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Postby mflorell » Sun Sep 03, 2006 5:14 am

This looks like a rather odd problem. You don't seem to have any meetme errors, but you say on an IAX trunk that there is no audio once the customer channel joins the conference? and that this only happens on auto-dial?

Yes, you can add that line and things should work just fine.

Yes, manual dial and auto dial both use meetme conferences although the calls are delivered to the meetme conference in a different way.
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Postby espencer » Sun Sep 03, 2006 10:30 am

yes, that is correct the no sound only happens on auto-dial. anybody can pop into the same conference room directly by dialing the conference room number.

more interesting yet (as i have done more testing) is that the agent can park the call and i can hear glenn gould? playing franz liszt? nice music either way. even more interesting is that i can send DTMF from the agent web interface and the remote party can hear some disturbance (not full DTMF) but they really appear to be connected in some way.

i have wondered if there is a codec issue somewhere so i have gone from allowing only ulaw to allowing all codecs but experience the same issues.

About the only thing i understand very well is the asterisk dialplan. I am wondering if all the info for next available agent is in vicidial_live_agents. if so, i can probably write some dialplan quickly that accesses this table directly and sends the call directly to the conference.

the more of this i see working, the more impressed i am. the client web interface is really impressive. i had not seen all the transfer stuff before. i just tried a blind transfer from the client web interface and i got sound in both directions once the call was transferred. really seems to be the meetme room only on autodial. almost like i need to click something or unmute. this is truly odd.

thanks for reading this.
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Postby mflorell » Sun Sep 03, 2006 8:56 pm

Are you going through a firewall in any way?

Have you tried a different client softphone?

The only times I have heard of this kind of issue is with either a firewall or the zaptel timer causing the problem. There are also sometimes issues with softphones not working sometimes.
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Postby espencer » Mon Sep 04, 2006 9:30 am

thanks for your help!

all the computers and phones are on a lan in my house so there should be no firewalls. the phones involved can all join the same conference room without any problem by dialing it directly.

i followed the scratch install yet another time on an old piii and it works well with an old x100p card in it and the 2.4.31 slackware linux kernel and the beta version of astguiclient. i can swap this box and the one that doesn't work back and forth with the same phones connected and the one with the wildcard works, the one with the te110p does not. i would have though for sure it was a zaptel issue except i can dial into the meetmes directly without any trouble.

On the te100p box i did this:

; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,AGI(call_log.agi,${EXTEN})
exten => 8365,2,Meetme(8600100)
;exten => 8365,2,Playback(custom/arrivaltelvicidial)
;exten => 8365,2,AGI(agi-VDADtransfer.agi,${EXTEN})
;exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
;exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
;exten => 8365,5,Hangup

and, just as before, the dialed number gets the call and the call gets dumped into the conference (can hear the beep and see it on the CLI when call is tranferred in, can hear the beep in the conference when the callee hangs up but "live call" doesn't light -- guess that is done by the agi script) but still no other sound. if that phone dials into the conference directly, the meetme room has sound.

i tried to put an x100p into the odd box but it won't boot (i think the slots are 3.3 volt). i have also switched back and forth between the 1.12 version and the 2.0 beta version but i am getting the same results either way.

the box that works is kind of old but is a ~1ghz piii with 512mb of ram. i set it up as a sort of dialing appliance with no phones connected directly to it that connects VIA an IAX trunk to my main house box and used the EXTERNAL (instead of SIP) setting for the phones and it works fine -- pretty flexible system you built -- so in a worst case scenario i just have 2 boxes instead of 1.

any guesses how many calls/agents a 1ghz piii/512mb ram/plain IDE 40 gig drive could do assuming no transcoding?
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Postby espencer » Mon Sep 04, 2006 9:44 am

just remembered (not sure if it is valuable info or not) ****begin shameless name dropping *****when i was having one of those foot-long costco polish dogs with Jared Smith (one of the authors of the O'Reilly book - The Future of Telephony) **** end shameless name dropping but can you blame me since he is the only asterisk celebrity i have met?**** a few months ago he mentioned that even if your timing/zaptel are perfect, SIP gets its timing from the return RTP stream or something like that (i could be misquoting completely) so if the remote end has issues, there is nothing you can do about it.

by the way, i just told jared about your project last week (he had already heard about it, of course) and i told him how cool it is. i think i will be showing it to him pretty soon. this piii i have it working on has the new beta version and works really well and shows it off alright (as long as you don't look at the old, scuffed beige mini-tower it is running on).

one other idea: i cannot seem to get a 2.6 kernel (using make menuconfig from the scratch install -- i am a linux moron) on any of my hardware so i can never use the kernel ztdummy timer. is there a 2.6 kernel-based install-cd using distro that will install on newer hardware (sata, p4, hyperthreading) and be a good base and similar to the slackware one that i could do the scratch install steps on? i am getting really fast at the scratch install. it is not nearly so daunting anymore.
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Postby mflorell » Mon Sep 04, 2006 11:51 am

That is true about SIP and timing, but you must have good zaptel timing for meetme to work well no matter the type of channel you put into it. There have also been several bug fixes since 1.0 Asterisk that have greatly helped to sync the SIP timing in a meetme room.

You should have seen the first test Asterisk box I setup over 3 years ago, it didn't even have sides :)

The instructions for building a 2.6 kernel in the SCRATCH_INSTALL are not correct. I have installed 2.6.17 on two VICIDIAL servers and they are doing fine in production, the only issue with building 2.6 is that you need to add a lot more modules just for Asterisk than you did under 2.4 kernels. I am planning on putting a 2.6 kernel build instruction into the formal 2.0.1 release. I just need to document things as I install a new server this week.

When I installed 2.6 I just did it on Slackware 10.2 with the newest stable kernel source downloaded from kernel.org. As for a recommendation of a distro to use with 2.6 in use by default. I don't really like any of them as much as Slackware, but you could try FC5(fedora core) or CentOS.
Last edited by mflorell on Mon Sep 04, 2006 7:13 pm, edited 1 time in total.
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Postby espencer » Mon Sep 04, 2006 6:55 pm

I'll bet your first asterisk computer had insides though. computers don't need to be pretty. i tell my wife that all the time but she doesn't believe me.

i noticed my directories in /var/spool/asterisk keep moving to /var/spool/asterisk/ORIG. i was getting CLI errors but changed asterisk.conf to point to the new location. is that normal? things seem to be working so i am not too worried.

also, i think my system with the sata drive is just not getting the right kernel modules installed. it can't seem to find the drive. i did download the latest linux kernel from the link in the scratch install (2.6.17.11) but maybe i need to borrow a .config file from somebody else with a p4 and sata drive. anybody out there with a Supermicro 5014C-MR running a 2.6 kernel? turns out my processor is not HT so that is a little easier.

next thing to try is to image the linux build from my old working piii, put it on my sata only machine, boot with a cd and move the sata kernel over the top of the freshly "restored" sata only machine.
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Postby mflorell » Mon Sep 04, 2006 7:12 pm

espencer wrote:I'll bet your first asterisk computer had insides though. computers don't need to be pretty. i tell my wife that all the time but she doesn't believe me.


yep it had insides, but nothing was screwed down, just held in with plastic clips if that.

i noticed my directories in /var/spool/asterisk keep moving to /var/spool/asterisk/ORIG. i was getting CLI errors but changed asterisk.conf to point to the new location. is that normal? things seem to be working so i am not too worried.


The standard location is supposed to be /var/spool/asterisk/monitor which is what we default to. If you want to change it as you did you would have to modify asterisk's conf files.

also, i think my system with the sata drive is just not getting the right kernel modules installed. it can't seem to find the drive. i did download the latest linux kernel from the link in the scratch install (2.6.17.11) but maybe i need to borrow a .config file from somebody else with a p4 and sata drive. anybody out there with a Supermicro 5014C-MR running a 2.6 kernel? turns out my processor is not HT so that is a little easier.


To start you need to use the sata.i kernel for slackware 10.2 install, then when building for 2.6.17 you need to enable SCSI-emulation and SATA drivers for your chipset on your motherboard. Not the easiest thing to get working the first time around, but there are several howtos out there if you google linux sata kernel

next thing to try is to image the linux build from my old working piii, put it on my sata only machine, boot with a cd and move the sata kernel over the top of the freshly "restored" sata only machine.


Good luck, let us know how it goes.
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