Re: [asterisk-users] conferencing help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Matt, I tried /usr/local/src/zaptel-1.2.22.1# ./zttest -v and it just freezes at this. Opened pseudo zap interface, measuring accuracy... no more outputs, when i cancelled this is what i got. --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 Yeah that's what I thought. Am just trying to remember what caused it though. Maybe Tzafrir will chime in :) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHhIkgDQNt8rg0Kp4RAsY2AKCft6fPiWHgBtdE7dS3FpeGRQqnxACdF/Ee tSjBUM1DzdI1XzSVjRxlJ4s= =SMN2 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote: Hi Matt, I tried /usr/local/src/zaptel-1.2.22.1# ./zttest -v and it just freezes at this. Opened pseudo zap interface, measuring accuracy... no more outputs, when i cancelled this is what i got. --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 does that mean my zaptel is bad? Well, yes. Is ztdummy loaded? cat /proc/zaptel/* What kernel version do you use? What version of Zaptel? What Linux distribution? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Hi Tzafrir, cat /proc/zaptel/* Span 1: ZTDUMMY/1 ZTDUMMY/1 1 Kernel: 2.6.18-5-686 #1 SMP Zaptel: zaptel-1.2.20.1 OS: Debian GNU/Linux 4.0 i downgraded my zaptel from 1.2.22.1 to 1.2.20.1 but still the same. thanks again regards, nhadie Tzafrir Cohen wrote: On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote: Hi Matt, I tried /usr/local/src/zaptel-1.2.22.1# ./zttest -v and it just freezes at this. Opened pseudo zap interface, measuring accuracy... no more outputs, when i cancelled this is what i got. --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 does that mean my zaptel is bad? Well, yes. Is ztdummy loaded? cat /proc/zaptel/* What kernel version do you use? What version of Zaptel? What Linux distribution? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Hi Steve, I see. I have this now, *CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault en *CLI load chan_zap.so Unable to load module chan_zap.so -- on the log file it says, it as already loaded that's why it's unable to load. i tried my calling to my conf 6000 -- Executing Set(SIP/100-081825b0, MEETME_OPTS=iM) in new stack -- Executing Goto(SIP/100-081825b0, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/100-081825b0, 6000|iM|) in new stack == Parsing '/etc/asterisk/meetme.conf': Found == Parsing '/etc/asterisk/meetme_additional.conf': Found -- Created MeetMe conference 1023 for conference '6000' -- Recording -- Playing 'vm-rec-name' (language 'en') -- Executing Set(SIP/100-081825b0, MEETME_OPTS=iM) in new stack -- Executing Goto(SIP/100-081825b0, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/100-081825b0, 6000|iM|) in new stack == Parsing '/etc/asterisk/meetme.conf': Found == Parsing '/etc/asterisk/meetme_additional.conf': Found -- Created MeetMe conference 1023 for conference '6000' -- Recording -- Playing 'vm-rec-name' (language 'en') it's trying to play something 'vm-rec-name' but i cannot hear anything on the phone. i'm using g711. i'm not using trixbox, i just installed asterisk, freepbx, zaptel, etc on a debian box. i'm using all the latest version i downloaded from the website (i used asterisk 1.2). /usr/include# modprobe -l | grep ztdum /lib/modules/2.6.18-5-686/misc/ztdummy.ko /usr/include# modprobe -l | grep zap /lib/modules/2.6.18-5-686/misc/zaptel.ko how do i know if my ztdummy is working properly? thanks again! regards, nhadie Steve Edwards wrote: dave cantera wrote: nhadie, meetme requires a zaptel timing device... ztdummy is unreliable when using meetme conferencing. On Wed, 9 Jan 2008, Nhadie wrote: hi dave thank you for the reply. i have loaded zap and using only ztdummy but still can't hear anything when i dial ti my conference, i think this explains it already. will a sangoma card do? I use ztdummy with meetme conferencing and it works fine on CentOS 4.5. Ztdummy is not an issue until you get xx callers in xx conferences. I think (but have no empirical data to back it up) that a card yields better sound quality at higher call levels. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Steve, I see. I have this now, *CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault en That means the zap channel should be ok. One thing you could do is go to the place you downloaded Zaptel and type: ./zttest -v Do you get numbers (i.e. something close or closish to 100%)? Also, if you just have the extensions: exten = 555,1,Answer() exten = 555,n,Background(demo-echotest) exten = 555,n,Echo() Do you get an answer? You don't really need the brackets on answer and echo but I usually type that way and then add options. :-) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHhFUKDQNt8rg0Kp4RAtfQAKC/mjeswAVxnkzv/HHC/4ZCL92SEwCfRoY7 rIAGfpE/0dh56i9myEbOFfA= =fHxG -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
I will be out of the office on Wednesday, January 9, 2008. If this is an emergency, please call Customer Service at (877) 791-7700. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Hi Matt, I tried /usr/local/src/zaptel-1.2.22.1# ./zttest -v and it just freezes at this. Opened pseudo zap interface, measuring accuracy... no more outputs, when i cancelled this is what i got. --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 does that mean my zaptel is bad? Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Steve, I see. I have this now, *CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault en That means the zap channel should be ok. One thing you could do is go to the place you downloaded Zaptel and type: ./zttest -v Do you get numbers (i.e. something close or closish to 100%)? Also, if you just have the extensions: exten = 555,1,Answer() exten = 555,n,Background(demo-echotest) exten = 555,n,Echo() Do you get an answer? You don't really need the brackets on answer and echo but I usually type that way and then add options. :-) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHhFUKDQNt8rg0Kp4RAtfQAKC/mjeswAVxnkzv/HHC/4ZCL92SEwCfRoY7 rIAGfpE/0dh56i9myEbOFfA= =fHxG -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Hi Matt, it seems i don't have that command. *CLI zap show channels No such command 'zap' (type 'help' for help) *CLI ! abort add ael agent agi cdr databasedebug dnsmgr dontdump dundi extensions feature group helpiax2include indication initloadlocal logger meetme mgcp mixmonitor moh no realtimereload remove restart rtp set showsip skinny soft stopunload *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Local Local Proxy Channel Driver no yes no Skinny Skinny Client Control Protocol no yes no Phone Standard Linux Telephony API D no no no SIP Session Initiation Protocol (S yes yes yes IAX2Inter Asterisk eXchange Driver yes yes yes MGCPMedia Gateway Control Protocol no yes no *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Local Local Proxy Channel Driver no yes no Skinny Skinny Client Control Protocol no yes no Phone Standard Linux Telephony API D no no no SIP Session Initiation Protocol (S yes yes yes IAX2Inter Asterisk eXchange Driver yes yes yes MGCPMedia Gateway Control Protocol no yes no -- Executing NoOp(SIP/104-58ae, Using CallerID 104 104) in new stack -- Executing Set(SIP/104-58ae, MEETME_ROOMNUM=6000) in new stack -- Executing GotoIf(SIP/104-58ae, 0?USER) in new stack -- Executing Answer(SIP/104-58ae, ) in new stack -- Executing Wait(SIP/104-58ae, 1) in new stack -- Executing Set(SIP/104-58ae, MEETME_OPTS=) in new stack -- Executing Goto(SIP/104-58ae, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/104-58ae, 6000||) in new stack Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: hi shane, thanks for your reply. i actually tried 3 phones dialled to the conference, but cant here anything from those phones. i also enabled the usercount so i can hear something at least. but still no sound. i'm using ztdummy, as i dont have a card yet. Can you do a zap show channels in the Asterisk console (without the ) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHgtckDQNt8rg0Kp4RAkw0AJ0R/xZowCQ1FGVNpblcUrdwAi5niACfQ5jh JEjcAt3QDqV3aN0rAZGNq9g= =Zqs+ -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Matt, it seems i don't have that command. :) You'll need to make sure that: 1. You have zaptel compiled 2. You compile Asterisk *after* zaptel is compiled and installed 3. You have either modprobed zaptel + ztdummy or made the service and started it. You didn't say, is this a straight Asterisk machine or trixbox/freepbx? If those are done and it still doesn't work then you can report the errors you get when you type (in the console): module load chan_zap.so - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHgw3pDQNt8rg0Kp4RAvJGAKCtI+GaFMCcNk/PB1VMoyOo67RAwACeM5pJ BH0EhGK4hD+oL7TXu0d33+M= =1HK0 -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Then it's time to build zaptel, then rebuild asterisk later, PaulH On Tue, 2008-01-08 at 13:16 +0800, Nhadie wrote: Hi Matt, it seems i don't have that command. *CLI zap show channels No such command 'zap' (type 'help' for help) *CLI ! abort add ael agent agi cdr databasedebug dnsmgr dontdump dundi extensions feature group helpiax2include indication initloadlocal logger meetme mgcp mixmonitor moh no realtimereload remove restart rtp set showsip skinny soft stopunload *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Local Local Proxy Channel Driver no yes no Skinny Skinny Client Control Protocol no yes no Phone Standard Linux Telephony API D no no no SIP Session Initiation Protocol (S yes yes yes IAX2Inter Asterisk eXchange Driver yes yes yes MGCPMedia Gateway Control Protocol no yes no *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Local Local Proxy Channel Driver no yes no Skinny Skinny Client Control Protocol no yes no Phone Standard Linux Telephony API D no no no SIP Session Initiation Protocol (S yes yes yes IAX2Inter Asterisk eXchange Driver yes yes yes MGCPMedia Gateway Control Protocol no yes no -- Executing NoOp(SIP/104-58ae, Using CallerID 104 104) in new stack -- Executing Set(SIP/104-58ae, MEETME_ROOMNUM=6000) in new stack -- Executing GotoIf(SIP/104-58ae, 0?USER) in new stack -- Executing Answer(SIP/104-58ae, ) in new stack -- Executing Wait(SIP/104-58ae, 1) in new stack -- Executing Set(SIP/104-58ae, MEETME_OPTS=) in new stack -- Executing Goto(SIP/104-58ae, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/104-58ae, 6000||) in new stack Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: hi shane, thanks for your reply. i actually tried 3 phones dialled to the conference, but cant here anything from those phones. i also enabled the usercount so i can hear something at least. but still no sound. i'm using ztdummy, as i dont have a card yet. Can you do a zap show channels in the Asterisk console (without the ) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHgtckDQNt8rg0Kp4RAkw0AJ0R/xZowCQ1FGVNpblcUrdwAi5niACfQ5jh JEjcAt3QDqV3aN0rAZGNq9g= =Zqs+ -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
On Tue, Jan 08, 2008 at 06:45:14PM +1300, Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Matt, it seems i don't have that command. :) You'll need to make sure that: 1. You have zaptel compiled 2. You compile Asterisk *after* zaptel is compiled and installed 3. You have either modprobed zaptel + ztdummy or made the service and started it. In other words, what is the output of the following command from the Asteris CLI: module load chan_zap.so (This tests both cases right away: gives different error messages in the different cases) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: (This tests both cases right away: gives different error messages in the different cases) Sweet :) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHgyysDQNt8rg0Kp4RAn6TAJ95CZiwFSgt8Vp+KKm/SOzfkJzi7QCgvSbO KgdOHiu1dEbD4qJ2BfTfqsY= =1zsg -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
