similar to: slightly OT: VoIP more expensive than Call-By-Call

We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users. Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM. SIP-Phones

HI, We need to get our number into the White Pages. Has anyone here actually tried it? Thanks, -- Leonard Burton, N9URK http://www.jiffyslides.com service at jiffyslides.com leonardburton at gmail.com "The prolonged evacuation would have dramatically affected the survivability of the occupants."

Hi, First off, let me state that _YES, I am fully aware that what I am doing is insane, prone to major havoc and bad for general health_ :-)) Scenario: My GF needs an analog modem to use with her banking software (sodding backs don't supply a decent web-application for company use). I am experimenting to see if we can get it to work (albeit slow) trough our ATA186 talking g711 to

I am looking for a voip provider that provides good rates (below 5 cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they do unlimited to NCFA but does not have the ability to actually termiate those calls as per the CTO Nathan Stratton, and last he said they dont even have contracts in place to get service provisioned for that. As such I am looking for another provider to take

Hi, Is there any default way to model something like the following situation in ActiveRecord?: A company has_many :buildings which are associated to :company, e.g. in a polymorphic way. Moreover one and only one of the buildings is the company''s headquarter. Thinking in terms of the database, I''d prefer to add an owning association, an thus have a has_many :buildings plus a

Good morning, I am trying to use the Windows iax client. My iax.conf looks like this: [general] port=5036 bindaddr=10.1.3.111 bandwidth=high allow=gsm ; Always allow GSM, it's cool :) tos=lowdelay [pos| type=friend context=default auth=plaintext secret=pos deny=0.0.0.0/0.0.0.0 permit=10.1.3.0/255.255.255.0 host=dynamic defaultip=10.1.3.2 In the registrations dialog

Hello, today I tried a DIAX -> * -> DIAX connection over the internet (768/128 ADSL connection on both sides). The sound quality was great. However, we had some latency problems, and also, if both sides where not talking the first words had some problems getting thru. Is this expected, is there anything that can be done on our setup, any magical iax.conf entry? Thanks and best regards

I have my asterisk box on the public network. I have a winders box on the public network, running diax. I have a winders box, same setup, behind my Linux iptables firewall, on a private network. Both boxes cann register iax2 to asterisk, and dial, but as soon as asterisk tries to do the native bridge, the call fails. I've poked at this way too much today... what ports do I need to open for

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

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The subject of tomorrow's VoIP Users Conference will be mobile VoIP. If you have any interest, please join us. I myself am tesing a bunch of iPod applications to use with all the usual suspects: OnSIP, Sipgate, Gizmo, Skype, your asterisk box, etc. Details for joining the call are are at http://VUC.me or http://VoipUsersConference.org IRC: #voip-users-conference on Freenode.net See you

asterisk-users at lists.digium.com Please excuse the ignorance of these questions. (I still have yet to install and configure a VoIP solution.) Depending on the feasibility of some of these questions, I was going to try my hand at installing something... --Is there a way to pick the best SIP service for each country? So, let's say I buy service from several companies, and one country is

Hello. I would like to know if somebody did a wireles voip with Asterisk PBX. I think to deploy a wireless for about 500 potential customers, it's a 3 km radius maximum coverage with houses without phone lines, I work for public places telephony small enterprises ( a common bussines in Spain) so I can get good rates from 4 telcos and do LCR at my asterisk PBX. Is anybody did this before

Hello I makeing a Domain Controller with Samba (v3.0.33) and LDAP (v2.4). I will install a PDC in the headquarter and a BDC in the subsidiary of the company that I work. The PDC and the BDC will have his own LDAP data base. I just install the PDC without problems and my next step is to install the BDC. I configured the LDAP that work in multi master mode. I made some test and the LDAP works

I''m building a 10 branch/1 headquarter network with Shorewall/Linux as gateway on all locations. The TI guy asked me if there is a way to ''cache'' TCP/UDP traffic between them. I crawled on Internet and I only find very expensive solutions for this. Some of them appeared in this comparison article: http://www.networkcomputing.com/showitem.jhtml?docid=1524f5 Does anyone

Hi, This is what I want to do: 1. Asterisk to answer calls via DID's, currently using SIPGATE 2. Provide a menu, and allow users to dial out. 3. According to the country and area they dial, the call should connect via one of up 4 carriers depending on cost. 4. If the carrier is busy it should go to the next one in line and so forth. I have tried to set this up, but it never answers the

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

Hi, I am just about to move out from my parents home and think about how I will phone from now on. In Germany there is a provider (QSC) who offers DSL (1024 down/256 up) with fastpath without volume or time limits. Does anybody know a comercial (or even semi-professional) provider who lets me dial out through H323 (or another protocol) and also offers an number where I can be called from

Hi people! I?m a fireman from S?o Paulo Fire Dapartment (Brasil) and I?m encharged to give support to the Computer System of the headquarter. We have for about 500 users connected in a fileserver running Red Hat 7.2, but we are authenticating the users using another machine running win NT. I would like to turn off the win NT but I don?t known how getting the users and passwords from the win NT and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, Asterisk currently uses the timestamps in incoming RTP packets to build outgoing voice frames. Is this true? Would it be possible for me to use i.e. zaprtc as a timing source for the outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on Ast1 because I don't trust the timestamps coming from