similar to: IAX provider in Colorado

Hello, I am trying to come up with a good way to enforce a limit on the number of simultaneous calls that can occupy an IAX trunk at any given time. I have searched around and so far can't locate a config option that would directly label a IAX trunk with a specific number to obey (is there one?). Based on examples for the SetGroup and CheckGroup commands, I am thinking of using SetGroup

I am pretty sure that there are no IAX providers that offer CallerID name but wanted to double check with the list in case something has changed recently. Is anyone aware of an IAX provider that offers incoming CallerID name? Is there a technical limitation within IAX which is preventing IAX providers from offering CallerID Name? Why is no one offering this? -------------- next part

Ok, ok, I know there has been plenty of discussion on asterisk and fax - from this I understand: 1) First and foremost, use g.711 ulaw 2) Packet loss, etc...makes faxing over the internet unreliable My need is for a fax to come in on a X100P and be forwarded to a fax machine on the local lan. I don't currently have any fxs as I'm using all sip phones at this point. I see the

Grasping at straws here...is anyone using a Dell PowerConnect 2224 24-port unmanaged 10/100 switch in a deployment? I have two separate asterisk installations with bad one-way audio where the only common elements left are the Dell switches and Polycom IP-500 phones. Two different ITSPs - one location uses IAX2 to the outside world, the other SIP. Two different bandwidth providers, one SDSL 1.5

> I too contacted CDW about the $9.37 Cisco support > contract. But because I did not buy my phone from them I was > not allowed to purchase it. The vendor I bought the phone > from does not provide them. What are the "magic words" to > get CDW to sell it to you? With all of this hassle I highly > doubt that I will buy more Cisco phones anyway. After >

Since upgrading to 7.2, I've noticed a random problem where I dial a number and hear all the correct tones in the handset, but the display won't show all the numbers I dialed. So you sit there waiting for the dialplan to kick the call off (b/c you heard the proper amount of tones played and think it's all good) but the phone is just sitting there b/c it somehow "missed"

Hi Yall, I would love to put a very compact phone on my wife's desk at work... Ideally this would be a very small IAX phone with 2 RJ-45's so I could drop it in without much notice and only have to beg for 1 port from the sysadmin. I have looked around and I don't see such an item in existence? What I would love is a "home" phone looking handset (wired is fine), which

Can anyone help me to understand what the significance of this output is? May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4 May 17 10:50:23 DEBUG[2030]: Bridge stops bridging channels SIP/105-1ae4 and SIP/outbound-7dc3 I searched for these phrases but am coming up short on what they really mean. I'm trying to investigate problems we are having with two

Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone:

Hello, I'm looking for any comments or user experiences from anyone who is using 7912G phones with SIP. Any installation issues? Usability problems? Do the features seem to work, etc...In short, I'm looking for your opinions on how suitable this phone is for an asterisk implementation for approx. 10 users. Next logical question: what other phones would you recommend for a situation

I am shopping around for an IAX provider that provides both outbound minutes and an inbound DID for a flat fee. I have been looking at TELIAX. http://teliax.com/ Does anyone have any experience with them? What codecs do they support? Inband or 2833 dtmf? Voice quality and reliability? Any comments and/or other recommendations would be appreciated. Thanks

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

Hello: I have read that music on hold requires a timing source (which I never had to worry about previously since the server had zaptel hardware in it)...now I'm configuring a server in a colo which has no zaptel hardware. If I use the dialplan to run MusicOnHold(), I do get the music upon dialling that extension, but if I try to use the hold button on either a 7960 or X-Lite I get

I know that the wiki has an extensive list of European VoIP providers out there....but there's so many that it's kind of hard to sort through. So I was wondering if anyone could recommend some reliable SIP/IAX termination providers in Europe? Something like VoicePulse Connect, NuFone, Vitelity, or Junction Networks based out of Europe. I really don't trust a US VoIP company for

I was looking at the sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me <http://www.voip-info.org/wiki-Asterisk+Tips+follow+me> ) which uses a dial modifier 'c' to enable Answer confirmation - "If the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user

We have been, unsuccessful, trying to implement inter-asterisk communications using IAX. I was wondering if anyone is having similar problems, or would share information about their succes/ Our environment consists of two servers with public IP addresses and DNS entries that resolve. (for the example we use "domain.net") Based on the information available to us, we have tried the

I have an account with a IAX service provider that I'm happy with but recently I started getting rather strange reports from users. They're saying that occasionally, they'll be on a call via the provider and the outbound audio appears to slowly fade out to nothing with a bit of static during the fade. Does this ring any bells with someone? I'm going to update * on the server

Does anyone have any experience with any SIP or IAX providers that support E911? I'd love to convert entirely to Asterisk at my house, but the lack of emergency dialing has been a major hold-up for me. Thanks in advance for any suggestions! -- Kyle Sexton

Hello everyone, I playing around with ENUM and have configured * to query a few sources for testing purposes (fierymoon, e164.arpa, e164.org). I'd like to know if there is a way to query these servers manually (ie outside of asterisk via nslookup or equivalent) to find out if particular exchanges are listed with wildcards, so as to terminate calls to those prefixes (I'm not trying to

Hi, I have been trying to work this out and haven't been able to. I have some incoming numbers that come in over IAX, from the same server, and wish to use different codecs for different calls. This doesn't seem to work for incoming either. I cant seem to get any combination of allow/disallow to work.. Ideally the following would work: [general] register => XXXXXX disallow=all