similar to: www.openpbx.org

I live in New Brunswick Canada. The phone company is Aliant. When you set up business service here, you can go with either analog or digital lines. This isn't a T1 or ISDN. They are talking individual lines direct to handsets that they provide. They offer the digital option with even very small ( 2 - 4) number of lines. What technology could this be? Is there any way to connect such a

Digium X100P / new cards are is available on ebay for $43. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 <http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 > Hope this helps to who want to play with X100P! Are these being sold by Digium ? I don't know ?? - SamW -------------- next part -------------- An HTML

I have a problem with my asterisk@home installation (configured with AMP) My question is this, can you have different ports for different contexts within IAX? [Faktortel] port = 5036 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow=all ; Allow all codecs register => XXXXX:XXXXX@iax.faktotel.com/EXTEN

I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com

Any chance of say digits being extended to recognise "*" & "# " ?? Heck these are digits on a normal keypad :-) Gary .

My setup is as follows: Handset -> Sipura SPA 2000 -> Asterisk -> VoicePulse and Handset -> Sipura SPA 2000 -> Asterisk -> Digium X100P -> POTS I notice when making VoicePulse calls (but *not* POTS calls through the X100P) that there is significant "popping" and "clicking" on the line. This isn't enough to interfere seriously with the call, and

I configured agi-test.agi on extension 111 when i dial into asterisk extension 111 using a IAX softphone and hangup while the AGI is playing asterisk crashes. Does anyone have any idea why this happens. -- regards Vikram (http://www.vicramresearch.com)

I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700 number is dialed. I must be missing something obvious. Thanks -brian -------------- next part -------------- An HTML attachment was scrubbed...

I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 => 3213,Bill Smith Thanks! Paul Mahler mail:pmahler@signate.com phone: 650.207.9855 fax: 877.408.0105 -------------- next part -------------- An HTML attachment was

Dear All, . After installing, testing and like many other I found that Asterisk is reliable and a great Open source telephony solutions for a professional use. I would like to create a business with *, offering it as a IP PBX solution to customer, as a server, whatever with some Digium HW and SIP phone, .. etc. Asking for something that might have been asked. What are the implications for

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

Another trivial question: Is there a "place" where all the parameters are documented ? In example (my example!) I would like to know the meaning of a lot of parameter that can be used in sip.conf, A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060; context=xxxxx) but other are not (at least for me) i.e.: type = peer, friend insecure=very host=dynamic and so on.

Two cvs-head asterisk boxes with iax2 working fine (without register statements). When two calls are placed simultanously from system A -> B and the packets are sniffed on the wire, I see the two calls using two different udp packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes (at both ends). I was expecting to see both calls handled within a single udp packet, but

Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call 6010532c6fedf9be383872e07e4be70c@192.168.1.2 for seqno 102 (Request) I'm running asterisk with a Cisco 7960G If anyone know's why i'd get this.....Any help would be appreciated!

Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off

Hi, Is Asterisk (or can it be set up as) a SIP proxy? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

My apologies for the cross-posting. If you are a business or individual providing Voice over IP services in Canada then we encourage you to read this email carefully otherwise please disregard. ----- As you are most likely aware, the CRTC has undertaken the roll of regulating VoIP services in Canada and is currently conducting hearings with the goal of putting in place regulatory requirements

We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am a total virgin to ISDN. I understand that we need two BRI circuits to provide four voice channels, and that the hardware to speak to the BRI circuits can be passive or active, with the active type being much more

Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten => 125,1,Ringing exten => 125,2,Wait(3) exten => 125,3,Answer exten => 125,4,Wait(2) exten =>

This is slightly OT in that it isn't specifically *-related, but I was wondering what the members of the * user community felt about these two subjects. I've been perusing the OpenPBX.org mail list and the current hot topic is the fact that their project has come to a grinding halt. They are concerned that they don't have enough people working on their project. They feel that * has