similar to: Info sip/h.323 interoperability

Hello All, Finally I realised that the Asterisk demo setup didn't include support for h323. (Maybe it should have been obvious) so I went to work out how to get the h323 channel running. I had openh323 and pwlib installed as I'd been playing with vocal so it didn't take long to do cd asterisk/channels/h323; make; make install; make samples, copy the pwlib and h323 libraries to

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

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.

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

Hi, ( As a total newbie to R ) I get this error when attempting to use read.table to read a data frame: "Error in type.convert(x, na.strings, as.is) : null string encountered" (Is there help documentation for the type.convert function ( I could not find it in the usual places) This does not always happen, but seems to occur when I am attempting to read data frames with more than a

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

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Shaun Ewing > Sent: Tuesday, November 04, 2003 7:15 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Anyone using * in a live > production environment? > [...] > The additional features are nice too.

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

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

Are reqional settings supported or planned to be supported in wine?. Attempting to use the application "Vantive" written by the Vantive corporation. Attempting to run the application without any windows code. Exported the windows registry using regedit to userdef.reg. Exported code 'disappears' from this file. Wine Version: 0.0.20010824-1 Linux kernel 2.4.6, Debian Unstable

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, This is my first post here - I started with Asterisk a few days ago and have "fallen in love" - fantastic product. I've only got softphones connected at the moment - I'll probably order the FXO/FXS cards in about a month (and then think about getting some hardware SIP phones). Our current phone system is quite a few years old and isn't growing with us (when a single

Hello, all, I had been using NUT's Ubuntu package (64 bit) with no problems until upgrading to Ubuntu 11.04. Now, I get the following message when I try usbhid-ups: $ sudo /lib/nut/usbhid-ups -a mybelkin Network UPS Tools - Generic HID driver 0.35 (2.6.0) USB communication driver 0.31 Can't claim USB device [050d:1100]: could not detach kernel driver from interface 0: Operation not

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

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