similar to: Telus: Overseas calling

We are receiving phone-calls at one Asterisk box, it forwards them to a different location over IAX. Calls are answered on the other end of this connection. However, the callerID seems to be lost in the process. The 1st box shows the original callerID, on the 2nd box callerID shows the callerID of the 1st box. Apparently the 1st Asterisk box replaces the original callerID with its own. Is

We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. The technician was stumped. I suspect there's something specific that I need to do to make it work, since many PBXs can do this. The switch is a

>I called Telus before Christmas requesting some sort of VOIP connection. >We are going with babytel. I'll advise how that works when it is up and >running, hopefully next week. [plug] www.thinktel.ca I know the guys they are competent they will sell IAX. Peered thru GT in Downtown Edmonton.

Hello All, As you can see below I am having a problem checking for updates. This happens repeatedly. I have to kill the process then rerun. I have tried "yum clean all" but no joy - the process hangs again on "extras" - see second listing below. Suggestions? Dave [root at cserver ~]# yum check-update Loaded plugins: fastestmirror, priorities Loading mirror speeds from

I will be out of the office starting 12/09/2005 and will not return until 16/09/2005. Dear Sir / Mdm, I'm currently on course and are not in office. During this period of time, I have minimal access to internet and email cccess. As such, I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. In the meantime, for any technical assitance, please

Hi, This is our test setup: 4 phones, 2 logged into the one queue, the other 2 phones are used to dial into the queue. If there are 2 calls coming into the queue at the same time, we would like to have the 2 queue phones ringing at the same time (one for each call). But as it is, the 2nd phone only starts ringing after the first one was picked up. Scenario: One of the non-queue phones dials

Config: AAH 2.2 Digium TDM card connecting to 3 x Telus POTS lines Polycom 501 phones pretty basic setup, working mostly just fine... When I dial a number such as: 96045551212 Telus automation will sometimes come online and tell me that the number I have dialled cannot be completed as dialled. If I hang up the Polycom 501 and redial the EXACT same number, it will work the second time. I

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

I just got the list confirmation and noticed it''s text only email so here it is again in plain text. Below is the oringal message. Hi all, I am really struggling with this one, I have built a lot of linux machines using IPSEC tunnels and shorewall gateways. I decied to build a new test machine with Debian running 2.4.25 and Shorewall 2.0.15. I have two subnets on their own switches and

Annoying :( S. -----Original Message----- From: Mail Delivery System [mailto:postmaster@bembang.com] Sent: Thursday, September 02, 2004 3:06 AM To: stormp@telus.net Subject: Mail delivery failed: returning message to sender This message was created automatically by mail delivery software (HiveMail). A message that you sent could not be delivered to one of its recipients: sysad The error

created 2 partitions on laptop (/ and swap) formatted them. mounted both source and target (under /mnt on each machine) rsync -a / from desktop -> laptop (Note: OS on desktop is on sdb5. On laptop it's going to sda1) Now I need to put grub on mbr of laptop boot: linux rescue chroot /mnt/sysimage Here is my problem grub-install /dev/sda The file /boot/grub/stage1 not read correctly. Can

Hi all, First, this is not my first PRI/T1 Asterisk deployement. Did several with Bell, Telus, AllStream, Rogers but this is my first with Videotron. Just spoke with the person taking the order and on top of the standard settings (switch, coding,...) she asked me about data rate (56k or 64k). Since I have never been asked this question before and can find anything relevant in the

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

I thought I sent this earlier this week, but I didn't see it. If I missed it, I apologize for the resend. We are running asterisk 1.2.2 with a TDM04B connected to PSTN lines. On incoming calls from cellphones located overseas, DTMF is not recognized - we have many single-digit choices in our menu so the problem isn't that some digits aren't working, it's not listening at

We have PSTN lines connected to FXO lines of a TDM400B. I just got a complaint that overseas callers who are using cellphones sometimes find that DTMF digits aren't working - they press digits and the menu goes on as if they hadn't pressed anything. Since it sometimes works, and other IVRs work over the same cellphones, it's not that the cellphone isn't sending the digits.

Im trying to figure out the base way to check the callerID being sent to my Asterisk box and use it if it is a valid NANP number, but replace it with a static NANP number if it is not. (Why? I have a few carriers that require this, and a few international users - if it happens to take one of the carriers that require it, I want it to set a static number that is valid). I'm playing

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Hello -- I desperately want to start using Ogg compression on music I release on the internet, but I am encountering a fair amount of artifacts. I noticed some in beta4 and was hoping they would disappear with RC1. However, I've been playing with RC2 and I've still been banging my head into a number of problems. A majority of the music I make has long drones with a fair amount of spectral

Hi all... Does anyone know if it is possible to override sip.conf settings in extensions.conf (for example: session-minse=90) without needing to create an overarching peer in sip.conf and selecting it specifically in the dial plan? I'm on the 1.4 stable code base and looking to implement session-timers on certain call flows in a modular dial plan. Thanks, Josh Fuller josh.fuller at