search for: netvoice

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

...al: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 (a) What is causing this? (b) How can it be fixed? (c) Why does Asterisk not recover automatically to what appears to be an intermittent problem? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)

Greetings, I am trying to get either of the above features to work with *, but can't seem to get it quite right. If anyone has them working, I'd sure appreciate an extract from the relevant config files. Or, maybe I'm tilting at windmills, and * doesn't support them - in which case, the underlying business need is to provide the one incoming call on more than one

...antronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007." http://news.com.com/8301-10784_3-9719931-7.html?part=rss&subj=news&tag=2547-1_3-0-20 -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)

...tgoing directory until we get a winner, throw the willing called party into the same conference. (2) Park the incoming call, make the outgoing calls, transfer the willing called party to the parked call extension (not sure this will work but?)? What is the quality solution? -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I'm looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using

I'm looking at switching VOIP providers, but want to ensure I move to a company with sufficient capacity. Can any Canadian VOIP users post/email me with feedback on their providers? I'll post the results for all to read...... -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am looking for some Linksys and GrandStream ATAs in Canada. I am looking for places that ship from Canada so I don't have to deal with the clearing of customs and tax remittance. Any suggestion? -- Thanks

Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be

I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used for data and VOIP, do they put the asterisk machine behind a firewall or do they put the firewall on the asteri...

Does anyone know how to change the source IP address/Source Interface of RTP packets? Changing the SIP source IP address in sip.conf has no apparent impact on RTP. RTP traffic still uses the address assigned to the outbound interface.

My question pertains to the use of IAE.. I would like to setup 2 Asterisk boxes. One would be located in our office behind the firewall and hooked up to our analog lines. The other would be located in a remote datacenter and used for our remote employees to connect to. I would like to be able to accept calls on the Office Asterisk server and route them to the Datacenter Asterisk server. Is

...m aware that one could use AGI or an external SQL database for such data sharing; I would just prefer to avoid such overhead or complications in this situation. One could even envision making this a configuration option (i.e. "astdb = shared"). Thoughts and flames please. George Pajari netVOICE communications www.netvoice.ca www.ip-centrex.ca

...ble branch builds with the same pwlib code but of course the h323 code in the stable branch doesn't work. So it seems those of us who need H.323 have too choices: (a) code the compiles but does not work (b) code that is reputed to work but does not build. Any suggestions? -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandsteam.ca www.sipura.ca www.snom.ca

>Date: Mon, 07 Feb 2005 02:22:07 -0800 >From: George Pajari <George.Pajari@netVOICE.ca> >Subject: [Asterisk-Users] Remote MWI via IAX? >To: Asterisk Users Mailing List - Non-Commercial Discussion. > <asterisk-users@lists.digium.com> >Message-ID: <4207414F.10807@netVOICE.ca> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed> > >We h...

Has anyone done interop testing with Level 3 and Asterisk. If so, would you be willing to share your experiences. Gene -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050308/061aea35/attachment.htm

...13 08:37:16 DEBUG[8480]: Bridge stops bridging channels Zap/1-1 and SIP/cronus-116-78ed May 13 08:37:16 DEBUG[8480]: update_user_counter(cronus-116) - decrement outUse counter May 13 08:37:16 DEBUG[8480]: Exiting with DIALSTATUS=ANSWER. May 13 08:37:16 VERBOSE[8480]: == Spawn extension (macro-netvoice-stdexten, s, 302) exited non-zero on 'Zap/1-1' in macro 'netvoice-stdexten May 13 08:37:16 VERBOSE[8480]: == Spawn extension (main-menu, 116, 1) exited non-zero on 'Zap/1-1' May 13 08:37:16 DEBUG[8480]: Hangup: channel: 1 index = 0, normal = 21, callwait = -1, thirdcall = -...

...13 08:37:16 DEBUG[8480]: Bridge stops bridging channels Zap/1-1 and SIP/cronus-116-78ed May 13 08:37:16 DEBUG[8480]: update_user_counter(cronus-116) - decrement outUse counter May 13 08:37:16 DEBUG[8480]: Exiting with DIALSTATUS=ANSWER. May 13 08:37:16 VERBOSE[8480]: == Spawn extension (macro-netvoice-stdexten, s, 302) exited non-zero on 'Zap/1-1' in macro 'netvoice-stdexten May 13 08:37:16 VERBOSE[8480]: == Spawn extension (main-menu, 116, 1) exited non-zero on 'Zap/1-1' May 13 08:37:16 DEBUG[8480]: Hangup: channel: 1 index = 0, normal = 21, callwait = -1, thirdcall = -...

...arms. Audio from Asterisk (i.e. playback of pre-recorded sounds is fine). Audio to Asterisk (i.e. sounds and DTMF from people calling Asterisk over the E1) sound as if played back on a tape recorder running at half-speed -- slow and down an octave. What could be causing this? -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

...0% DTMF recognition. Anyone with experience with these units have any suggestions? ABP Technical Support has been unable to diagnose the problem and is now sending random guesses and requests to try unrelated things in a desperate attempt to fix the problem, all to no avail. -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca