similar to: Debit/Credit Card Terminals

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

test only. good luck! james.zhu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/0fd2b827/attachment.htm

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

I need help to optimize a query (sort of ...) if params[:debit] == "on" @schools = School.find(:all, :conditions => ["name like ?", "%#{params[:search]}%"], :order => ''name'') @schools.delete_if { |s| s.debit <= 0 } @school_pages = Paginator.new self, @schools.length, 20 The problem here is that

Hello, While reading Dan North''s BDD tutorial <http://dannorth.net/introducing-bdd>, I tried to implement his ATM example as spec stubs. When I first implemented it creating a context for each of his scenarios, I noticed that there is duplication and a combinatorial explosion of the specs. I attached the full files to this email. For brevity, I will use scenario 1 in the body of

Hello, I just got off the phone with my PRI provider, and was told that I am not sending an expected message when I reject a call with a Cause Code for Unassigned(1) and Congestion (42). Busy works fine though... Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1, however the tech told me they expect a PROGRESS indicator with a value between 1 and 10. Any ideas on how

Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick

I don't suppose anyone might know why I hear ringing transposed over itself when I place a call out via PRI? SIP to SIP is fine SIP to IAX is fine SIP to PRI is always transposed I mean sometimes you don't notice it much because it's lined up right, but other times you'll hear a really long ring (starts sounding normal, then sounds "weird" -- like two rings played at

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike

I've seen lots about presence and Polycom phones recently. I've got one here for evaluation but noticed other phones only seem to appear busy when they initiate a call. If they receive a call, they still show as available. Is this a config problem on my part, or is that as far as presence is working right now? Thanks! Trev

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

Hello, I've been playing around with two generic X100P analog cards to create a proof-of-concept system before we go ahead and hook up a PRI. I'm running into a reproducible problem with sound quality of bridged calls, and am hoping someone will be able to point me in the right direction. I have in my dial plan a _9. extension so outgoing calls can be made... the first thing is

> Meanwhile banks like Chase charge poor people $12.00 a month just have > checking and push debit card paychecks on low income jobs where they > charge just for the poor to check how much they have on it. That bad, huh?

Hello, I know not too long ago I saw /something/ _somewhere_ about an adjustment to call parking that allowed blind transfers from SIP phones to park a call and still be able to hear the parking lot stall number. Unfortunately, I have no idea where I saw that (google turned up little, couldn't find it on the list either). I'm using Sipura SPA-2000 adapters and it doesn't seem to

Hello, I've got my dialplan configured to do a double ring when a customer service call comes in, and a normal ring when an extension is dialed directly. When a customer service call is transferred, I want to ring to revert back to normal. In the local extension macro, I have the following ; make sure ring is set to default exten => s,n,NoOp(${ALERT_INFO}) exten =>

We have a PRI from Telepacific. Asterisk 1.2 and a Sangoma A101 T1 card. Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but the calls are never answered. All other calls, and most toll-free numbers are not affected. The numbers that are affected are all travel related companies (United Airlines, American Airlines, US Air, Starwood Hotels, etc.) we cannot connect to

Can anyone recommend any wholesale SIP termination providers that will automatically charge a credit card? Most seem to want upfront payment and a credit balance but that sucks when you have to watch it like a hawk to make sure the balance never hits zero. I'm looking for a provider that can automatically charge a credit card. Douglas. -------------- next part -------------- An HTML

Hi Folks, Apologies to anyone who'd prefer not to see this query on this list; but I'm asking because it is probably the forum where I'm most likely to get a good answer! I'm interested in the provenance of the name "normal distribution" (for what I'd really prefer to call the "Gaussian" distribution). According to Wikipedia, "The name "normal

Hello, Has anyone tried the combination of asterisk 1.2.24, zaptel 1.4.5 and HPEC 9.00.003? In particular, with a hardware configuration similar to: Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) I have two fully independent systems

Hi, I am new to this group. I use this script tc qdisc add dev eth0 root handle 1: htb tc class add dev eth0 parent 1:0 classid 1:1 htb rate 500kbit ceil 500kbit tc class add dev eth0 parent 1:1 classid 1:2 htb rate 300Kbit ceil 500kbit tc class add dev eth0 parent 1:1 classid 1:3 htb rate 200kbit ceil 500kbit I like to know: If two customers of the same class (for example 1:2) work