similar to: Number forwarding and porting?

What is the deal with "CSR's"? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for

Hi, All How do I install Zaptel drivers on a system running Suse? Make results: grep: /lib/modules/2.6.16.13-4-smp/build/include/linux/autoconf.h: No existe el fichero o el directorio make[1]: Entering directory /usr/src/asterisk/zaptel/zaptel-1.4.0-beta2 make -C /lib/modules/2.6.16.13-4-smp/build SUBDIRS=/usr/src/asterisk/zaptel/zaptel-1.4.0-beta2 modules make[2]: Entering directory

Hi list, This is not necessarily an asterisk issue, but a lot of you guys know way more then me, so I have a question: someone at my company sets his phone to forward calls to his cellphone, so someone calls our office, call is forwarded to his cell, and the callerID that shows up on his cell is of course our office number, because asterisk originates a new call to his cell and then bridges

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

Hello, We currently have a scenario where a large percentage of inbound calls on a telco PRI are intended for professors who aren't currently in their offices/at their desks. My question is, is anyone actively using the Asterisk "RLT" (Release Link Trunking) feature to bounce these sorts of calls back to the telco? The idea being to forward the call to their mobile phone without

Well, this is weird. After receiving a sip subscribe message from peer 2529266, here's what Asterisk responds with: -- (14 headers 0 lines)--- Found user '2529266' Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com) Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'bell_CallStart' Transmitting (no NAT) to

Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. On the adept line I got a digium card in an opteron supermicro server. ztcfg gives me over 99.99 pretty much all the time.

I have a Debian 8 64-bit machine set up as a server and apt-got the tinc package. I configured tinc as a bridge and everything seems normal except that the tunnel does not forward multicast traffic. I used tcpdump to examine the br0, eth0 and tap interfaces. I could see multicast packets on both br0 and eth0, but there is no such packet present on the tap interface. I don't quite know why

Hi all, Has anyone implemented trunk to trunk forwarding with an asterisk PBX. For the purpose I have in mind its quite important that once the call has been sent onwards to the new desination the lines into the PBX are no longer held. If anyone has UK-specific experience of getting this up and running that would be incredibly useful! Nic

I checked out the latest merb ''trunk'' and after installing it as a gem... (fresh install) I got the dreaded "fastthread not loaded: fastthread must be required before thread" error when starting up an instance of merb. Even without dRB. So I patched the merb.rb file to include fastthread before anything else if it''s installed. Check it. Merry Christmas!

Hi, I'm very new to Asterisk and I have the following scenario. 1. Let's say I have a number of 1-222-222-2222 from my SIP service provider (VoicePulse). 2. I point my phone, Verizon wireless cellphone (1-111-111-1111), voicemail to the number provided by SIP service provider (1-222-222-2222). 3. I use another phone (1-333-333-333) to call 1-111-111-1111 and leave a voicemail message.

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

Hi, hm at seneca.muc.de schrieb am 29.05.03 21:40:34: > I would like to load pxelinux as a second stage bootloader from a machine > which does not have a PXE BIOS (namely, from a VMware guest machine) in > order to perform an automated Linux installation. A boot floppy would > be nice. It seems that there are some commercial solutions around but > is there someting in the OSS

Hi, I'm using linux 2.4.20 here and noticed that even when STP is disabled for a bridge, the forwarding delay still takes affect. Is this correct? I.E. Does the bridge need to spend time in the "listening" and "learning" states when STP isn't used? It's an easy work around to set the delay to 0, I'm just wondering whether it's a buglet or by design.

Hi, Incoming calls to one of my lines are set to ring two internal lines and simultaneously start ringing my cell phone. Something like this: exten => s,1,Dial(SIP/2201&SIP/2202&IAX2/my_cell at carrier),90) The internal lines 2201 and 2202 will both see the callerID for the incoming call, but my cell phone will show the callerID for asterisk, not the calling party. What's the

Is there a configuration file in asterisk used for modifying Daylight savings time or is this strictly being dictated by linux? ________________________________ Mark Galley * Systems Administrator acgroup <http://www.bflo.com/> Aurora Consulting Group, Inc. * 7625 Seneca Street * East Aurora, NY 14052 Phone 716.655.9000 X118 * Cell 716.238.3535 * Fax 716.655.4957 * www.BFLO.com

We have a number of DID's that do the standard VoIP tricks: ringing multiple locations, findme-followme etc. What is happening more and more is that customers call those DID numbers, and draw the reasonable conclusion that they are calling mobile numbers because they literally can HEAR that the called party is on a mobile. Consequently many of those customers draw the conclusion that they

Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. I was also thinking maybe people present could contribute some information on this list for now. The countries I am after are below. To start this off I will provide the information for Australia +61 and New Zealand +64. NZ Cellular: area code 21