similar to: Asterisk vs. Vocal (Vovida) vs. Bayonne

The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options

I have a DG-104S (which I reset to factory settings, it's DHCP'ing off my network, plugged into the WAN port). The system comes up, and I through the web browser set under Call Agent IP Address to: Notify Entry: dlinkgw@[192.168.1.1]:2427 (192.168.1.1 is the * server) I have RGW Name: and DNS IP addres the DNS IP of the MGCP box and DNS State disabled (not sure what to set it to) --

All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards,

I was trying to do a little searching to see if there has even been a comparison between Asterisk and VOCAL or any of the other OSS packages? "Practical Voice Over IP using VOCAL" published by O'Reilly and Associates, attempts to make a strong case about how scalable VOCAL. Of course, considering that the book is written by the makers of VOCAL, it tends to have a one sided slant.

Ok, Skinny gurus. (btw, I'm super pleased to see development happen on this). Thoughts on this?? I added this context to my skinny.conf: [ppc] device=SEP00022D494F2A context=employees line => 50 ; Dial(Skinny/1@ppc) I've downloaded the 30 day Window eval of VTGO! (PPC) from www.ipblue.com and it hangs on Registering. *CLI> skinny debug Skinny Debugging Enabled

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

Could someone familiar with both break down the most memorable pro's & con's and why you have decided to use Asterisk? Thanks _________________________________________________________________ Add photos to your e-mail with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail

I found an old Dialogic card in an abandoned PC, that I think is a Dialogic D/41EPCI based on some googling.. The lspci output says: 00:09.0 Bridge: PLX Technology, Inc. PCI <-> IOBus Bridge (rev 01) Subsystem: Dialogic Corp: Unknown device 0529 I'm just getting started with Asterisk to build a SOHO system. Any suggestions for what I can do with this card? It would be

Hello, I am very new to Asterisk, so please forgive me for my basic question. I couldn't find a good answer to this on the info boards. I'd like to know if the following is possible and reasonable to accomplish: I'd like to configure a voice recording system using Asterisk and a Tormenta2 Quad T1 card. A co-worker was able to create this system a while back with Bayonne and a

Hi i have an application developed with bayonne. Recentely i'm experiencing some problems and i am planning to migrate to asterisk. I would like to know if i can do these things whit asterisk: - IVR integration with database (mysql, insert,delete,update,select) - TTS - record exploration (for example, check if some resources are available in the database, and list them to the user (via

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

Not sure if it's alright to talk about this here??? compiled the STUN server from Vovida on RedHat 7.3. Looks simple to configure. It isn't starting...it tries to for a long time and then just craps out. Here is my config:/etc/sysconfig/stund #!/bin/echo Not to execute. # Path to stund STUND=/usr/sbin/stund # Set the required args for STUND STUNDPRIMARYHOSTNAME=208.x.x.x # The hostname

Hi..... I have just successfully setup Asterisk with 2 Cisco 7940 phones (converted for SIP) and a SIP softphone on a W2K box.....and it all seems to work very well.....to those who wrote this software, it is really cool. Anyway, I am new to this software, and I have a lot of questions which I am hoping someone on the mailing list might be able to answer for me.....I am basically trying to

These taken tonight (9/22/2004) at the Expo and Reception Enjoy. http://photos.tropiano.org/gallery/astricon-2004 Lenny

Newt is a separate library from asterisk that is used to easily create a ncurses based program. If you want to see some examples of how to use newt, look under the 'zaptel' source for zttool.c and/or under the asterisk/astman source for astman.c Newt pretty easy to code off of and can make a quick & easy frontend for things. Check out the /usr/share/doc/<newt directory> for more

Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use 555@mainmenu and 666@mainmenu) for the Icon extensions. IAX softphone seems to work

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based

I want to start a community based voip network projcet and am thinkimg of using VOCAL and asterisk gateways..... my question is, has anyone bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or Asterisk all the way.........am expecting 1000 -> 5000 users.. your thoughts would be appreciated. _________________________________________________________________ Don't

I'm trying to get Asterisk to talk SIP to Vocal and so far have only managed to get it partially working. Calls in from Vocal are working fine but outbound calls aren't. In sip.conf I have: [ivv] secret=SECRET username=08452416761 host=sip.intervivo.net fromuser=08452416761 externip=mt104.dyndns.org nat=yes canreinvite=no reinvite=no notransfer=yes In extensions.conf I

Jean-Marc Valin wrote: >>I'm about to try to integrate SPEEX into the VOCAL project. > > > Good. Just a detail, but the correct spelling in "Speex". > > >>1) To encode, it appears I need an array of floats. If >>I am playing wav files, what is the best way to turn these >>into something speex can encode? > > > Speex version