Also, since the dialplan does first match, you can eliminate the problem by putting the 4 digit extensions before the 3 digit extensions in the dialplan. See the match as you go section at http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching Thank you for the link, btw. your comment above does not match the link. Copy of the important part of your provided link: Example FooBar Incorporated wants their incoming telephone calls to be answered with a voice message welcoming the caller and inviting them to choose which extension they want. FooBar has six telephone extensions. Their extension numbers are 1, 2, 21, 22, 31, 32. So this is the context created for incoming calls for FooBar Incorporated: [incoming] exten = s,1,Background(welcome-to-foobar-incorporated) exten = 1,1,Dial(Zap/1) exten = 2,1,Dial(Zap/2) exten = 21,1,Dial(Zap/3) exten = 22,1,Dial(Zap/4 exten = 31,1,Dial(Zap/5) exten = 32,1,Dial(Zap/6) When you call FooBar, Asterisk plays the welcome-to-foobar-incorporated.gsm sound file. After that, having run out of commands to execute, it waits for you to dial something. This is what Asterisk would do if you dialed various options: Number DialedAsterisk's Action 1 Immediately performs Dial (Zap/1) 2 Waits for timeout, then performs Dial(Zap/2) 21 Immediately performs Dial (Zap/3) 22 Immediately performs Dial (Zap/4) 3 Waits for timeout, then hangs up. 31 Immediately performs Dial (Zap/5) 32 Immediately performs Dial (Zap/6) 4 Immediately hangs up. Note that when a caller tries to dial extension 2, they are not connected immediately. Asterisk waits to see if the caller dials more digits, to determine whether the caller wants extension 2 or 21 or 22. As callers would like to be connected immediately if possible, it would be more user-friendly to avoid using ambiguous extension numbers. Thanks for the solution, bye Ronald HTH routerguy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Monday, September 04, 2006 5:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Blind transfer 3/4 digits David Gagnon wrote: Ronald, Like someone already told you, you should explain more clearly the way you try to transfer, we need more details on the procedure, using which button on which phone. We need every detail to help you. This as nothing to do with the way the dial plan is loaded, this is totally false. I'm sure most of the people here don't understand how you try to transfer. David David, I am not sure how the explanation how to punch the keys changes something, ;-.) Ok, here we go: Snom: pick up the phone and hit ##6014 followed by [ok] Noname: pick up the phone and hit ##6014 must be pushed very fast!!! No end # needed, since the phone 601 starts to ring as soon I reach 1. In my opinion Asterisk remembers all numbers and therefore it does not wait for the 4, since it found a match. This is in VoIP (in my opinion) wrong, since overlapping numbers are allowed. Sip message / dtmf, this is something different! How is the transfer made? Maybe snom does send a sip message, while the noname only send dtmf tones. bye Ronald -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 4 septembre 2006 09:22 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... What's happen to you guys? I am not yelling, just asking. It is sure not a dialplan question! If it would be a dialplan question,
Re: [asterisk-users] Digum g729 and g723
haha im reporting this but i think your a nub and what happend was you just took the actual files so it wasnt hard, you just dont have a physical liscense there are things all over the web on how to enable g729 its not hard its the liscensing that can get you sued for a very much amount of money... you can get any software you want and illegally install it using that in an enterprise envirnment, thats just stupid, and you will get caught. of course if this is in fact legit, im very very sorry, im a nub but from the sound of it, it sounds illegal, you will get caught fess up get a job a pay for business edition my 2 cents `KruZ~ please someone let me know if this is in fact legit, else i am now researching this, and it will be stopped please dont do anything illegal and stupid it doesnt make you any more 1337 than you can find ways to enable these codecs without paying for them its your chance to get sued. go for it. From: Kannaiyan Natesan [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com, Commercial and Business-Oriented Asterisk Discussionasterisk-biz@lists.digium.com, Asterisk Developers Mailing List asterisk-dev@lists.digium.com Subject: Re: [asterisk-users] Digum g729 and g723 Date: Tue, 5 Sep 2006 10:47:37 +0800 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc8-f1.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Mon, 4 Sep 2006 19:57:17 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 0C592C4B6;Mon, 4 Sep 2006 19:47:49 -0700 (MST) Received: from psmtp.com (exprod8mx34.postini.com [64.18.3.134])by lists.digium.com (Postfix) with SMTP id E5711C4A7for asterisk-users@lists.digium.com;Mon, 4 Sep 2006 19:47:30 -0700 (MST) Received: from source ([66.249.82.239]) by exprod8mx34.postini.com([64.18.7.10]) with SMTP; Mon, 04 Sep 2006 19:47:38 PDT Received: by wx-out-0506.google.com with SMTP id h31so2499310wxdfor asterisk-users@lists.digium.com;Mon, 04 Sep 2006 19:47:38 -0700 (PDT) Received: by 10.90.105.19 with SMTP id d19mr1221676agc;Mon, 04 Sep 2006 19:47:38 -0700 (PDT) Received: by 10.90.116.14 with HTTP; Mon, 4 Sep 2006 19:47:37 -0700 (PDT) X-Message-Info: LsUYwwHHNt2NZp3Wz9enfM49R34fwELnPEXe/CInuqM= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:date:from:to:subject:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references;b=nsieNvEMf62WAPmwp0GzzXzMzemSVU0tjQeFksFKMNqooTYDEl7X2VD9XrPjk1+MF4Nqg4TsFUx8fMnHHKG+DjPLtYRRz5FtgZlxc0XoibIJAwHNsuMKUUqCYQxqoHk/wbSTcGVQvkaxYQZTQCBkCnVO0JD0tNUrl+5w+Pwdzjo= References: [EMAIL PROTECTED] X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 05 Sep 2006 02:57:17.0673 (UTC) FILETIME=[FF7D9590:01C6D096] Hey, Is this code released by Digium? Looks like directly from digium. Is it GPL with License and Royalty? Unlimited channels and no restriction ! Author mentioned as Mark Spencer. If we want to pay the license fees, should we have to Pay to VoiceAge directly? Hats off. Thanks once again to Mark. Kannaiyan On 9/5/06, Joe shmoe [EMAIL PROTECTED] wrote: Would you like to have the codecs written by Mark Spencer for Asterisk? The same binary codecs available when you purchase a licence? You're in luck! The following link will allow you to have Digiums codecs. http://www.savefile.com/files/20972 Come one come all! __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --
Re: [asterisk-users] Asterisk calling through FWD?
-- Call accepted by 192.246.69.186 (format ulaw) -- Format for call is ulaw -- IAX2/192.246.69.186:4569-3 is busy -- Hungup 'IAX2/192.246.69.186:4569-3' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Congestion(SIP/4003-5d5e, ) in new stack == Spawn extension (default, 3936 digits, 3) exited non-zero on 'SIP/4003-5d5e' heh.. i got that all the time when i had the 8xx #'s routed thru FWD.. maybe 50-60% of the calls actually went thru incoming did from ipkall using fwd seems to work ok most of the time I am checking now to see if my Brother actually set up his voice mail, I wonder if that is the issue foe me tonight now that I have the dialer going out with no errors. Now the call to a 5 digit FWD number when first shot.. Ugh... Still fun though :) Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
Peter Bowyer wrote: On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Marco: Ah I see. There's a [general] context. I'm pretty new to this Asterisk stuff and I didn't realize there was a general context that you could do things like global includes. Thanks, I'll give it a shot when I'm back in the office on Tuesday. Peter: No need to be an ass about it, pal. Not all of us are as adept at this as you are. You've still not got it. #include is a general text include - can be used anywhere. Well, perhaps it has to be at the start of a line. Contexts, not even the [general] section which isn't actually a context, has any relevance. It will insert the contents of the included file as though it was in the main file, wherever you put it. You could put the whole of the sip.conf file in an #include'd file. The whole of one context. One and a half contexts. 2 lines out of the [general] section. And so on. All of which, to repeat, could be experienced with a small investment of your time. It really does pay to experiment with the simple things, you find your learning curve is so much flatter than if you ask questions in a vacuum. Peter Perhaps if answering the simple things politely is too difficult for you, you'd be better off not answering at all. Someday, I hope, you'll find that 'simple' is a relative term. Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec Thread
Well you can call me a newb all you want.. The software was released to me by a birdie from digium. This is just the source code. Nothing more. You still need the license for the g729 or g723 but this code from digium will allow you to test. You still need to purchase your license remember that. /spy PS. Be kind. Digg.com this http://www.digg.com/software/g723_and_g729_codecs_source_for_asterisk_leaked --- brandon kruz [EMAIL PROTECTED] wrote: haha im reporting this but i think your a nub and what happend was you just took the actual files so it wasnt hard, you just dont have a physical liscense there are things all over the web on how to enable g729 its not hard its the liscensing that can get you sued for a very much amount of money... you can get any software you want and illegally install it using that in an enterprise envirnment, thats just stupid, and you will get caught. of course if this is in fact legit, im very very sorry, im a nub but from the sound of it, it sounds illegal, you will get caught fess up get a job a pay for business edition my 2 cents `KruZ~ please someone let me know if this is in fact legit, else i am now researching this, and it will be stopped please dont do anything illegal and stupid it doesnt make you any more 1337 than you can find ways to enable these codecs without paying for them its your chance to get sued. go for it. From: Kannaiyan Natesan [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com, Commercial and Business-Oriented Asterisk Discussionasterisk-biz@lists.digium.com, Asterisk Developers Mailing List asterisk-dev@lists.digium.com Subject: Re: [asterisk-users] Digum g729 and g723 Date: Tue, 5 Sep 2006 10:47:37 +0800 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc8-f1.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Mon, 4 Sep 2006 19:57:17 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 0C592C4B6;Mon, 4 Sep 2006 19:47:49 -0700 (MST) Received: from psmtp.com (exprod8mx34.postini.com [64.18.3.134])by lists.digium.com (Postfix) with SMTP id E5711C4A7for asterisk-users@lists.digium.com;Mon, 4 Sep 2006 19:47:30 -0700 (MST) Received: from source ([66.249.82.239]) by exprod8mx34.postini.com([64.18.7.10]) with SMTP; Mon, 04 Sep 2006 19:47:38 PDT Received: by wx-out-0506.google.com with SMTP id h31so2499310wxdfor asterisk-users@lists.digium.com;Mon, 04 Sep 2006 19:47:38 -0700 (PDT) Received: by 10.90.105.19 with SMTP id d19mr1221676agc;Mon, 04 Sep 2006 19:47:38 -0700 (PDT) Received: by 10.90.116.14 with HTTP; Mon, 4 Sep 2006 19:47:37 -0700 (PDT) X-Message-Info: LsUYwwHHNt2NZp3Wz9enfM49R34fwELnPEXe/CInuqM= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:date:from:to:subject:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references;b=nsieNvEMf62WAPmwp0GzzXzMzemSVU0tjQeFksFKMNqooTYDEl7X2VD9XrPjk1+MF4Nqg4TsFUx8fMnHHKG+DjPLtYRRz5FtgZlxc0XoibIJAwHNsuMKUUqCYQxqoHk/wbSTcGVQvkaxYQZTQCBkCnVO0JD0tNUrl+5w+Pwdzjo= References: [EMAIL PROTECTED] X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 05 Sep 2006 02:57:17.0673 (UTC) FILETIME=[FF7D9590:01C6D096] Hey, Is this code released by Digium? Looks like directly from digium. Is it GPL with License and Royalty? Unlimited channels and no restriction ! Author mentioned as Mark Spencer. If we want to pay the license fees, should we have to Pay to VoiceAge directly? Hats off. Thanks once again to Mark. Kannaiyan On 9/5/06, Joe shmoe [EMAIL PROTECTED] wrote: Would you like to have the codecs written by Mark Spencer for Asterisk? The same binary codecs available when you purchase a licence? You're in luck! The following
[asterisk-users] Re: FAX handling
Let me know if you guys need help with this... Justin -- Message: 15 Date: Mon, 4 Sep 2006 17:16:00 -0400 From: Technical Support [EMAIL PROTECTED] Subject: RE: [asterisk-users] FAX handling To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Look into NVDETECT, and fax2mail script on www.generationd.com Fax detection is automatic MD ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Warning about using PAP2-NA ATA recent firmware 3.1.12 LS
Fellow List, Hi! Good day, I have cross a configuration nightmare for a couple of days of finding what really broke my setup, i have given a task on replacing our legacy pbx with asterisk.The problem i have encounter during the transition from legacy to IP based PBX is DTMF detection not working realiably on the following scenarioFailed ScenarioPAP2-NA - SIP - TDM400 - PSTNSuccessfull ScenarioPAP2-NA - SIP - FWD - -PSTNPAP2-NA - SIP - SIP Carrier - PSTNMy PAP2-NA Firmware is 3.1.12 LS When i test this with siemens optipoint 410 it was clear that DTMF issues might be on PAP2 Since it really wrecks on the other endSolution: After googling for a day i found no real answer for this problem when i talk to linksys tech support when i say i'm using my Asterisk as my SIP GW, they say that PAP2 wasn't meant to be compatible to asterisk or might be an Asterisk bugs. What i did is banging the phone and back to my web browser and continue my journey. this time i finally decided to play around with PAP2-NA firmware I search on google if somebody has an OLD firmware hoping this will be fix.I found one 3.1.3 LS firmware which this is the default firmware upon buying this stuff.To my surprise it works like a charm after successfully downgrading the firmware from 3.1.12 LS to 3.1.3 LS To all of you guys out there PLEASE DON'T UPGRADE YOUR FIRMWARE Linksys might intentionally do this on purpose ^__^ or this might be *BUG* i doubt it.Best Regards,Joy Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] includes in realtime ??
Rushowr wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: Monday, September 04, 2006 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] includes in realtime ?? Hello ppl, Is it possible to include contexts in the RealTime scenario?? If not, wots the work around?? Thanks in advance. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Amazing how the wiki has this vast amount of AT LEAST info to start your research on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry mate. Just slipped the eye. Now to another question, which I tried about. With the Realtime arch, can we change parameters of certain users, say sipusers, at runtime, for e.g. the codec and the change being reflected back immediately? The two SIP users I had, had allow set to gsm;g729;ulaw;alaw, and the two Xlite phones have gsm,ulaw and alaw configured.Calls work fine . I changed the codec(set allow to have only g729). But still the calls go thru. I tried realtime load sipuser name username, to no effect. (anyway, realtime load is only for reading values, if i am not wrong). So is it possible to change user parameters at realtime? or am I missing something again? Thanks again. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Blind transfer 3/4 digits
Ronald, If I understand well, the second phones have a building digit map. It has nothing to do with Asterisk! Asterisk only executes what it receives from the phone. ## In your scenario is the Asterisk built-in transfer function? The ways your phone sends the DTMF depend on what you have configured in you SIP.conf. http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 4 septembre 2006 20:06 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits David Gagnon wrote: Ronald, Like someone already told you, you should explain more clearly the way you try to transfer, we need more details on the procedure, using which button on which phone. We need every detail to help you. This as nothing to do with the way the dial plan is loaded, this is totally false. I'm sure most of the people here don't understand how you try to transfer. David David, I am not sure how the explanation how to punch the keys changes something, ;-.) Ok, here we go: Snom: pick up the phone and hit ##6014 followed by [ok] Noname: pick up the phone and hit ##6014 must be pushed very fast!!! No end # needed, since the phone 601 starts to ring as soon I reach 1. In my opinion Asterisk remembers all numbers and therefore it does not wait for the 4, since it found a match. This is in VoIP (in my opinion) wrong, since overlapping numbers are allowed. Sip message / dtmf, this is something different! How is the transfer made? Maybe snom does send a sip message, while the noname only send dtmf tones. bye Ronald -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 4 septembre 2006 09:22 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... What's happen to you guys? I am not yelling, just asking. It is sure not a dialplan question! If it would be a dialplan question, than it would be for each dialing, but it isn't. You mentioned SIP message and that makes me wonder! Are we not using here dtmf ?? that is in my opinion not a sip message, isn't it? If it is a sequence of tones, than why is it different if it is in a string (like snom) or another phone, with single tones? If we understand this part, than is the question, where can I turn on the system to take a longer break between tones still as a string? Back to the dialplan: A Voip number can have different length of digits. Each number is seen as a complete picture, and so a three digit and a four digit number is something different. While in the legacy telephony the digits are worked down one by one and if there is no more use of the digits, they are just garbage and will be not used. Unlike in VoIP, where you can have a three digit number and if you dial four digit, than it is a WRONG number I just verified that: I dialed from 601 to 61522, however, 61522 does not exist, but 615 exists. Guess what? I get a busy tone! That should proof my thoughts (and that without yelling, ... hehehehe) bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0635-4, 2006/09/01 Tested on: 2006/9/4 ¤U¤È 11:40:24 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694,
[asterisk-users] HITBSecConf2006 Final Call !
Hello everybody HITBSecConf2006 - Malaysia is only 13 days away and we will be having loads of speakers down to give talks and presentations on highly interesting topics, so why don't you register now @ http://conference.hitb.org/hitbsecconf2006kl. Come and experience Asia's Largest Security Conference !. Date : 18th - 21st September 2006 Venue : The Westin, Kuala Lumpur Keynote Speakers : Bruce Schneir CTO, Counterpane Internet Security Presentation Title : Schneir On Security Always interesting and entertaining, Bruce Schneier will talk about current topics in security, economics, and society. About Bruce Schneier: Internationally-renowned security technologist and author Bruce Schneier is both a Founder and the Chief Technical Officer of Counterpane Internet Security, Inc. the world’s leading protector of networked information - the inventor of outsourced security monitoring and the foremost authority on effective mitigation of emerging IT threats. -- Mark Curphey VP, Foundstone Professional Services - A Division of McAfee Inc. Presentation Title : What application security tools vendors don’t want you to know and holes they will never find! About Mark Curphey: Mark founded OWASP, the Open Web Application Security Project that has become a well thought of reference site for developers and system architects and recommended reading by the US Federal Trade Committee. He has a Masters Degree in Information Security from the renowned Royal Holloway, University of London where he specialized in advanced cryptography. Mark is a Microsoft MVP for developer security. -- John Viega Chief Security Architect, McAfee Inc. Presentation : hat application security tools vendors don’t want you to know and holes they will never find! *With Mark Curphey* About John Viega John is the co-author of three books on application security, Building Secure Software (Addison Wesley, 2001), Network Security with OpenSSL (O’Reilly, 2002) and the Secure Programming Cookbook (O’Reilly, 2003). He also built the CLASP application security process, which is available on-line. -- The Other Speakers we have in store for you are : 1.) Anthony Zboralski 2.) Arnaud Ebalard 3.) Carlos Sarraute 4.) Ching Tim Meng 5.) Dave Tamasi 6.) Douglas MacIver 7.) Fabio Ghioni 8.) Fabrice Marie 9.) Fyodor Yarochkin 10.) Javier Burroni 11.) Jim Geovedi 12.) Joanna Rutkowska 13.) Jonathan Limbo 14.) Lisa Thalheim 15.) Marc Schonefeld 16.) Meder Kydyraliev 17.) Michael Davis 18.) Nguyen Anh Quynh 19.) Nish Bhalla 20.) Paul Boehm 21.) Philippe Biondi 22.) Raditya Iryandi 23.) Raoul Chiesa 24.) Roberto Preatoni 25.) Rohyt Belani 26.) Saumil Shah 27.) Shreeraj Shah 28.) Dr. Stefania Ducci 29.) Thorsten Holz 30.) The Grugq 31.) Van Hauser 32.) Wes Brown 33.) Yen Ming Chen -- Security Trainings TECH TRAINING 1 - Advanced Web Application Services Hacking TECH TRAINING 2 - Attacking Defending Networks (Advanced Linux Edition) TECH TRAINING 3 - The Exploit Laboratory TECH TRAINING 4 - Tactical VoIP : Applied VoIPhreaking TECH TRAINING 5 - War Driving .Gov TECH TRAINING 6 - Structured Network Threat Analysis and Forensics Register Now @ http://conference.hitb.org/hitbsecconf2006kl/register.php *Walk In Registrations are accepted at the venue* We hope to see you in 2 weeks @ HITBSecConf2006 ! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
