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So a guy shows up at the the office, after making an appointment with the office manager / receptionist to talk 'phone systems'. After her eyes glaze over, with him talking T1 and Frame-Relay I get to see him. He's from Norvergence. Well dressed. Tells me they can do a T1 for $79, with unlimited local & long distance for free. It also does 'internet'. 'Just give me

Don & others, Thank you for your answer. The fog maybe lifting ;). The zaptel.conf file has the following in its comments: # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a

...t; _5.,1,Dial(IAX2/iax-a2:secret@a1.mystrx.com /${EXTEN}@inbound-calls So when you will dial anything beginning with 5, the call will be dialed in the context inbound-calls of a1.mystrx.com -----asterisk-users-admin@lists.digium.com a ?crit : ----- Pour: asterisk-users@lists.digium.com De: willy@yponeinc.com Envoy? par: asterisk-users-admin@lists.digium.com Date: 29-04-2004 22:59 Objet: [Asterisk-Users] IAX Example Needed Hi All! I have two [*]s, and both work OK as a simple local PBX. Now, I try to link them using IAX. Let's call those babies a1 and a2. From a1, I want to dial a phone connect...

All, When calling an invalid number using, I expect to hear: "dooh-deeh-daah We're sorry you have reached a number which has been disconnected ..." And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I continually hear ringing, and then the call gets hungup. Any ideas

All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing

To the list ... I got the IAX2 stuff simplified & working (for now). See my earlier posting to the list. Now, here's a question for you all. I found a posting by J Todd where he gives BW utilization for various IAX2 codecs with trunking on. Now, the number of calls I can sustain over an IAX channel, obviously is going to be determined by the capacity and state of the physical pipe.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. I'm getting the following error at random intervals on my TE410P with Asterisk CVS-03/30/04-11:49:01-CEST. I have two spans active, one connected to my Telco, the other to a Siemens PABX. Both spans display this behavior at random intervals. All calls are dropped when this happens. Spans are not necessarily in use when this happens.

My PRI is being reset at least once a day with the following errors in the logs. zaptel, zapata, libpri, and asterisk are from CVS this morning.. This has been happening for weeks on all versions (including -stable). the T100P card appears to NOT be sharing an IRQ. xenon# cat /proc/interupts CPU0 0: 1203977 XT-PIC timer 1: 3 XT-PIC keyboard 2:

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14

Hi all .. Maybe I am just missing something, but when I press '0' and then '3' to record my name, it gives me an 'after the tone .. '. Then I say my name and press #. It says: your message has been saved. So HOW do I listen to my recording, make sure it sounds OK, and then CONFIRM that is what I want the mailsystem to use OR CANCEL out and leave things the way they are?

Hi All! I have two [*]s, and both work OK as a simple local PBX. Now, I try to link them using IAX. Let's call those babies a1 and a2. From a1, I want to dial a phone connected to a2. Both boxes have a fixed IP address, and use standard port 5036, say a1.mystrx.com and a2.mystrx.com. Where I (obviously) get confused, is when it comes to inbound, outbound, registration, etc. Taking a hint

Hi Akk, If this has been discussed/done then apologies be-4-hand. I did not find it in the Wiki or the Archives. Here's the question. We have incoming PRI lines, all on the same main number. An attendant is supposed to handle all incoming calls. Now, let's say I have a multi-line SIP phone. For argument's sake (and to keep it simple) say I only have two lines. We'll call them