similar to: Solution for 1 time blast of 200, 000 recorded calls

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

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 | -------------

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

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

I have the following situation: Asterisk 1.2.1 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 Some 501's local to my network, some across the great INTERNET divide. PRI connected to Sangoma card. Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker phone when calling from extension to extension. echo not present when calling outbound

Dear R users, I would like to align two protein sequences using BLAST (bl2seq). The question is whether this programm have been implemented in R. Thank you for your help, Alla.

Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:wsiler@education2020.com <mailto:wsiler@education2020.com>

Hello, I am trying to make the following... Can someone tell me if it is possible? Is someone willing to do it from an asterisk@home box? 1. I send an http request to asterisk@home box. Ex: http://asterisk@home/call.php?phone=0033102030405&code=12345 2. Application will call phone number 0033102030405 (using a sip provider) 3. Application will play a pre-recorded voice prompt 4. Application

hello if there is a package can do blast in R? [[alternative HTML version deleted]]

Hi, I'm interested in developing an automated phone survey and am curious if Asterisk could be configured to run such a system.. My idea is to record a message and a series of sub-questions. The system would call each number on a list and play the message, Depending on the touch tone response another message would be played. Is it possible for asterisk to manage a survey like this?

Hi All, I have use the BLAST program provided by NCBI to compare large databases and have generated a lot of output in one long text file. I was looking for a way to parse out the results in a nice way so that it could be easily visualized. I was thinking of trying to write something in R, but I wanted to check in with the help group to make sure I am not re-inventing the wheel. Any help

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:

Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140110/18f3f1e2/attachment.html>

me at tdiehl.org wrote: > On Wed, 4 Oct 2017, hw wrote: > >> Jose Maria Terry Jimenez wrote: >>> El 4/10/17 a las 17:45, david escribi?: >>> >>>> Folks >>>> >>>> A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83. It interfaces to my land-line (POTS) telephone line in the United States. On Windows, I had a good answering

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

Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? Thanks, Roman

We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What we're trying to avoid is hardcoding the IP address in the ACL. We were trying to match by TOS set by Asterisk however it seems we've run into a snag where the packet TOS tends to get reset somewhere on our network. Has anyone had this issue? We're running Cisco everywhere inbetween (even the switches). Is

I'm installing an office in a couple of weeks that will have some nice Polycom FX video units in the conference rooms. I'm thinking that with asterisk-oh323 http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/#section2 I should hopefully get the ability for phone users to dial an extension and participate in video conferences, or just simply phone conference with users in the

Is there any way to set Asterisk to choose what codec to allow for a new call based on current usage? In other words... be able to define a max number of ulaw calls, then after that only allowing g729? The idea here is that in general, a T-1 should be enough for our offices to have phone + citrix + some video (got good QoS in place already). But for usage spikes, user experience would be kept

I've got a SIP channel that appears to be hung up. It's an extension that records a .gsm file and fortunately the recording has stopped. I tried zap destroy channel but I guess that doesn't apply to SIP channels. Any ideas? I issued a restart when convenient but figure there must be a better way. TIA, -Ron -------------- next part -------------- An HTML attachment was scrubbed...