similar to: * with external sip proxy

Greetings, I am progressing well with this great product, the *. SIP to SIP calling, Vocal to *, Voicemail all in the past. Did Iconnect, FWD etc. Also, purchased couple of FXO cards and did zaptel as well. It's time to get to h323 now. Read the mailing list for H323 and OH323 etc. need some help to where to start. Requirement is very simple, SIP calls need to be routed to a third party

Hi all, It has been a year or so since I have run this code to plot temporal activity. It was working now I am getting an error related to MUTATE. Error in UseMethod("mutate") : ? no applicable method for 'mutate' applied to an object of class "character" Any help/suggestions welcomed. Tnx to all the R code gurus out there. R v 4.4.0 Bat Dude The saved code I

Hi all, I would like to set up an iterative read & write sequence to avoid reading and writing each file one at a time. Hundreds of data sets to re-calculate.? The code I have works well individually, but would like to set up an iterative read, calculate and write changing the input and output file names each iteration. I? think I have read that there is an R? feature using

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

Hi there! I use an gnome-terminal for using R. When I resize the termial to the maximum size, R uses only the left side of the window. Can I tell R to use the whole window somehow? Thanks, Martin -- Ihr Partner f?r Webdesign, Webapplikationen und Webspace. http://www.roomandspace.com/ Martin Kaffanke +43 650 4514224 -------------- next part -------------- A non-text attachment was

Hi Rowland, i've modified, the dns like below: RODC-1: - themself - DC-1 - DC-2 DC-1 And DC-2 dns configuration not modified But the problems remains - samba-tool drs replicate rodc-1 dc-2 dc=scratch,dc=lan -U administrator did not replicate - network browsing anyway require authentication and not work Enrico Manzini -----Messaggio originale----- Da: samba <samba-bounces at

On Fri, 3 Jan 2025 08:29:59 +0000 Manzini Enrico <emanzini at zensistemi.com> wrote: > Hi Rowland, below, the servers and the remote client dns configuration > > Server's dns configuration: > DC-1: > - themself > - DC-2 > > DC-2 > - themself > - DC-1 > > RODC-1 > - DC-1 > - DC-2 > - themself > In my opinion, all Samba AD DCs

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.

Hi Rowland, below, the servers and the remote client dns configuration Server's dns configuration: DC-1: - themself - DC-2 DC-2 - themself - DC-1 RODC-1 - DC-1 - DC-2 - themself REMOTE-CLIENT - directly point as dns to the RODC-1 ip - nltest /dsgetdc:domain_name report RODC-1 as logon server DNS SRV RECORDS: The ad srv records uses default configuration Enrico Manzini

All, I'm running (attempting to) ooh323 with Asterisk and a Cisco 2621XM router operating as a H.323 GateKeeper, however when I bring the Asterisk box up it registers successfully with the GateKeeper (exchanges GRQ/GCF, then RRQ/RCF) it notes the GateKeeper supports keepalive at 300 seconds, when it gets to time to re-register its sends an RRQ again and gets rejected with RRJ (unspecified

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

Has anyone had any luck getting Asterisk 1.6.2.13 to register to a Tandberg Gatekeeper? The logs on the Asterisk end seem to show that the registration request is sent, and the Tandberg Gatekeeper responds. However, the response doesn't seem to be what Asterisk was expecting. Here is my ooh323.conf, followed by the relevant portion of the h323_log: [general] port = 1720 bindaddr =

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

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form

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 have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - AVS@210.21.118.XXX Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn-xxxx e.164 - 22xx2912

Stephan, Ok, I'll re-state the problem... I have two devices that I want to talk to each other: 1. an Asterisk PBX 2. a Damm Cellular TETRAFLEX digital radio system (www.damm.dk) both devices are effectively "gateways" because they have many subscribers behind them. The Damm Cellular system controller is based on Windows-XP Embedded and its sub-systems used the OpenH323

I am looking for an app that would run from the terminal and would emulate a bash shell (or pass everything to the shell) that would allow me to set a log file and then record all my input and the output to the screen from the commands. As an added bonus if it would allow me to run it from two terminals (or more) on the same machine and log all the input and output to the same file while still

All, I've fixed my problem getting Asterisk ooh323 channel to stay registered with my Cisco IOPS gatekeeper, now I need to get dialling working. I have the following: [Asterisk with ooh323] ----h323---- [Cisco IOS GK] ----h323---- [Radio system OpenH323] 192.168.1.5 192.168.1.6 192.168.1.7 the Asterisk box has numbers