similar to: MeetMe & ztdummy

I've created a pretty complete HOWTO on creating a Linux Bridge (using Fedora) to shape LAN <--> WAN traffic. It includes installation instructions, a script to configure the bridge (which you install as a service), and 2 scripts to configure the network interfaces using traffic control. http://www.burnpc.com/website.nsf/all/3a64a6369757819686256f960068ad75!OpenDocument If anyone

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

For an office that is using VoIP phones to connect to Asterisk, is gigabit ethernet really necessary for the Asterisk box to connect to the switch? I know that I won't even approach the limits of 100 Mbps, but would gigabit help with latency / collisions when several calls are underway? The fact is, anything going outside the office will be over a data T1, so intuition tells me that 100

Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: ***

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

Hello all, We've discovered that VoIP (IAX2) + Citrix + Video is pegging the measly CPU on the Netopia router our ISP provided. We've got 3Mb/3Mb and will increase to 4/4 next year. The Netopia simply breaks out our WAN IPs, and we've got a switch hooked up to it on the inside (Actually I've got a QoS box in-between). ------------- | Internet | | on Cat5 | -------------

Hi, I'm still strugling with getting an easy to use conference system implemented. I did have app_conference running, but today I upgraded asterisk to 1.0.9 and it stopped working. I've tried following the instructions for compiling app_conference on 1.0.7 but it didn't work. So I went back to ztdummy (I've not had any luck getting this to compile on FC2). Anyhoo, I've

Any help would be appreciated as I am a novice trying to work around a difficult situation. This is what the zaprtc helpfile says: zaprtc, getting zaptel timing out of your realtime clock ======================================================== Make sure that you _dont_ have rtc support compiled into your kernel! INSTALL: make USE: make load REMOVE: make unload I interpreted this as

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video

I have a system with no Digium hardware in it (two others with 2 X100P cards in each of them as well). I'm interested in using MeetMe in the one without the hardware (it works great in the ones with the hardware). I can't use ztdummy, because the system has usb-ohci drivers, rather than usb-uhci. I have read the little there is about zaprtc, and I am wondering whether there is a

I have Asterisk running in Xen virtual machines. Unfortunately, this kind of virtualization makes a real time clock impossible, which in turn makes ztdummy or a Zaptel driver impossible to load, which also makes MeetMe conferences impossible. As an alternative, I have downloaded, patched, compiled and installed the app_conference source code against the headers in Asterisk CVS HEAD. I can load

I feel there is some unclarity about the need for a timer in Asterisk. As Asterisk moves into production environments, this needs to be sorted out. I'm setting up an Asterisk in a production environment with no Digium cards at all. It is planned to be used as a clean SIP hosted PABX, where we connect to the PSTN via gateways with proper echo cancellation. There is no USB driver on the card.

Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652&scid=29 -Ron -------------- next part -------------- An HTML attachment was scrubbed... URL:

Do anybody could compile app_conference on x86_64??? I tryied with two versions of app_conference and got the same problem on compiling: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared recompile with -fPIC app_conference.o: could not read symbols: Bad value" ENVIRONMENT:

Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on

Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference

Hi everyone. I am looking at providing a conference for up to 300 people and was wondering if anyone has scaled meetme to 300 people. Here are some points: 1) I am using an IAX2 gateway hosted on a VOIP service provider. 2) The machine is hosted at the providers site so one has to assume that bandwidth is not going to be an issue. 3) Everything is coming in as ULAW so we won't need to

I ran in to an almost impossible ability to run MeetMe on our poweredge. Our PE is a 4 proc with OHCI and we had no need to buy any PCI cards other than for running MeetMe. If you know how to recompile your kernel, recompile it and make RTC a module (its compiled in by default). Then you can use zaprtc as your timing source. Be sure to `rmmod rtc` before you `insmod zaprtc` because zaprtc is a

Here is the situation: A T1 router going into an office which then plugs into the firewall box then into the switch. None of these devices support QOS.. Is there some sort of box/device that I can place between the T1 router and the firewall box which will allow me to prioritize voice traffic on this link? I can't change the T1 router to something that supports QOS because it has

The USB core was completely rewritten in 2.6, and as such the functions that ztdummy depends on do not exist in 2.6. I get the feeling that these changes are too much to easily fix ztdummy, so I don't expect to see it working on 2.6 any time soon (if ever) I made some small changes to zaprtc to work on 2.6 and I have MoH and Meetme functions working fine in my lab. For production I would