Re: [Asterisk-Users] Call transfer with voicemail password
http://www.voip-info.org/wiki/view/Asterisk+authenticate+using+voicemail+passwords Cheers 2005/12/1, Joe Pukepail [EMAIL PROTECTED]: Look into the findme feature, there is a patch on the bug tracker to add this feature. I believe that someone shows how to do it in the dial plan. I plan on implementing this, but haven't gotten around to it yet. On 11/30/05, Benjamin Lenard [EMAIL PROTECTED] wrote: Hi, I'm trying to have an extension ring my SIP phone then try my cell phone. I can transfer the call fine to the cell but I want it to ask for a pin , voicemail pin, before transferring the call. This is so if my cell's voicemail answers , the call doesn't transfer to it. Any ideas? Thanks, 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 ___ --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 -- Giovanni Miano ___ --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] Motherboard choice for asterisk?
If u want kernel 2.6 dont use SMP support I use asus with celeron,amd and it works fine. 2005/12/1, John Brookes [EMAIL PROTECTED]: I am putting together a box to run asterisk. Which version of linux and MB-cpu do you suggest? TIA John B ___ --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 -- Giovanni Miano ___ --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] Voice Mail
U can use AMI (asterisk management interface) write small application that connect to ami and check voicebox status Configure manager.conf and try telnet [asterisk's ip] 5038] 2005/12/1, Hiu Yen Onn [EMAIL PROTECTED]: I have been using xlite client, FOC. There is no sign of image displayed on the screen. Jan Saell wrote: A SIP phone with the possibility of showing message waiting can get that information from Asterisk. My EyeBeam is showing a small image of a letter in the display to show that there are messages waiting. SO you can use this without mail being sent-out. Best regards jan --On Wednesday, November 30, 2005 15:23:33 +0800 Hiu Yen Onn [EMAIL PROTECTED] wrote: How normally SIP user is informed by having a new incoming voicemail and then, how are they read their mails then i have known that, asterisk will send a mail for the users. then, how to configure the mail smtp and pop3 for asterisk to send mail then. 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 ___ --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 -- Giovanni Miano ___ --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] optipoint 410 and MWI
Hi, I want to setup the MWI on the Optipoint 410 Standard SIP. Until now haven't found any information about this. Probably anyone know the right way and other tips for this phone? Thanks and regards Wolfgang ___ --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] Compiling Asterisk 1.2 from Source on Debian Sarge- Problems
Hi! I just built Asterisk on Debian Sarge myself and it worked without any problems. Can you cut 'n paste the error messages? I can't make any sense from the output ... Greetings, Marcus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hagen Rode Sent: Wednesday, November 30, 2005 6:38 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Compiling Asterisk 1.2 from Source on Debian Sarge- Problems Hi I am trying to compile Asterisk 1.2 from source on Debian Sarge but am getting errors. I have looked at the errors, Googled extensively and now at a last resort am posting on this list. Believe me I have tried, but have come up with nothing. I've also installed the following packages from Debian Sarge UNSTABLE: gcc kernel-headers-2.4.27 bison openssl libssl0.9.7: libssl-dev libeditline0 libeditline-dev libedit-dev libedit2 libncurses5 libncurses5-dev zlib1g-dev (Note: needed for cvs head) as well as numerous other packages that I have now lost track of. The error remains the same. It would be great if someone could help me out. I'm aware that I can apt-get Asterisk, but I want to do some tweaking in the code before installing. Here is the first bit of the install message: build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp if cmp -s .cleancount .lastclean ; then echo ; else \ make clean; cp -f .cleancount .lastclean;\ fi build_tools/make_defaults_h defaults.h.tmp if cmp -s defaults.h.tmp defaults.h ; then echo ; else \ mv defaults.h.tmp defaults.h ; \ fi rm -f defaults.h.tmp for x in res channels pbx apps codecs formats agi cdr funcs utils stdtime; do make -C $x depend || exit 1 ; done make[1]: Entering directory `/opt/asterisk-1.2.0/res' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/res' make[1]: Entering directory `/opt/asterisk-1.2.0/channels' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/channels' make[1]: Entering directory `/opt/asterisk-1.2.0/pbx' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/pbx' make[1]: Entering directory `/opt/asterisk-1.2.0/apps' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/apps' make[1]: Entering directory `/opt/asterisk-1.2.0/codecs' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/codecs' make[1]: Entering directory `/opt/asterisk-1.2.0/formats' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/formats' make[1]: Entering directory `/opt/asterisk-1.2.0/agi' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/agi' make[1]: Entering directory `/opt/asterisk-1.2.0/cdr' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/cdr' make[1]: Entering directory `/opt/asterisk-1.2.0/funcs' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/funcs' make[1]: Entering directory `/opt/asterisk-1.2.0/utils' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -DNO_AST_MM `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/utils' make[1]: Entering directory `/opt/asterisk-1.2.0/stdtime' ../build_tools/mkdep -pipe -Wall
[Asterisk-Users] unable to make calls out using [EMAIL PROTECTED]
Hi as suggested in the group I have downloaded the [EMAIL PROTECTED] installed one of my PC for testing and made 2 extentions to test iam able to talk each other now i have setup one Trunk and made Out going when ever i call to out side i get a voice tone saying that all trunks are busy how can i resolve this problem ram ___ --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] unable to make calls out using [EMAIL PROTECTED]
Have u availability tranks ? 2005/12/1, ram [EMAIL PROTECTED]: Hi as suggested in the group I have downloaded the [EMAIL PROTECTED] installed one of my PC for testing and made 2 extentions to test iam able to talk each other now i have setup one Trunk and made Out going when ever i call to out side i get a voice tone saying that all trunks are busy how can i resolve this problem ram ___ --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 -- Giovanni Miano ___ --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] unable to make calls out using [EMAIL PROTECTED]
yes i got new account from provider and i have registered no one using, iam using that account for testing ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: Have u availability tranks ?2005/12/1, ram [EMAIL PROTECTED] : Hi as suggested in the group I have downloaded the [EMAIL PROTECTED] installed one of my PC for testing and made 2 extentions to test iam able to talk each other now i have setup one Trunk and made Out going when ever i call to out side i get a voice tone saying that all trunks are busy how can i resolve this problem ram ___ --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 --Giovanni Miano___--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] Compiling Asterisk 1.2 from Source on Debian Sarge- Problems
Hi Marcus, haven't you got an Unable to initialize mISDN error during asterisk startup? I have a problem with chan_misdnI'm trying to understand where is the prob...I haven't recompiled my kernel with mISDN support because Digium claims it is inside Asterisk 1.2, maybe it's my kernel.my kernel is 2.6.8-2-386 which is yours? Have you installed a particular package like for example /misdn-kernel-headers/ or /misdn-kernel-source ??/ TIA Giorgio Incantalupo Marcus Deluigi (intern) wrote: Hi! I just built Asterisk on Debian Sarge myself and it worked without any problems. Can you cut 'n paste the error messages? I can't make any sense from the output ... Greetings, Marcus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hagen Rode Sent: Wednesday, November 30, 2005 6:38 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Compiling Asterisk 1.2 from Source on Debian Sarge- Problems Hi I am trying to compile Asterisk 1.2 from source on Debian Sarge but am getting errors. I have looked at the errors, Googled extensively and now at a last resort am posting on this list. Believe me I have tried, but have come up with nothing. I've also installed the following packages from Debian Sarge UNSTABLE: gcc kernel-headers-2.4.27 bison openssl libssl0.9.7: libssl-dev libeditline0 libeditline-dev libedit-dev libedit2 libncurses5 libncurses5-dev zlib1g-dev (Note: needed for cvs head) as well as numerous other packages that I have now lost track of. The error remains the same. It would be great if someone could help me out. I'm aware that I can apt-get Asterisk, but I want to do some tweaking in the code before installing. Here is the first bit of the install message: build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp if cmp -s .cleancount .lastclean ; then echo ; else \ make clean; cp -f .cleancount .lastclean;\ fi build_tools/make_defaults_h defaults.h.tmp if cmp -s defaults.h.tmp defaults.h ; then echo ; else \ mv defaults.h.tmp defaults.h ; \ fi rm -f defaults.h.tmp for x in res channels pbx apps codecs formats agi cdr funcs utils stdtime; do make -C $x depend || exit 1 ; done make[1]: Entering directory `/opt/asterisk-1.2.0/res' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/res' make[1]: Entering directory `/opt/asterisk-1.2.0/channels' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/channels' make[1]: Entering directory `/opt/asterisk-1.2.0/pbx' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/pbx' make[1]: Entering directory `/opt/asterisk-1.2.0/apps' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/apps' make[1]: Entering directory `/opt/asterisk-1.2.0/codecs' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/codecs' make[1]: Entering directory `/opt/asterisk-1.2.0/formats' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/formats' make[1]: Entering directory `/opt/asterisk-1.2.0/agi' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/agi' make[1]: Entering directory `/opt/asterisk-1.2.0/cdr' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/cdr' make[1]: Entering directory `/opt/asterisk-1.2.0/funcs' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/funcs' make[1]: Entering directory
Re: [Asterisk-Users] unable to make calls out using [EMAIL PROTECTED]
type in console: sip show registry and verify status of your trunk 2005/12/1, ram [EMAIL PROTECTED]: yes i got new account from provider and i have registered no one using, iam using that account for testing ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: Have u availability tranks ? 2005/12/1, ram [EMAIL PROTECTED] : Hi as suggested in the group I have downloaded the [EMAIL PROTECTED] installed one of my PC for testing and made 2 extentions to test iam able to talk each other now i have setup one Trunk and made Out going when ever i call to out side i get a voice tone saying that all trunks are busy how can i resolve this problem ram ___ --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 -- Giovanni Miano ___ --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 -- Giovanni Miano ___ --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] cannot dial on console on asterisk 1.2
HI: I ve downloded asterisk 1.2 and when i tried to dial on console this message appears: No such command 'dial' (type 'help' for help) Despite iam not running any audio softwares. __ 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] unable to make calls out using [EMAIL PROTECTED]
Hi here is the results asterisk1*CLI sip show registryHost Username Refresh Statex.x.x..2:5060 xx 105 Registered i have edited the orginals ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: type in console: sip show registryand verify status of your trunk2005/12/1, ram [EMAIL PROTECTED]: yes i got new account from provider and i have registered no one using, iam using that account for testing ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: Have u availability tranks ? 2005/12/1, ram [EMAIL PROTECTED] : Hi as suggested in the group I have downloaded the [EMAIL PROTECTED] installed one of my PC for testing and made 2 extentions to test iam able to talk each other now i have setup one Trunk and made Out going when ever i call to out side i get a voice tone saying that all trunks are busy how can i resolve this problem ram ___ --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 -- Giovanni Miano ___ --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 --Giovanni Miano___--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] unable to make calls out using [EMAIL PROTECTED]
pastme context for outgoing 2005/12/1, ram [EMAIL PROTECTED]: Hi here is the results asterisk1*CLI sip show registry HostUsername Refresh State x.x.x..2:5060 xx 105 Registered i have edited the orginals ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: type in console: sip show registry and verify status of your trunk 2005/12/1, ram [EMAIL PROTECTED]: yes i got new account from provider and i have registered no one using, iam using that account for testing ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: Have u availability tranks ? 2005/12/1, ram [EMAIL PROTECTED] : Hi as suggested in the group I have downloaded the [EMAIL PROTECTED] installed one of my PC for testing and made 2 extentions to test iam able to talk each other now i have setup one Trunk and made Out going when ever i call to out side i get a voice tone saying that all trunks are busy how can i resolve this problem ram ___ --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 -- Giovanni Miano ___ --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 -- Giovanni Miano ___ --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 -- Giovanni Miano ___ --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] 1.2.0 PRI dropping calls occasionally...
On 12/1/05, Rob Thomas [EMAIL PROTECTED] wrote: After upgrading to 1.2.0 (from a three-week-prior CVS version), I've suddenly had people starting to complain of lost calls. They'd be there, and suddenly they'd drop out - they could be in a conversation, or more often, the caller would be on old, and suddenly the light would go out, and the caller would be gone. I haven't noticed anything unusual in the logs, nothing that stands out anyway. Is anyone else experiencing anything like this? This is probably completely unrelated, particularly as it occurs on a 1.0.9 system, but we have one box which occasionally reports a PRI D-Channel up message, resulting in killed calls, even though the D-Channel is already up and working. In our case it was fax-tone detection (for echo canceller switch-off) in zaptel which seems to cause it. We have compiled this feature out on this one server as a workaround. This can be spotted by looking for the zaptel message in dmesg showing tone detection. Strangely, the EXACT same software build and O/S with the same PRI hardware and driver does not cause this problem on any of our other servers. Regards, Steve ___ --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] unable to make calls out using [EMAIL PROTECTED]
hi iam not sure is the right answer iam posting let me know is this correct or not Sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060)bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)disallow=allallow=ulawallow=alawcontext = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown sip_additional.conf register=myaccount:[EMAIL PROTECTED] [tel]username=myaccounttype=peersecret=mysecrethost=sipprovider IP ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: pastme context for outgoing2005/12/1, ram [EMAIL PROTECTED] : Hi here is the results asterisk1*CLI sip show registry HostUsername Refresh State x.x.x..2:5060 xx 105 Registered i have edited the orginals ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: type in console: sip show registry and verify status of your trunk 2005/12/1, ram [EMAIL PROTECTED]: yes i got new account from provider and i have registered no one using, iam using that account for testing ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: Have u availability tranks ? 2005/12/1, ram [EMAIL PROTECTED] : Hi as suggested in the group I have downloaded the [EMAIL PROTECTED] installed one of my PC for testing and made 2 extentions to test iam able to talk each other now i have setup one Trunk and made Out going when ever i call to out side i get a voice tone saying that all trunks are busy how can i resolve this problem ram ___ --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 --Giovanni Miano ___--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 -- Giovanni Miano ___ --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--Giovanni Miano___--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] unable to make calls out using [EMAIL PROTECTED]
Context in extensions.conf 2005/12/1, ram [EMAIL PROTECTED]: hi iam not sure is the right answer iam posting let me know is this correct or not Sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown sip_additional.conf register=myaccount:[EMAIL PROTECTED] [tel] username=myaccount type=peer secret=mysecret host=sipprovider IP ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: pastme context for outgoing 2005/12/1, ram [EMAIL PROTECTED] : Hi here is the results asterisk1*CLI sip show registry HostUsername Refresh State x.x.x..2:5060 xx 105 Registered i have edited the orginals ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: type in console: sip show registry and verify status of your trunk 2005/12/1, ram [EMAIL PROTECTED]: yes i got new account from provider and i have registered no one using, iam using that account for testing ram On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote: Have u availability tranks ? 2005/12/1, ram [EMAIL PROTECTED] : Hi as suggested in the group I have downloaded the [EMAIL PROTECTED] installed one of my PC for testing and made 2 extentions to test iam able to talk each other now i have setup one Trunk and made Out going when ever i call to out side i get a voice tone saying that all trunks are busy how can i resolve this problem ram ___ --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 -- Giovanni Miano ___ --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 -- Giovanni Miano ___ --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 -- Giovanni Miano ___ --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 -- Giovanni Miano ___ --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 Problem
Hi all, I was trying to use G.723.1 codec for my terminator as Pass through. But when the second party pickup phone the call is going dropted automatically with the following error: No path to translate from SIP/123456-fca7(1) to SIP/myterminator.com-ff11(4) Dec 1 10:54:39 WARNING[7480]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't make SIP/123456-fca7 compatible with SIP/myterminator.com-ff11 Please advice me how i can make it work? -- Best Regards, Code Lover Nepal ___ --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: RE : [Asterisk-Users] Asterisk doesn't start
On 11/25/05 18:32 Olivier Taylor said the following: Yes, beta2 works perfectly, but 1.2 released version gives me this error. looks like you did not clean out your modules directory when you installed 1.2 over 1.2 beta. try doing that and reinstalling. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Compiling Asterisk 1.2 from Source on
Hi Thanks for the replies. Its fixed now. The problem I had was that I had a mixed Debian Stable and Unstable system. Some of the libraries I downloaded from Unstable caused breakages. Basically, long story short, I re-installed Debian, got all the required packages from Stable and installed Zaptel and Asterisk without a problem. Hi! I just built Asterisk on Debian Sarge myself and it worked without any problems. Can you cut 'n paste the error messages? I can't make any sense from the output ... Greetings, Marcus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Hagen Rode Sent: Wednesday, November 30, 2005 6:38 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Compiling Asterisk 1.2 from Source on Debian Sarge- Problems Hi I am trying to compile Asterisk 1.2 from source on Debian Sarge but am getting errors. I have looked at the errors, Googled extensively and now at a last resort am posting on this list. Believe me I have tried, but have come up with nothing. I've also installed the following packages from Debian Sarge UNSTABLE: gcc kernel-headers-2.4.27 bison openssl libssl0.9.7: libssl-dev libeditline0 libeditline-dev libedit-dev libedit2 libncurses5 libncurses5-dev zlib1g-dev (Note: needed for cvs head) as well as numerous other packages that I have now lost track of. The error remains the same. It would be great if someone could help me out. I'm aware that I can apt-get Asterisk, but I want to do some tweaking in the code before installing. Here is the first bit of the install message: build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp if cmp -s .cleancount .lastclean ; then echo ; else \ make clean; cp -f .cleancount .lastclean;\ fi build_tools/make_defaults_h defaults.h.tmp if cmp -s defaults.h.tmp defaults.h ; then echo ; else \ mv defaults.h.tmp defaults.h ; \ fi ___ --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: pbx or asterisk?
On Wed, Nov 30, 2005 at 01:17:33PM -0500, Sean Cook wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Looks like your zap channels are droping into the default context... better to set up a from-pstn context and start there. hi sean you have a example please? Pablo Allietti wrote: hi all i have a pbx siemens connect via E1 to my asterisk box. the asterisk box can call without problems to pbx extensions. but when y press the numbers form example 402 in the pbx phones asterisk give me this -- Saved useragent X-Lite release 1103m for peer 402 -- Going to extension s|1 because of Complete received -- Executing Playback(Zap/31-1, vm-goodbye) in new stack -- Accepting call from '' to 's' on channel 0/31, span 1 == Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1' -- Hungup 'Zap/31-1' -- Accepting call from '' to 's' on channel 0/31, span 1did not receive any number or i have miss configure somenthing in asterisk box? -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDjey8y9wPyZpnL2URAiVCAJ4hQCz+eb1/MaABy2gxUMOcMw1AMwCfYEJI VTt9lDiRDMLZhJ2aOL4Qpnw= =KqmL -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 ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --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] Re: pbx or asterisk?
2005/12/1, Pablo Allietti [EMAIL PROTECTED]: hi sean you have a example please? In your zapata.conf ensure there is context=from-pstn like this: [channels] context=from-pstn -- Alejandro Vargas ___ --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] MGCP problem when through internet
I'm using mediatrix mgcp device without problems with [EMAIL PROTECTED] 2.0 over the LAN. But now I trying one of this devices through internet. My firs problem was nat, but I decided to leave this problem for later and try it through a vpn. I used gvpe because it is very transparent. The device connects ok but in many cases (I suspect when the bandwidth is low) the call is droped after a few seconds with a message like this: No command found on [192.168.0.105] for transaction 404. Ignoring.. The number (676 in this case) is variable. Isee with etereal that Asterisk is sending a message of RQNT 404 aaln/[EMAIL PROTECTED] MGCP 1.0 and the device responds 200 404 OK, but following this, the asterisk server starts rejecting the packets of the port it were using to communicate to the device. -- Alejandro Vargas ___ --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] two sip phone communication using asterisk server
2005/12/1, Tejas Shah [EMAIL PROTECTED]: I am a newbie to asterisk. I installed a asterisk server to make communication between 2 X-Lite's SIP based phones. I made following configuration in sip.conf : For newbies (like me) a good start is to use amp or install directly asteriskathome. It solves all the problems of configuring and creating extensions. Then you can start lerning how to do the difficult things. -- Alejandro Vargas ___ --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: Newbie question
Thanks Mr.Miano Thanks a lot. Now I think I wont have to bother about balming all my problems to zapata. I have also succeeded quite a bit and installed a basic PBX system without it. Thanks a lot again. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. Giovanni Miano wrote: I dont need to configure zaptel device, you dont use it :) 2005/11/30, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hello friends, I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My question is I am using a Welltech FXO box and ip phones by Welltech. Do I still need to configure zapata.conf and zaptel.conf which I read in the documentation from asterisk pdf file downoladed from asterisk.org ? I think I dont because I dont use a digium card but do I have to still confugure for FXO and FXS ports? Kindly help me solving my doubt. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends. ___ --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 -- Giovanni Miano ___ --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] Codec Problem
Please advice me how i can make it work? It looks like your Phone is not compatible to G.723.1 or this codec is disabled within sip.conf Elmar ___ --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] cannot dial on console on asterisk 1.2
I believe you have to have a sound card. -Original Message- From: jonny hashem [mailto:[EMAIL PROTECTED] Sent: Thursday, December 01, 2005 5:16 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cannot dial on console on asterisk 1.2 HI: I ve downloded asterisk 1.2 and when i tried to dial on console this message appears: No such command 'dial' (type 'help' for help) Despite iam not running any audio softwares. ___ --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: MGCP problem when through internet
Is there any way to tell asterisk that ignore protocol errors instead of dropping the call? -- Alejandro Vargas ___ --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] mail2fax and fax2mail
I am looking about those two script because I am not able to find them on www.generationd.com May some one help me please? Thanks 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
[Asterisk-Users] Re: US e911 reminder
How are you updating the E911 address information? We have literally been pulling teeth at Verizon to get access to their PS/ALI database to make the updates that we need to. On 11/30/05, Jan Saell [EMAIL PROTECTED] wrote: Just a small note that we have used a cluster of asterisk to connect or Voip systems to the HFB E911 service and it worked without any problems. SO one can defenetly use asterisk in one of these environments. Best regards jan --On Monday, November 28, 2005 03:01:25 PM -0800 trixter aka Bret McDanel [EMAIL PROTECTED] wrote: Just a reminder tonight at midnight is the deadline for pstn connected VoIP providers operating in the US to provide E911 or face fines upto $11,000 per day. There is also a filing requirement with the FCC which is due tonight as well. Enforcement Bureau Outlines Requirements of November 28, 2005 Interconnected Voice Over Internet Protocol 911 Compliance Letters http://www.fcc.gov/eb/Public_Notices/DA-05-2945A1.html Consumer page but has some basic info http://ftp.fcc.gov/cgb/consumerfacts/voip911.html -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B ___ --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] Error on using queue.
I am trying to use * as ACD server for our sip
proxy.
I first dial 55 to login 98 as ACD
agent it worked fine and then when I dialed 98, I got these messages from * CLI:
-- Executing
Answer("SIP/98-f718", "") in new stack --
Executing Ringing("SIP/98-f718", "") in new stack
-- Executing Wait("SIP/98-f718", "2") in new stack
-- Executing Queue("SIP/98-f718", "queue1") in new stackNov 30
16:54:12 WARNING[7579]: app_queue.c:3078 queue_exec: Unable to join queue
'queue1' -- Executing Hangup("SIP/98-f718", "") in
new stack == Spawn extension (default, 99, 5) exited non-zero
on 'SIP/5025155598-f718'
Can anybody tell me what cause this
problem?
The followings are my configuration
files:
extensions.conf:
[default]
;For incoming call to ring into the
queue.exten= 99,1,Answerexten=
99,2,Ringingexten= 99,3,Wait(2)exten=
99,4,Queue(queue1)exten= 99,5,Hangup
;Agent loginexten =
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent
logoutexten = 55,1,AgentCallBackLogin(|1)
exten =
97,1,Dial(SIP/97)exten =
98,1,Dial(SIP/98)
agents.conf:
[Agent1]agent =
97,,Gary1agent = 98,,Gary2
queues.conf:
[queue1]musiconhold =
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen =
0announce-frequency = 0announce-holdtime = nomember =
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw
[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
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RE: [Asterisk-Users] Problems with auto dialout
Hi Tim
Thanks for the info.
I see what your example is doing.
However what if I want Asterisk to call someone
that isnt on the local network?
So if someone is out and about they can be
called on a mobile to let them know something is down?
Tony
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tim panton
Sent: 29 November 2005 18:37
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Problems with auto dialout
Channel: Local/[EMAIL PROTECTED]
Callerid: 01612370660
MaxRetries: 5
RetryTime: 300
WaitTime: 45
Context: serverdown
Extension: s
Priority: 1
On 29 Nov 2005, at 15:39, Tony Spencer wrote:
I'm a bit of newbie to Asterisk so I'm not to sure.
I was just given the task to try and make this work.
You could be correct but I'd have to do some further investigation and
speak
to the person that used to admin this server.
All I want to do is call a phone number and play a audio file and
hangup.
Is there a way of doing this by dropping a file in the outgoing queue
or
even from a script/commandline..
Thanks
Tony
I have a simple system like this, the call file looks like:
Channel: Local/[EMAIL PROTECTED]
Callerid: 01612370660
MaxRetries: 5
RetryTime: 300
WaitTime: 45
Context: serverdown
Extension: s
Priority: 1
SetVar: SITENAME=importantCustomerName
And the following in extensions.conf:
[serverdown]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Playback(serverdown/${SITENAME})
exten = s,4,Wait(10)
exten = s,5,Playback(serverdown/${SITENAME})
exten = s,6,Hangup
I have a file pre-recorded with a customer specific message in
serverdown/importantCustomerName.gsm
The trick with Local/[EMAIL PROTECTED] is to distribute the call to multiple
users:
[default]
exten =
60,1,Dial(Sip/billSip/benSip/flowerSip/potSip/weed,30)
Good luck,
Tim.
http://www.westhawk.co.uk/
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[Asterisk-Users] WG: App_rxfax problem
nobody has problems like me? Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 1. Dezember 2005 08:35An: 'asterisk-users@lists.digium.com'Betreff: App_rxfax problem When i load the fax modules into the asterisk i got this errors but compile was ok! I have the latest cvs head [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold'[app_rxfax.so]Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Ouch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipe ___ --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] WG: App_rxfax problem
check /var/log/asterisk/full 2005/12/1, René Enskat [Teamware GmbH] [EMAIL PROTECTED]: nobody has problems like me? Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 1. Dezember 2005 08:35 An: 'asterisk-users@lists.digium.com' Betreff: App_rxfax problem When i load the fax modules into the asterisk i got this errors but compile was ok! I have the latest cvs head [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ --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 -- Giovanni Miano ___ --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] prepaid application
Hi All I am using prepaid auth (callingcards), the idea is for a prepaid support line. It is up and running but I have a couple of questions with regards to modifications I would like to make. When a user calls and they go through the process of entering their card number. They are then asked for a destination. What I would like to be able to do is not have it ask for a destination and automatically dial a number? At present I ask them to enter a default number when it ask for a destination and this then takes them to a queue, if someone is available it rings and goes through, if no one is available rather than sit in the queue and listen to the lovely onhold music prepaid auth comes back and says that destination is unreachable, is there a way to get it to just wait in the queue? Many Thanks In Advance Scott Pinhorne ___ --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
AW: [Asterisk-Users] WG: App_rxfax problem
Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]Dec 1 15:01:08 WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so failed! -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Giovanni Miano Gesendet: Donnerstag, 1. Dezember 2005 14:49 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] WG: App_rxfax problem check /var/log/asterisk/full 2005/12/1, René Enskat [Teamware GmbH] [EMAIL PROTECTED]: nobody has problems like me? Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 1. Dezember 2005 08:35 An: 'asterisk-users@lists.digium.com' Betreff: App_rxfax problem When i load the fax modules into the asterisk i got this errors but compile was ok! I have the latest cvs head [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ --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 -- Giovanni Miano ___ --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] Call transfer error
When I arrived a call, I would the call transfer in to another telephone number, but Asterisk show error: Executing GotoIfTime("Zap/4-1", "08:30-12:30|mon-fri|*|*?4") in new stack -- Executing GotoIfTime("Zap/4-1", "15:30-18:30|mon-fri|*|*?4") in new stack -- Executing Goto("Zap/4-1", "6") in new stack -- Goto (isdn_incoming,0445363378,6) -- Executing Dial("Zap/4-1", "ZAP/g2/0445384225|60") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/0445384225 -- Zap/5-1 is proceeding passing it to Zap/4-1 -- Channel 0/2, span 2 got hangup request -- Hungup 'Zap/5-1' == No one is available to answer at this time (1:0/0/0) -- Executing Hangup("Zap/4-1", "") in new stack == Spawn exten
sion
(isdn_incoming, 0445363378, 7) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' MY ZAPATA.CONF IS: [channels] switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel = yes context=isdn_incoming group = 1 channel = 1-2 group = 2 channel = 4-5 group = 3 channel = 7-8 group = 4 channel = 10-11 MY ZAPTEL.CONF is loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 MY EXTENSIONS.CONF is [isdn_incoming] exten = 0445363378,1,GotoIfTime(${ORAMATTINO}?4) exten = 0445363378,2,GotoIfTime(${ORAPOMERIGGIO}?4) exten = 0445363378,3,Goto(6) exten =
0445363378,4,Dial(${TELEIN},60) exten = 0445363378,5,Hangup exten = 0445363378,6,Dial(ZAP/g2/0445384225,60) exten = 0445363378,7,Hangup What can I doing? Thanks
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Re: [Asterisk-Users] WG: App_rxfax problem
nobody has problems like me? --- == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ---End of Original Message- If you are talking about the Ouch message, yes lots of people have seen the error and its usually the result of some misconfiguration in one of your files (likely zapata.conf). Since you didn't provide anything reasonable for anyone to look at or comment on, its impossible to guess at what you might have done. The message would suggest that musiconhold probably has something to do with the problem because of the flexibel rate 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
Re: [Asterisk-Users] Asterisk cluster and astdb
Matt Riddell wrote: dashy dude wrote: Dear All I am trying to build a high availability cluster of asterisk. I am using RedHat cluster management suit on Enterprise edition AS3 Origianally, astdb was located on native hard disk of each server. All my end points are configured for Reinvite=Yes Everrything was working fine and if active server is rebooted, the standby would take over and the ongoing calls will continue without any problem. But this had a problem that the astdb file is not updated with latest end-point information and phones dont get a call untill they re register. To avoid this, I moved the astdb file on the shared storage and created sym links from individual servers. Now, when the active server is rebooted, all the active calls are dropped. Please help me in resolving this. Why don't you use Asterisk RealTime? correct if I'm wrong (frequently) but the call state isn't stored in the realtime db is it? linux-ha uses a form of shared disk called DRBD that might solve this if you forced the astdb onto that. Only one node of the cluster is currently allowed to write to astdb on that though. Just a thought ___ --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 question
From: Jan Saell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 30, 2005 9:32 AM Subject: Re: [Asterisk-Users] Blind transfer question I did a quick check on the blindxfer config parameter and i cant find any referense to that in the sourcecode for 1.2! The features are defined in ... tada... res/res_features.c ! :-) I've found features are detected most reliably when the phone sends DTMF as sip-events, not via RTP (RFC2833) or in-audio Leif ___ --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
AW: [Asterisk-Users] WG: App_rxfax problem
I just sent the error in full log: Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]Dec 1 15:01:08 WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so failed! -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Rich Adamson Gesendet: Donnerstag, 1. Dezember 2005 15:05 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] WG: App_rxfax problem nobody has problems like me? --- == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ---End of Original Message- If you are talking about the Ouch message, yes lots of people have seen the error and its usually the result of some misconfiguration in one of your files (likely zapata.conf). Since you didn't provide anything reasonable for anyone to look at or comment on, its impossible to guess at what you might have done. The message would suggest that musiconhold probably has something to do with the problem because of the flexibel rate 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 ___ --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] Complete Removal of Asterisk
Is there any documentation for the complete removal of Asterisk from a Linux/Unix system? I want to install a fresh copy of asterisk. Thanks, Chip ___ --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] show queue in BE
Hi,
I am using Asterisk Business Edition A.1.6 (but I
guess it is the same logic for 1.2)
I am running the show queue
command for a queue that had a 36 calls and the C: parameter is growing up very
fastly, no reflecting the real calls to this queue.
lv09*CLI show queue cobranca
cobranca has 0 calls (max unlimited) in 'leastrecent' strategy (32s
holdtime), W:0, C:1006994, A:16, SL:0.0% within 45s
Here is my queues.conf
[cobranca]musiconhold=filajoinempty=yesstrategy=leastrecenteventwhencalled=yestimeout=14maxlen=0retry=0servicelevel=45wrapuptime=5announce-holdtime=nomember
= Agent/5120member = Agent/5130member = Agent/5410member
= Agent/5100member = Agent/2110member = Agent/5420
My agents.conf is
[agents]autologoff=150ackcall=nowrapuptime=5000musiconhold
=
filaupdatecdr=yesrecordagentcalls=norecordformat=wav49savecallsin=/home/asterisk/spool/monitorgroup
= 1 ; fila cobrancaagent = 5120,1234,Alessandra Barrosagent =
5130,1234,Ana Paula Furuyaagent = 5410,1234,Ana Silvaagent =
5100,1234,Bruno Tolentino Alvesagent = 2110,1234,Debora
Goncalvesagent = 5420,1234,Fabiana Montera
My extensions.conf for entering the queue:
exten = cobranca,1,NoOp(Ligacao para Fila de Cobranca)exten =
cobranca,2,SetVar(prioridade=0)exten = cobranca,3,SetCIDName(Cobranca
${CALLERIDNAME})exten =
cobranca,4,SetVar(QUEUE_PRIO=${prioridade})exten =
cobranca,5,Answerexten = cobranca,6,Queue(cobranca|tT|||50)exten
= cobranca,7,Hangup
This is very important since is it is preventing my Call Center Monitoring
application to work (it worked well while running 1.0.9 open source).
Does the C: value mean a different thing? Or is there any configuration that
I am missing somewhere?
Thank you very much
Dov
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[Asterisk-Users] Better transfer
I find the transfer functions a little lacking. Examples: I get a call I do an attended transfer, but the called extension never answers/I get impatient/I discover I have dialed the wrong extension. I can not get the call back. If I hangup, the caller is also hung up. I'd prefer the caller to stay online and be ringing my phone again. If I do an attended transfer, and hangup before the 3. part answers, the caller is disconnected. I'd prefer the transfer to be turned into a blind transfer, the caller coming back to me if the called ext is not answering If I do a blind transfer, and the called ext is not answereing, I'd like the call to come back to me. Can this be done in dialplan, or must it be changed in the source? Leif ___ --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] Complete Removal of Asterisk
your distro shoudl supply uninstaller, something like emerge unmerge asterisk in gentoo. Tough that would not remove your configuration files. Best RegardsOn 12/1/05, cp [EMAIL PROTECTED] wrote: Is there any documentation for the complete removal of Asterisk from a Linux/Unix system? I want to install a fresh copy of asterisk. Thanks, Chip ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --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] Better transfer
On Thu, 2005-12-01 at 15:50 +0100, Leif Neland wrote: I find the transfer functions a little lacking. Examples: I get a call I do an attended transfer, but the called extension never answers/I get impatient/I discover I have dialed the wrong extension. I can not get the call back. Iirc in 1.2 you can get the call back with #0. see features.conf Regards, Patrick ___ --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] Problems with auto dialout
On 1 Dec 2005, at 13:33, Tony Spencer wrote: Hi Tim Thanks for the info.I see what your example is doing.However what if I want Asterisk to call someone that isn’t on the local network?So if someone is out and about they can be called on a mobile to let them know something is down?Just put a suitable set of commands in your Dial string in extensions.confSay: Dial(Sip/workZap/g1/01612370660Zap/g1/07900,30)Which dials the local Sip, the phone PSTN number and a mobile, whoever answers first gets the call. (rings for up to 60 secs). The only problem is if the mobile is offand goes straight to answerphone, that will always answer first.Personally for mobiles I prefer to use sms for notification and voice for office/home. Tony From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of tim panton Sent: 29 November 2005 18:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems with auto dialout Channel: Local/[EMAIL PROTECTED] Callerid: 01612370660 MaxRetries: 5 RetryTime: 300 WaitTime: 45 Context: serverdown Extension: s Priority: 1 On 29 Nov 2005, at 15:39, Tony Spencer wrote:I'm a bit of newbie to Asterisk so I'm not to sure. I was just given the task to try and make this work. You could be correct but I'd have to do some further investigation and speak to the person that used to admin this server. All I want to do is call a phone number and play a audio file and hangup. Is there a way of doing this by dropping a file in the outgoing queue or even from a script/commandline.. Thanks Tony I have a simple system like this, the call file looks like: Channel: Local/[EMAIL PROTECTED] Callerid: 01612370660 MaxRetries: 5 RetryTime: 300 WaitTime: 45 Context: serverdown Extension: s Priority: 1 SetVar: SITENAME=importantCustomerName And the following in extensions.conf: [serverdown] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Playback(serverdown/${SITENAME}) exten = s,4,Wait(10) exten = s,5,Playback(serverdown/${SITENAME}) exten = s,6,Hangup I have a file pre-recorded with a customer specific message in serverdown/importantCustomerName.gsm The trick with Local/[EMAIL PROTECTED] is to distribute the call to multiple users: [default] exten = 60,1,Dial(Sip/billSip/benSip/flowerSip/potSip/weed,30) Good luck, Tim. http://www.westhawk.co.uk/ -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.10/189 - Release Date: 30/11/2005-- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.10/189 - Release Date: 30/11/2005 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.westhawk.co.uk/ ___
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Re: [Asterisk-Users] Complete Removal of Asterisk
Hi, Is there any documentation for the complete removal of Asterisk from a Linux/Unix system? I want to install a fresh copy of asterisk. Depend of your distro, You can use emerge for Gentoo or rpm with CentOS/RHEL/Fedora. Debian also have dpkg command. If you have installed from source, you can do a : rm -rf /usr/lib/asterisk rm -rf /var/lib/asterisk rm -rf /usr/sbin/asterisk If you want to remove configs files: rm -rf /etc/asterisk If you want to delete Voicemail and Outgoing Call queues: rm -rf /var/spool/asterisk Thanks, -- Joel Vandal ScopServ Inc. http://www.scopserv.com/ Complete Web GUI for Asterisk PBX ___ --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] voipbuster
I was testing voipbuster. With a new account, with no credit, I can make calls perfectly but of 1 minute. But I tried the username and passwrord of an account with credit, and the registration is refused. With the voipbuster propietary software it works ok (I sniffed the packets and I think it is not using standard iax or sip ports). Are the acconts with credit blocked for avoiding it's use with ohter software than voipbuster's? I tryed to send a mail to voipbuster's support but I never received an answer (then do not support other thing than their software). -- Alejandro Vargas ___ --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 Realtime 2 Servers calling each other
Hi guys I have a question, im trying asterisk realtime in 2 servers. Im trying to make calls from one server to another, example I call a sip registered in sip server 1 with a phone register in sip server2 and both using the same database and family both use canreinvite=yes but still cant make the calls any ideas? ___ --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] Better transfer
- Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 01, 2005 4:00 PM Subject: Re: [Asterisk-Users] Better transfer On Thu, 2005-12-01 at 15:50 +0100, Leif Neland wrote: I find the transfer functions a little lacking. Examples: I get a call I do an attended transfer, but the called extension never answers/I get impatient/I discover I have dialed the wrong extension. I can not get the call back. Iirc in 1.2 you can get the call back with #0. see features.conf My features.conf.sample doesn't have #0: [featuremap] ;blindxfer = #1; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record ;atxfer = *2 ; Attended transfer Neither can I see any hints in res/res_features.c Unless disconnect above really means abort transfer Leif ___ --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] Error on using queue.
If you are using 1.2, it might be the joinempty and
leavewhenempty parameters.
Their default are different than the 1.0.x
releases
- Original Message -
From:
gc
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 11:27
AM
Subject: [Asterisk-Users] Error on using
queue.
I am trying to use * as ACD server for our sip
proxy.
I first dial 55 to login 98 as
ACD agent it worked fine and then when I dialed 98, I got these messages from * CLI:
-- Executing
Answer("SIP/98-f718", "") in new stack --
Executing Ringing("SIP/98-f718", "") in new
stack -- Executing Wait("SIP/98-f718", "2") in
new stack -- Executing Queue("SIP/98-f718",
"queue1") in new stackNov 30 16:54:12 WARNING[7579]: app_queue.c:3078
queue_exec: Unable to join queue 'queue1' -- Executing
Hangup("SIP/98-f718", "") in new stack == Spawn extension
(default, 99, 5) exited non-zero on
'SIP/5025155598-f718'
Can anybody tell me what cause this
problem?
The followings are my configuration
files:
extensions.conf:
[default]
;For incoming call to ring into the
queue.exten= 99,1,Answerexten=
99,2,Ringingexten= 99,3,Wait(2)exten=
99,4,Queue(queue1)exten= 99,5,Hangup
;Agent loginexten =
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent
logoutexten = 55,1,AgentCallBackLogin(|1)
exten =
97,1,Dial(SIP/97)exten =
98,1,Dial(SIP/98)
agents.conf:
[Agent1]agent =
97,,Gary1agent = 98,,Gary2
queues.conf:
[queue1]musiconhold =
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen
= 0announce-frequency = 0announce-holdtime = nomember =
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw
[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
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Re: [Asterisk-Users] Queue calls...
Hi Trey, It is done automagically by the system - see the setting named announce in the queue definition. Hope this helps l. On Wed, 30 Nov 2005 20:38:20 +0100, Trey Blancher [EMAIL PROTECTED] wrote: I want to play a file for an agent that answers a queue call, before the agent is actually connected with the call. I want something along the lines of,Answer as member of team X, or similar, before the agent is connected with the caller. Is this possible? And how would I do it? -- Trey Blancher Systems Administrator, USA Debt Management LLC (251)445-0683 ext 8601 ___ --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 -- Loway Research - Home of QueueMetrics 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] Better transfer
On Thu, 2005-12-01 at 16:30 +0100, Leif Neland wrote: [snip] Unless disconnect above really means abort transfer Yup and you could have found that out easily by trying it :) Regards, Patrick ___ --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] meet me message
Since upgrade to BE A.1-6I get the following messages on my console... -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-2-3 format: sln, 0x9e454b8 And several .sln files are saved on /var/spool/asterisk/meetme/ What do this mean? Thank you Dov ___ --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] Complete Removal of Asterisk
From the asterisk source directory a make uninstall should also do it. On 12/1/05, Joel Vandal [EMAIL PROTECTED] wrote: Hi, Is there any documentation for the complete removal of Asterisk from a Linux/Unix system? I want to install a fresh copy of asterisk. Depend of your distro, You can use emerge for Gentoo or rpm with CentOS/RHEL/Fedora. Debian also have dpkg command. If you have installed from source, you can do a : rm -rf /usr/lib/asterisk rm -rf /var/lib/asterisk rm -rf /usr/sbin/asterisk If you want to remove configs files: rm -rf /etc/asterisk If you want to delete Voicemail and Outgoing Call queues: rm -rf /var/spool/asterisk Thanks, -- Joel Vandal ScopServ Inc. http://www.scopserv.com/ Complete Web GUI for Asterisk PBX ___ --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] voipbuster
I ended up buying a second 1 euro account because of this. But it does work fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Thursday, December 01, 2005 7:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voipbuster I was testing voipbuster. With a new account, with no credit, I can make calls perfectly but of 1 minute. But I tried the username and passwrord of an account with credit, and the registration is refused. With the voipbuster propietary software it works ok (I sniffed the packets and I think it is not using standard iax or sip ports). Are the acconts with credit blocked for avoiding it's use with ohter software than voipbuster's? I tryed to send a mail to voipbuster's support but I never received an answer (then do not support other thing than their software). -- Alejandro Vargas ___ --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 and Broadvoice, no incoming voice
My incoming BV has been intermittant for the last two days as well. It has gone down somewhere around 4:30 PM Eastern two days in a row, then been back up in the morning. In the 10:00 AM hour today, it was down for about ten minutes. Jason Schafer writes: I have been trying on and off for a couple of weeks to no avail... Darren Wright wrote: I am also a long time client, and have no incoming BV today. -Darren http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Janina SajkaPhone: +1.240.715.1272 Partner, Capital Accessibility LLC http://www.CapitalAccessibility.Com Marketing the Owasys 22C talking screenless cell phone in the U.S. and Canada--Go to http://www.ScreenlessPhone.Com to learn more. Chair, Accessibility Workgroup Free Standards Group (FSG) [EMAIL PROTECTED] http://a11y.org ___ --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] voipbuster
Alejandro Vargas wrote: But I tried the username and passwrord of an account with credit, and the registration is refused. With the voipbuster propietary software it works ok (I sniffed the packets and I think it is not using standard iax or sip ports). Are the acconts with credit blocked for avoiding it's use with ohter software than voipbuster's? The following works in iax.conf for me: [voipbuster] host=iax.voipbuster.com type=peer username=username secret=password qualify=yes context=inbound Also the same company (finarea) runs sipdiscount (which are currently slightly better rates than voipbuster, but it varies a lot so you have to keep on top of it.. they'll probably start a new company with different rates again next week!). btw. does anyone have a definitive list of all the finarea VOIP companies? I can think of: call1899 call18866 voipbuster sipdiscount voipcheap (note: this one uses a proprietary protocol, similar to IAX but over different ports and not compatibile). Tony ___ --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] Re: pbx or asterisk?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Pablo Allietti wrote:
On Wed, Nov 30, 2005 at 01:17:33PM -0500, Sean Cook wrote:
Looks like your zap channels are droping into the default context...
better to set up a from-pstn context and start there.
hi sean you have a example please?
sure... this will at least get you going in the right direction... the
context will dump all variables into debug (noop) so you can find out
how the system is passing DID into the system.
in zapata.conf
[channels]
context=from-pstn
in extensions.conf
[from-pstn]
exten = s,1,Noop(ACCOUNTCODE=${ACCOUNTCODE})
exten = s,2,Noop(ANSWEREDTIME=${ANSWEREDTIME})
exten = s,3,Noop(BLINDTRANSFER=${BLINDTRANSFER})
exten = s,4,Noop(CALLERID=${CALLERID})
exten = s,5,Noop(CALLERIDNAME=${CALLERIDNAME})
exten = s,6,Noop(CALLERIDNUM=${CALLERIDNUM})
exten = s,7,Noop(CALLINGPRES=${CALLINGPRES})
exten = s,8,Noop(CHANNEL=${CHANNEL})
exten = s,9,Noop(CONTEXT=${CONTEXT})
exten = s,10,Noop(DATETIME=${DATETIME})
exten = s,11,Noop(DIALEDPEERNAME=${DIALEDPEERNAME})
exten = s,12,Noop(DIALEDPEERNUMBER=${DIALEDPEERNUMBER})
exten = s,13,Noop(DIALEDTIME=${DIALEDTIME})
exten = s,14,Noop(DIALSTATUS=${DIALSTATUS})
exten = s,15,Noop(DNID=${DNID})
exten = s,16,Noop(EPOCH=${EPOCH})
exten = s,17,Noop(EXTEN=${EXTEN})
exten = s,18,Noop(HANGUPCAUSE=${HANGUPCAUSE})
exten = s,19,Noop(INVALID_EXTEN=${INVALID_EXTEN})
exten = s,20,Noop(LANGUAGE=${LANGUAGE})
exten = s,21,Noop(MEETMESECS=${MEETMESECS})
exten = s,22,Noop(PRIORITY=${PRIORITY})
exten = s,23,Noop(RDNIS=${RDNIS})
exten = s,24,Noop(SIPDOMAIN=${SIPDOMAIN})
exten = s,25,Noop(SIP_CODEC=${SIP_CODEC})
exten = s,26,Noop(SIPCALLID=${SIPCALLID})
exten = s,27,Noop(SIPUSERAGENT=${SIPUSERAGENT})
exten = s,28,Noop(TIMESTAMP=${TIMESTAMP})
exten = s,29,Noop(TXTCIDNAME=${TXTCIDNAME})
exten = s,30,Noop(UNIQUEID=${UNIQUEID})
exten = s,31,Noop(TOUCH_MONITOR=${TOUCH_MONITOR})
exten = s,32,Noop(MACRO_CONTEXT=${MACRO_CONTEXT})
exten = s,33,Noop(MACRO_EXTEN=${MACRO_EXTEN})
exten = s,34,Noop(MACRO_PRIORITY=${MACRO_PRIORITY})
Pablo Allietti wrote:
hi all i have a pbx siemens connect via E1 to my asterisk box.
the asterisk box can call without problems to pbx extensions. but when y
press the numbers form example 402 in the pbx phones asterisk give me
this
-- Saved useragent X-Lite release 1103m for peer 402
-- Going to extension s|1 because of Complete received
-- Executing Playback(Zap/31-1, vm-goodbye) in new stack
-- Accepting call from '' to 's' on channel 0/31, span 1
== Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1'
-- Hungup 'Zap/31-1'
-- Accepting call from '' to 's' on channel 0/31, span 1did not
receive any number or i have miss configure somenthing in asterisk box?
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Re: [Asterisk-Users] Queue calls...
announce is exactly what I'm looking for. I had originally thought that meant playback for the caller, not the agent who answers the call. If I had time I'd add that to the wiki, since it needs to be there, and not buried in an example. On 12/1/05, Lenz [EMAIL PROTECTED] wrote: Hi Trey, It is done automagically by the system - see the setting named announce in the queue definition. Hope this helps l. On Wed, 30 Nov 2005 20:38:20 +0100, Trey Blancher [EMAIL PROTECTED] wrote: I want to play a file for an agent that answers a queue call, before the agent is actually connected with the call. I want something along the lines of,Answer as member of team X, or similar, before the agent is connected with the caller. Is this possible? And how would I do it? -- Trey Blancher Systems Administrator, USA Debt Management LLC (251)445-0683 ext 8601 ___ --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 -- Loway Research - Home of QueueMetrics 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 -- Trey Blancher Systems Administrator, USA Debt Management LLC (251)445-0683 ext 8601 ___ --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 Perl AGI, bug with stream_file() ?
Hello, On http://www.voip-info.org/tiki-index.php?page=Asterisk%20perl%20agi, it's said that stream_file() might returns -1 on error or hangup, 0 if playback completes without a digit being pressed, or the ASCII numerical value of the digit if a digit was pressed. But actually when I hangup my phone stream_file returns 0 and the AGI is stopped even if I have a callback function, set with setcallback(), to catch the hangup signal and exit properly the AGI. In the Asterisk logs, instead of having AGI Script myagi.agi completed, returning 0, I have Spawn extension (default, 777, 2) exited non-zero on 'IAX2/7-5' ... (777 is the extension to launch myagi.agi, and 7 is my phone's username/number). Is that a bug? -- Benoit Merouze Ingenieur Developpement d'Application Reseau [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] US e911 reminder
On 11/28/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: Enforcement Bureau Outlines Requirements of November 28, 2005 Interconnected Voice Over Internet Protocol 911 Compliance Letters http://www.fcc.gov/eb/Public_Notices/DA-05-2945A1.html I'm just trying to clarify this, according to note 1 on that paper: 1 ``Interconnected VoIP service'' means an interconnected voice over Internet Protocol service that: (1) enables real-time, two-way voice communications; (2) requires a broadband connection from the user's location; (3) requires Internet protocol-compatible customer premises equipment (CPE); and (4) permits users generally to receive calls that originate on the public switched telephone network and to terminate calls to the public switched telephone network. See 47 C.F.R. � 9.3. = Does that mean: 1. That if I allow only outbound, I'm not required to comply? (number 4 above) 2. That if I setup the customer, as a remote node on my hosted asterisk server for inbound and outbound of just their business phone, but have it setup that 911 when dialed from the phone should use a backup proxy that is local and doesnt require a broadband connection (it's using the LAN), that I'm not required to comply? (number 2 above), the setup in mind is where I use a polycom phone with a Sipura SPA 3000, configured that when 911 is dialed it uses the Sipura and not Asterisk. 3. If the answer to 2 is that yes I have to comply, then why when installing a lagecy pbx (like avaya) that doesn't have battery backup I'm not required to mail stickers? is that because I don't use the name VoIP? and the media hasn't caught up to this one (I'm sure it happended before)? Well thats it for now, I'm sure I have more questions :) ___ --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] US e911 reminder
I guess this one answers some questions, and it also gives me someone bigger than me (for now anyhow :) ) that will fight it for me: https://www.stanaphone.com/index/news_Nov2205.html On 12/1/05, C F [EMAIL PROTECTED] wrote: On 11/28/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: Enforcement Bureau Outlines Requirements of November 28, 2005 Interconnected Voice Over Internet Protocol 911 Compliance Letters http://www.fcc.gov/eb/Public_Notices/DA-05-2945A1.html I'm just trying to clarify this, according to note 1 on that paper: 1 ``Interconnected VoIP service'' means an interconnected voice over Internet Protocol service that: (1) enables real-time, two-way voice communications; (2) requires a broadband connection from the user's location; (3) requires Internet protocol-compatible customer premises equipment (CPE); and (4) permits users generally to receive calls that originate on the public switched telephone network and to terminate calls to the public switched telephone network. See 47 C.F.R. � 9.3. = Does that mean: 1. That if I allow only outbound, I'm not required to comply? (number 4 above) 2. That if I setup the customer, as a remote node on my hosted asterisk server for inbound and outbound of just their business phone, but have it setup that 911 when dialed from the phone should use a backup proxy that is local and doesnt require a broadband connection (it's using the LAN), that I'm not required to comply? (number 2 above), the setup in mind is where I use a polycom phone with a Sipura SPA 3000, configured that when 911 is dialed it uses the Sipura and not Asterisk. 3. If the answer to 2 is that yes I have to comply, then why when installing a lagecy pbx (like avaya) that doesn't have battery backup I'm not required to mail stickers? is that because I don't use the name VoIP? and the media hasn't caught up to this one (I'm sure it happended before)? Well thats it for now, I'm sure I have more questions :) ___ --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] Error on using queue.
Thanks. I made change to joinempty=yes. And now I
can hear the music on hold. But it would not ring the agent even if I login
agent in. When I run show queue command under CLI, I got these
messages:
queue1 has 1
calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2,
SL:0.0% within 0s Members:
Agent/555997 (Unavailable) has taken no calls
yet Agent/555998 (Unavailable) has taken
no calls yet
It seems that something wrong with my config file,
it did not login any agent.
- Original Message -
From:
Dov Bigio
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 8:33
AM
Subject: Re: [Asterisk-Users] Error on
using queue.
If you are using 1.2, it might be the joinempty
and leavewhenempty parameters.
Their default are different than the 1.0.x
releases
- Original Message -
From:
gc
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 11:27
AM
Subject: [Asterisk-Users] Error on
using queue.
I am trying to use * as ACD server for our sip
proxy.
I first dial 55 to login 98 as
ACD agent it worked fine and then when I dialed 98, I got these messages from * CLI:
-- Executing
Answer("SIP/98-f718", "") in new stack --
Executing Ringing("SIP/98-f718", "") in new
stack -- Executing Wait("SIP/98-f718", "2") in
new stack -- Executing Queue("SIP/98-f718",
"queue1") in new stackNov 30 16:54:12 WARNING[7579]: app_queue.c:3078
queue_exec: Unable to join queue 'queue1' -- Executing
Hangup("SIP/98-f718", "") in new stack == Spawn extension
(default, 99, 5) exited non-zero on
'SIP/5025155598-f718'
Can anybody tell me what cause this
problem?
The followings are my configuration
files:
extensions.conf:
[default]
;For incoming call to ring into the
queue.exten= 99,1,Answerexten=
99,2,Ringingexten= 99,3,Wait(2)exten=
99,4,Queue(queue1)exten= 99,5,Hangup
;Agent loginexten =
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent
logoutexten = 55,1,AgentCallBackLogin(|1)
exten =
97,1,Dial(SIP/97)exten =
98,1,Dial(SIP/98)
agents.conf:
[Agent1]agent =
97,,Gary1agent = 98,,Gary2
queues.conf:
[queue1]musiconhold =
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen
= 0announce-frequency = 0announce-holdtime = nomember =
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw
[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
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Re: [Asterisk-Users] US e911 reminder
On Thu, 2005-12-01 at 11:40 -0500, C F wrote: On 11/28/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: Enforcement Bureau Outlines Requirements of November 28, 2005 Interconnected Voice Over Internet Protocol 911 Compliance Letters http://www.fcc.gov/eb/Public_Notices/DA-05-2945A1.html I'm just trying to clarify this, according to note 1 on that paper: 1 ``Interconnected VoIP service'' means an interconnected voice over Internet Protocol service that: (1) enables real-time, two-way voice communications; (2) requires a broadband connection from the user's location; (3) requires Internet protocol-compatible customer premises equipment (CPE); and (4) permits users generally to receive calls that originate on the public switched telephone network and to terminate calls to the public switched telephone network. See 47 C.F.R. � 9.3. The FCC is quoting 47 CFR section 9.3. http://www.access.gpo.gov/nara/cfr/waisidx_05/47cfr9_05.html There are a lot of legal precedents on how to interpret law in the US. The CFR is law so those rules apply. Generally use of the owrds 'and' and 'or' are important. In element 4 it uses 'and' to discuss in/out bound calls. So you 'generally' have to provide both (I dislike the use of the word generally as it is unclear what that means exactly). You have to provide all 4 elements as well, although generally speaking the first 3 are a given - but one could make an argument you dont require broadband becuase you use low bitrate codecs :) = Does that mean: 1. That if I allow only outbound, I'm not required to comply? (number 4 above) correct 'and' is critical. The supreme court has ruled on many occasions that lawmakers words are infact binding, specifically over interpretation of laws. 2. That if I setup the customer, as a remote node on my hosted asterisk server for inbound and outbound of just their business phone, but have it setup that 911 when dialed from the phone should use a backup proxy that is local and doesnt require a broadband connection (it's using the LAN), that I'm not required to comply? (number 2 above), the setup in mind is where I use a polycom phone with a Sipura SPA 3000, configured that when 911 is dialed it uses the Sipura and not Asterisk. That I think is pushing it. I really dont know how they would rule on that, my guess is they would say that their intent was to cover that type of activity so it sucks to be you. I did not see a definition of broadband internet though, it could be that they consider broadband anything greater than 56k, or it could be that they consider it a lot higher. Maybe I just missed the definition, but regardless if you dont *require* a broadband internet connection to use your service because you allow people at most 2 calls at a time via the gsm codec for example, you dont fit all the elements to be required. Now I am sure that was in the intent of the fccs decision but that is a little more clear than tossing equipment at a customer site and providing service off the lan vs off a 'broadband connection' that you provide ... 3. If the answer to 2 is that yes I have to comply, then why when installing a lagecy pbx (like avaya) that doesn't have battery backup I'm not required to mail stickers? is that because I don't use the name VoIP? and the media hasn't caught up to this one (I'm sure it happended before)? I am sure that is part of it. In effect this new legislation forces VoIP companies to pay RBOCs for access to the PSAP (generally e911 is provided by a direct connect to the switch where the psap is connected). By providing a passthrough the RBOCs (who have been complaining about unfair competition from voip for years) now get some money from their competition. Interresting isnt it? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group 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] voipbuster
Tony Hoyle a écrit : [...] call1899 call18866 voipbuster sipdiscount voipcheap (note: this one uses a proprietary protocol, similar to IAX but over different ports and not compatibile). NetAppel.fr -- Daniel ___ --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: Codec Problem
Hi, My IP Phone is using well G.723.1 because when i am testing it with another SIP GK, working well with G.723.1. But the problem is only accuring in Asterisk, my sip.conf is already having the configuration of this codec. [123456] disallow=all allow=g723 -- Thank You, Code Lover ___ --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 Perl AGI, bug with stream_file() ?
Benoît Mérouze wrote:
Hello,
On http://www.voip-info.org/tiki-index.php?page=Asterisk%20perl%20agi,
it's said that stream_file() might returns -1 on error or hangup, 0
if playback completes without a digit being pressed, or the ASCII
numerical value of the digit if a digit was pressed.
But actually when I hangup my phone stream_file returns 0 and the AGI
is stopped even if I have a callback function, set with setcallback(),
to catch the hangup signal and exit properly the AGI.
In the Asterisk logs, instead of having AGI Script myagi.agi
completed, returning 0, I have Spawn extension (default, 777, 2)
exited non-zero on 'IAX2/7-5' ...
(777 is the extension to launch myagi.agi, and 7 is my phone's
username/number).
Is that a bug?
After reading some documentations, I discovered it was possible to
handle the 'hup' signal.
I still have to use the setcallback method to use my callback function
when the user hangs up, but I've also added $SIG{HUP} = \callback; to
call this function when the user hangs up during the execution of
stream_file().
Then I guess I need to use both $SIG{HUP} and $AGI-setcallback()
methods to catch any hungup.
Am I right ?
--
Benoit Merouze
Ingenieur Developpement d'Application Reseau
[EMAIL PROTECTED]
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Re[2]: [Asterisk-Users] voipbuster
http://www.mujtelefon.com -- [EMAIL PROTECTED] Alejandro Vargas wrote: btw. does anyone have a definitive list of all the finarea VOIP companies? I can think of: call1899 call18866 voipbuster sipdiscount voipcheap (note: this one uses a proprietary protocol, similar to IAX but over different ports and not compatibile). Tony ___ --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] Problem compiling libmfcr2 on FC4
I am installing a new Asterisk server that needs mfcr2 on a machine running Fedora Core 4. I have compiled both asterisk and spandsp without any problem. On the last step, compiling libmfcr2-0.0.3 I get the following error: libtool: link: only absolute run-paths are allowed make[1]: *** [protocol_mfcr2.la] Error 1 make[1]: Leaving directory `/usr/src/libmfcr2-0.0.3' make: *** [all] Error 2 I have tried both versions pre7 and pre8 and both give the same error. Am I missing something? I compiled spandsp, then libsupertone and then libunicall. They all compiled and installed. The configure script for libmfcr2 does not sned any error that I can see. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico 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] iaxmodem
Hi: I want to use the same phone number for the fax and voice conversations. If it is a fax calling, I don't want any interactive menu, I just want to redirect the calling to the iaxmodem extension, and if is a normal calling the interactive menu will be deployed. How can I detect that is fax calling? Regards Miguel ___ --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] Altering Incoming CallerID
What do I need to do to alter incoming CallerID? The below isn't
working...
Running Asterisk 1.2 CVS HEAD
exten = NXXNXX,1,Wait(1)
exten = NXXNXX,2,Set(CALLERID(name) = Fred)
exten = NXXNXX,3,NoOp(${CALLERID(name)})
-- Executing Wait(IAX2/A-9, 1) in new stack
-- Executing Set(IAX2/A-9, CALLERID(name) = Fred) in new stack
-- Executing NoOp(IAX2/A-9, Hugh Johnson) in new stack
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Re: [Asterisk-Users] Problem compiling libmfcr2 on FC4
Ho Carlos, When you build the software specify an install directory explicitly on the command line, like: ./configure --prefix=/usr/local There is an error in the configuration files when you let the installation default to /usr/local. If you specify it, things work. The next revision will fix this. Steve Carlos Chavez wrote: I am installing a new Asterisk server that needs mfcr2 on a machine running Fedora Core 4. I have compiled both asterisk and spandsp without any problem. On the last step, compiling libmfcr2-0.0.3 I get the following error: libtool: link: only absolute run-paths are allowed make[1]: *** [protocol_mfcr2.la] Error 1 make[1]: Leaving directory `/usr/src/libmfcr2-0.0.3' make: *** [all] Error 2 I have tried both versions pre7 and pre8 and both give the same error. Am I missing something? I compiled spandsp, then libsupertone and then libunicall. They all compiled and installed. The configure script for libmfcr2 does not sned any error that I can see. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] Motherboard choice for asterisk?
On 09:47, Thu 01 Dec 05, Giovanni Miano wrote: If u want kernel 2.6 dont use SMP support Why not ? Seems to workout pretty nice here. Intel 865 board with HyperThreading P4 3Ghz. Linux 2.6.10 SMP PREEMPT HIGHMEM Haven't seen any trouble here. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Re: US e911 reminder
Without going thru the ditail to much - im not shure that im allowed to reveal tom much - but we are using a webservice to update their database. Best regards jan --On Thursday, December 01, 2005 08:21:23 AM -0500 Matt [EMAIL PROTECTED] wrote: How are you updating the E911 address information? We have literally been pulling teeth at Verizon to get access to their PS/ALI database to make the updates that we need to. On 11/30/05, Jan Saell [EMAIL PROTECTED] wrote: Just a small note that we have used a cluster of asterisk to connect or Voip systems to the HFB E911 service and it worked without any problems. SO one can defenetly use asterisk in one of these environments. Best regards jan --On Monday, November 28, 2005 03:01:25 PM -0800 trixter aka Bret McDanel [EMAIL PROTECTED] wrote: Just a reminder tonight at midnight is the deadline for pstn connected VoIP providers operating in the US to provide E911 or face fines upto $11,000 per day. There is also a filing requirement with the FCC which is due tonight as well. Enforcement Bureau Outlines Requirements of November 28, 2005 Interconnected Voice Over Internet Protocol 911 Compliance Letters http://www.fcc.gov/eb/Public_Notices/DA-05-2945A1.html Consumer page but has some basic info http://ftp.fcc.gov/cgb/consumerfacts/voip911.html -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B ___ --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 -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B pgpB1tMcbMreM.pgp Description: 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
Re: [Asterisk-Users] Problem compiling libmfcr2 on FC4
On Fri, 2005-12-02 at 01:52 +0800, Steve Underwood wrote: Ho Carlos, When you build the software specify an install directory explicitly on the command line, like: ./configure --prefix=/usr/local There is an error in the configuration files when you let the installation default to /usr/local. If you specify it, things work. The next revision will fix this. Steve Thank you. I will try it later today. By the way, do you think having 8 E1 with MFCR2 on a single dual Xeon server viable? -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico 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[3]: [Asterisk-Users] voipbuster
sorry, this is mistake -- [EMAIL PROTECTED] http://www.mujtelefon.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] prepaid application
Hi Scott, Yes, its possible pass 'm' option to Dial command for MusicOnHold If destination is unreachable, you need to get the return value of Dial and from that value you will know whether a call was connected or not. Based on that value you can execute Dial again or not. You can put everything in an AGI script. AGI is really fun ! Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005 07:51:58 + To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] prepaid application Hi All I am using prepaid auth (callingcards), the idea is for a prepaid support line. It is up and running but I have a couple of questions with regards to modifications I would like to make. When a user calls and they go through the process of entering their card number. They are then asked for a destination. What I would like to be able to do is not have it ask for a destination and automatically dial a number? At present I ask them to enter a default number when it ask for a destination and this then takes them to a queue, if someone is available it rings and goes through, if no one is available rather than sit in the queue and listen to the lovely onhold music prepaid auth comes back and says that destination is unreachable, is there a way to get it to just wait in the queue? Many Thanks In Advance Scott Pinhorne ___ --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] Call Recording
Hi all, Perhaps a newby question, perhaps something impossible. While waiting for my HW to arrive, i've been studying the wiki's and TFOT to be preparred when it comes. Info is overwhelming. It seems that anything is possible... Is it possible to record allways from begin to end an entire conversation, without anybody having to press keys? I know it is possible as in an answering machine. But is it also possible for an answered call? And most ideal would be if it was ogg (or mp3), stereo, each call-party its own (L/R)channel. So if both parties speak at the same moment, eachr can still be heard, by turning either the left or right volume off. HtH, Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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 Recording
Hi, On 19:27, Thu 01 Dec 05, Hans Witvliet wrote: Hi all, Perhaps a newby question, perhaps something impossible. While waiting for my HW to arrive, i've been studying the wiki's and TFOT to be preparred when it comes. Info is overwhelming. It seems that anything is possible... Is it possible to record allways from begin to end an entire conversation, without anybody having to press keys? Yes, have a look at the Monitor command. I know it is possible as in an answering machine. But is it also possible for an answered call? And most ideal would be if it was ogg (or mp3), stereo, each call-party its own (L/R)channel. So if both parties speak at the same moment, eachr can still be heard, by turning either the left or right volume off. Monitor can mix the 2 sides together, or leave the 2 sides in a seperate file. Just put together a shellscript to run when channel is hangup and do the mixing yourself. Soxmix should be able to mix it L/R I guess. The wav can then be converted to .ogg. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Altering Incoming CallerID
Why aren't you using the SetCallerID() cmd?
--
Tom
On 12/1/05, Hugh L. Johnson [EMAIL PROTECTED] wrote:
What do I need to do to alter incoming CallerID? The below isn't
working...
Running Asterisk 1.2 CVS HEAD
exten = NXXNXX,1,Wait(1)
exten = NXXNXX,2,Set(CALLERID(name) = Fred)
exten = NXXNXX,3,NoOp(${CALLERID(name)})
-- Executing Wait(IAX2/A-9, 1) in new stack
-- Executing Set(IAX2/A-9, CALLERID(name) = Fred) in new stack
-- Executing NoOp(IAX2/A-9, Hugh Johnson) in new stack
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--
Tom
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RE: [Asterisk-Users] Altering Incoming CallerID
Try,
Set(CALLERIDNAME=Innocent Evil)
Thanks,
--
You don't have any choice, you already made it before you came here.
-Original Message-
From: [EMAIL PROTECTED]
Sent: Thu, 01 Dec 2005 12:48:13 -0500
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Altering Incoming CallerID
What do I need to do to alter incoming CallerID? The below isn't
working...
Running Asterisk 1.2 CVS HEAD
exten = NXXNXX,1,Wait(1)
exten = NXXNXX,2,Set(CALLERID(name) = Fred)
exten = NXXNXX,3,NoOp(${CALLERID(name)})
-- Executing Wait(IAX2/A-9, 1) in new stack
-- Executing Set(IAX2/A-9, CALLERID(name) = Fred) in new stack
-- Executing NoOp(IAX2/A-9, Hugh Johnson) in new stack
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Re: [Asterisk-Users] format
Youch. That's quite the switch! I'm surprised you couldn't HEAR the difference. :) -- Tom On 11/30/05, Steve Totaro [EMAIL PROTECTED] wrote: I think if you type show codecs in the CLI you can see what codecs are what by the number. It shows that you tried for g728 but got iLBC. -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 30, 2005 8:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] format Can anyone tell me what this line means? -- Accepting AUTHENTICATED call from 202.125.42.141, requested format = 256, actual format = 1024 does this mean a certain codec was requested but another one was delivered? Is there some configuration that I can make to improve the call quality? Currently my IAX2 Outbound trunk looks like; allow=ilbcg726ulaw auth=md5 context=from-pstn disallow=all host=202.125.42.252 qualify=3000 secret=82XXX type=friend username=0960XXX tia, Dean ___ --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 -- Tom ___ --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 Recording
What you wanna to do if there have more than 2 parties in the conversation ? !! -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005 19:27:45 +0100 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Call Recording Hi all, Perhaps a newby question, perhaps something impossible. While waiting for my HW to arrive, i've been studying the wiki's and TFOT to be preparred when it comes. Info is overwhelming. It seems that anything is possible... Is it possible to record allways from begin to end an entire conversation, without anybody having to press keys? I know it is possible as in an answering machine. But is it also possible for an answered call? And most ideal would be if it was ogg (or mp3), stereo, each call-party its own (L/R)channel. So if both parties speak at the same moment, eachr can still be heard, by turning either the left or right volume off. HtH, Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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] Altering Incoming CallerID
Didn't work. On Thu, 2005-12-01 at 10:36 -0800, Innocent Evil wrote: Set(CALLERIDNAME=Innocent Evil) ___ --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 Recording
Use sox to make a quadriphonic (4 channels) audio file. Any more than 4 in a call would be silly ;-) Innocent Evil wrote: What you wanna to do if there have more than 2 parties in the conversation ? !! -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005 19:27:45 +0100 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Call Recording Hi all, Perhaps a newby question, perhaps something impossible. While waiting for my HW to arrive, i've been studying the wiki's and TFOT to be preparred when it comes. Info is overwhelming. It seems that anything is possible... Is it possible to record allways from begin to end an entire conversation, without anybody having to press keys? I know it is possible as in an answering machine. But is it also possible for an answered call? And most ideal would be if it was ogg (or mp3), stereo, each call-party its own (L/R)channel. So if both parties speak at the same moment, eachr can still be heard, by turning either the left or right volume off. HtH, Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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
Re: [Asterisk-Users] Altering Incoming CallerID
It worked, but... SetCIDName() SetCIDNum() are depreciated; I figured SetCallerID() is on the way out, too. I'd rather just touch part than have to mess with the whole. On Thu, 2005-12-01 at 13:35 -0500, Tom Hayden wrote: Why aren't you using the SetCallerID() cmd? ___ --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] prepaid application
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of scott
Sent: Wednesday, November 30, 2005 11:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] prepaid application
Hi All
I am using prepaid auth (callingcards), the idea is for a
prepaid support line. It is up and running but I have a
couple of questions with regards to modifications I would
like to make.
When a user calls and they go through the process of entering
their card number. They are then asked for a destination.
What I would like to be able to do is not have it ask for a
destination and automatically dial a number?
How about something like:
exten = 1234567,1,read(CARDNUM,promptfile)
exten = 1234567,2,agi(astcc.agi,${CARDNUM},5566)
...where promptfile is the name of the prompt instructing the caller
to enter his account number, ...followed by the pound sign, and 5566
is the extension you want dialed after authentication.
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.362 / Virus Database: 267.13.10/189 - Release Date:
11/30/2005
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Re: [Asterisk-Users] Altering Incoming CallerID
Hugh L. Johnson wrote: It worked, but... SetCIDName() SetCIDNum() are depreciated; I figured SetCallerID() is on the way out, too. I'd rather just touch part than have to mess with the whole. On Thu, 2005-12-01 at 13:35 -0500, Tom Hayden wrote: Why aren't you using the SetCallerID() cmd? ___ --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 CLI* show function CALLERID -- Christopher L. Wade, CCNA, CCDA, CQS-CIPCES, CQS-CWLSS ___ --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] prepaid application
Hi All I am using prepaid auth (callingcards), the idea is for a prepaid support line. It is up and running but I have a couple of questions with regards to modifications I would like to make. When a user calls and they go through the process of entering their card number. They are then asked for a destination. What I would like to be able to do is not have it ask for a destination and automatically dial a number? I have done this exact setup for a prepaid information service. I did everything by editing astcc.agi. I also have it setup to handle three different languages. What I did for this was to edit the agi exec dial and hard code the dial number right in. I also modified the script to not disconnect a caller when their credit runs out since that is pretty rude. Email me offlist for help. Thanks, Steve ___ --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] voipbuster
On Thu, December 1, 2005 17:09, Don Fanning said: I ended up buying a second 1 euro account because of this. But it does work fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Thursday, December 01, 2005 7:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voipbuster I was testing voipbuster. With a new account, with no credit, I can make calls perfectly but of 1 minute. But I tried the username and passwrord of an account with credit, and the registration is refused. With the voipbuster propietary software it works ok (I sniffed the packets and I think it is not using standard iax or sip ports). Are the acconts with credit blocked for avoiding it's use with ohter software than voipbuster's? I tryed to send a mail to voipbuster's support but I never received an answer (then do not support other thing than their software). Mine works just fine. It's a pain though if you have to get a new account, as minimum amount is now EUR 5... OTOH, for free calls, it might be worth it... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] chan_bluetooth and Ericsson/SonyEricsson models
Hi, They are any succes stories with chan_bluetooth and one of the following phone models? - Ericsson R520m - SonyEricsson T68i - SonyEricsson W800i I have tried with all of them with different kind of errors... Thank you and best regard, Dan ___ --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 Recording
Or put everyone in a Meetme room and record the conversation in the meetme room -- just an idea. - Waldo On Dec 1, 2005, at 2:00 PM, Dave Walker wrote: Use sox to make a quadriphonic (4 channels) audio file. Any more than 4 in a call would be silly ;-) Innocent Evil wrote: What you wanna to do if there have more than 2 parties in the conversation ? !! -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005 19:27:45 +0100 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Call Recording Hi all, Perhaps a newby question, perhaps something impossible. While waiting for my HW to arrive, i've been studying the wiki's and TFOT to be preparred when it comes. Info is overwhelming. It seems that anything is possible... Is it possible to record allways from begin to end an entire conversation, without anybody having to press keys? I know it is possible as in an answering machine. But is it also possible for an answered call? And most ideal would be if it was ogg (or mp3), stereo, each call- party its own (L/R)channel. So if both parties speak at the same moment, eachr can still be heard, by turning either the left or right volume off. HtH, Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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 ___ --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] Sip trunk between Avaya S8700 and Asterisk
Has anyone been able to set up a sip trunk between and Avaya S8700 and Asterisk? I can't seem to find any good docs on the subject. Any help would be greatly appreciated. Thanks! =A= ___ --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] sixtel
Just curious... Is there anyone out there who has given this outfit money and actually received any service from them? ___ --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] Cisco 7970
Thank you Kerry. I was able to download the firmware. Does anybody know what files need to reside on the tfpt server. If someone is willing to help get my 7970 phone functional again, I would really appreciate it. -John You have to have a login to the Cisco site to download the firmware. -Kerry -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Riek Sent: Tuesday, November 29, 2005 2:02 PM To: asterisk user list Subject: [Asterisk-Users] Cisco 7970 I have the same problem after doing a factory reset. Does anybody have the website link to download firmware for the Cisco phones? Thanks, John Riek I ran into this same problem the other day. What you need to do is put all firmware files in the tftp root directory. The trick with the files is you need to match the case of the filename that the phone is looking for. My XmlDefault.cnf.xml needed to have the proper case. If you do a tcpdump on your server you can see what file its getting stuck on. This is how I figured out what it is looking for: tcpdump -i eth1 port tftp -vv It will output what file the phone is looking for. Have my 7970 working great with *. Hope this helps. Jeremiah On Nov 7, 2005, at 10:24 AM, asterisk-users-request at lists.digium.com wrote: Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine dan at cytexone.com 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --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] Error on using queue.
How is your agents.conf ? How is your login in
extensions.conf?
- Original Message -
From:
gc
To: Dov Bigio ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 2:53
PM
Subject: Re: [Asterisk-Users] Error on
using queue.
Thanks. I made change to joinempty=yes. And now I
can hear the music on hold. But it would not ring the agent even if I login
agent in. When I run show queue command under CLI, I got these
messages:
queue1 has 1
calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2,
SL:0.0% within 0s Members:
Agent/555997 (Unavailable) has taken no calls
yet Agent/555998 (Unavailable) has taken
no calls yet
It seems that something wrong with my config
file, it did not login any agent.
- Original Message -
From:
Dov Bigio
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 8:33
AM
Subject: Re: [Asterisk-Users] Error on
using queue.
If you are using 1.2, it might be the joinempty
and leavewhenempty parameters.
Their default are different than the 1.0.x
releases
- Original Message -
From:
gc
To: Asterisk Users Mailing List
- Non-Commercial Discussion
Sent: Thursday, December 01, 2005
11:27 AM
Subject: [Asterisk-Users] Error on
using queue.
I am trying to use * as ACD server for our
sip proxy.
I first dial 55 to login 98
as ACD agent it worked fine and then when I dialed 98,
I got these messages from *
CLI:
-- Executing
Answer("SIP/98-f718", "") in new stack --
Executing Ringing("SIP/98-f718", "") in new
stack -- Executing Wait("SIP/98-f718", "2")
in new stack -- Executing
Queue("SIP/98-f718", "queue1") in new stackNov 30 16:54:12
WARNING[7579]: app_queue.c:3078 queue_exec: Unable to join queue
'queue1' -- Executing Hangup("SIP/98-f718",
"") in new stack == Spawn extension (default, 99, 5)
exited non-zero on 'SIP/5025155598-f718'
Can anybody tell me what cause this
problem?
The followings are my configuration
files:
extensions.conf:
[default]
;For incoming call to ring into the
queue.exten= 99,1,Answerexten=
99,2,Ringingexten= 99,3,Wait(2)exten=
99,4,Queue(queue1)exten= 99,5,Hangup
;Agent loginexten =
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent
logoutexten = 55,1,AgentCallBackLogin(|1)
exten =
97,1,Dial(SIP/97)exten =
98,1,Dial(SIP/98)
agents.conf:
[Agent1]agent =
97,,Gary1agent = 98,,Gary2
queues.conf:
[queue1]musiconhold =
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen
= 0announce-frequency = 0announce-holdtime = nomember =
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw
[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
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Re: [Asterisk-Users] chan_sip.c error
I get a similar warning with 1.2b1 Anyone have a clue as to what this means?? John Novack asterisk183 wrote: Why Asterisk show this message: Nov 30 17:05:17 WARNING[1351]: chan_sip.c:9600 handle_response_register: Got 200 OK on REGISTER that isn't a register 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
[Asterisk-Users] CID text stripped over IAX
I have two servers connected via IAX2; one is connected to PRIs where I receive CallerID along with CID text while the other is located over my network connected to some channel banks providing analog dialtone. Relevant output of show channel on the PRI box for one call is here: CDR Variables: level 1: clid=Joe Q Iglou 7005551234 level 1: src=7005551234 level 1: dst=700666 The show channel output of the same call on the other side of the IAX2 link: CDR Variables: level 1: clid=7005551234 level 1: src=7005551234 level 1: dst=700666 Poof, CID text gone! The iax.conf config of both doesn't attempt to override CID in any fashion, unless there is some sort of subtle default to eat the CID text that I am unaware of. I have a basic sort of config (peer/user pairing) to connect these two servers together. Is this normal behavior? SIP calls get the CID text passed just fine but once they go over this IAX2 connection, the text gets chopped off. -- Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/ GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A FABB 599D FF67 6C9E disclaimer: My opinions are my own. Don't bother my employer about them. pgpAUO3oE5XK6.pgp Description: 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] default user name and password for a2billing
What is the default username and password for [EMAIL PROTECTED] a2billing module. 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
