similar to: Fair comparison

Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a

I have seen a number of people in this newsgroup asking for information regarding asterisk interworking with Vocal. I was able to configure Vocal and Asterisk so that calls originating from vocal can land on an extension in Asterisk. I would like to share this info with the group The scenario that I tested was as follows. A call was originated from extn. 1001 on Vocal and the call was made to

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

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

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 am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the

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'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

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

2012/6/21 Andr?s Gonz?lez <acandido at hi-iberia.es> > ** > On 20/06/12 15:01, Marc wrote: > > Hello List, > > > Hello Marc, > > > as an long time macintosh user , musican/producer/programmer , i am very > upset that another great technology (DSS ) vanished because of http > streaming so i turned my interest towards icecast, whitch seems an >

Hi all, I'm a student (my thesis work consist in testing interopearbility SIP/H.323) and I begin to work with asterisk in this days. I have to testing to SIP/H.323, since today I have used Vocal system, but there are some problem for this features. In the asterisk mailing list, in the next message I've seen an e-mail """ [Asterisk-Users] Cisco

Are there any other codecs that can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030723/34e950e0/attachment.htm

Hi, I'm trying to set up a "secure" VoIP channel between a Windows softphone client and an Asterisk 1.6... server running with OpenBSD. By "secure" I mean to prevent any man in the middle to reconstitute any vocal exchange nor sender/addressee/any header data/ of the VoIP call (in first step, I would be glad to secure vocal data ans see later for the header...) I had a

Boy, am I stuck... I'm officially ready to toss Asterisk out the window. I have to admit it isn't necessarily all the fault of Asterisk either. It just seems that every option I turn to suddenly ends in failure. I don't know if it's me that's bitten of more than I can chew with this project, or maybe Asterisk just isn't mature enough yet. Nothing complicated really....

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 am an experienced developer with Windows and familiar with Linux. I am looking for a SIP solution. 1) How does Asterisk compare to VOCAL in terms of support. 2) Is Asterisk free? 3) Where are the docs? Or even better. Where do I start? 4) Will it run on RH9? Thanks in advance. Costas -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Hi all, we face a problem with SMS reception sended to _landlines_, at least in France. Normally operators -tested with France Telecom and SFR- are sending voice SMSs from a particular CID number, so no problem. But today we discover that -at least SFR- send from time to time voice SMSs with original callerID which means that the call is terminated like a normal call and not recognized as

Hello List, as an long time macintosh user , musican/producer/programmer , i am very upset that another great technology (DSS ) vanished because of http streaming so i turned my interest towards icecast, whitch seems an fantastic and evolved media streaming server. I am very interested in Midi, especialy the possibility to *sync Audio with Midi*. So my question , would it be possible to stream a

Hello! I'm about to try to integrate SPEEX into the VOCAL project. If anyone has any pointers as to the best way to do this, please let me know. After reading the speex api man page, I have a few questions: 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? 2) Are there any commercially

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