similar to: Configuration details for Asterisk interaction with Vocal

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

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'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 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! 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

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

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

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

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.

> 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 1.0.x (stable) expects float's in the -32768 to +32767 range, so it's just

Hi folks, To provide MWI, * will send out a sip:notify message to the UA. The originator of this message is asterisk, as shown below; NOTIFY sip:1001@www.mysipproxy.com:5065 SIP/2.0 Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21 From: "asterisk" <sip:asterisk@66.121.xxx.yyy>;tag=as0ffb1bdc <=============== To: <sip:1001@www.mysipproxy.com:5065> Contact:

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

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 >

we have an SUN E450 server with soalris 2.6 and 512mb RAM running samba. we have a network and all the machines are connect through a switch. there are about 40 users who simultaneously use the server running samba for file service. The set up is like this. there is an application on the server which is shared and all the users have mapped their drive letter to that share. this application when

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

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

I was always puzzled by posts to the list about people having problems getting hints to work on a Snom, since I always seem to have no problem making it work. That is, until today when I tried to get a sidecar to work. All I could do was get a monitored extension light to light up continuously, regardless of state. Frustrating! Going back to my working dialplans where I got 1 or 2 lights working

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 --

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 have a question. I set up an extension to 1234 exten => 1234,1,Dial(SIP/1234@sip.greentone.com:5060) And when I dial that extension I got in SIP message "403 FORBIDDEN" Can somebody tell me why I cannot call that extension? When I am not using Asterisk I can call that extension without any problems. My SIP proxy is VOCAL. I am new here so I do not know a lot yet. Thank