similar to: Stable Relase Broken ?

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

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

-----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.

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

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

I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 => 3213,Bill Smith Thanks! Paul Mahler mail:pmahler@signate.com phone: 650.207.9855 fax: 877.408.0105 -------------- next part -------------- An HTML attachment was

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

I've been to the WIKI and I've searched the archives. Is anyone on the list successfully using iconnecthere behind NAT? I was, for over a year, and then I changed my "plan" with them. Now all my calls get intercepted immediately, "We're sorry, but your account is temporarily unavailable." Incoming calls work just fine. I contacted their so-called

Hello All, I am getting a set of errors when I boot Asterisk that I have not been able to solve. What is causing these error(s)? Asterisk boot output: ============== Asterisk CVS-04/10/04-21:44:51, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= [

When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start

I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables

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:

Greetings, Below is part of the contents of my extensions.conf file. exten => s,1,Wait,1 ; Wait a second before answering. exten => s,2,Answer exten => s,3,ResponseTimeout,10 ; Set the amount of time the user ; has to make a selection. exten => s,4,DigitTimeout,5

Firstly, let me just say I am new to asterisk and if anything I've said is covered in an FAQ or in previous posts I apologise but I have tried searching and I've attempted a few of the things I found but they didn't help. Has anybody got any experience using an X100P on an NTL phone line in the UK (I'm in an ex Cable & Wireless area if that makes any difference). The

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

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.

Hello, I am using asterisk-1.2 with chan_misdn and hava a usb-isdn adapter. Calling in and calling out works quite well, except that it is not possible to set the callerid. I have a connection with 4 telephone numbers (german ISDN), the problem is if I make a call to my mobile the basenumber is shown on my mobile as the source of the call. How can I set the phone number which is shown as the

Hi, Looks ok for native MIPS, I have two failures on debian8: Failing Tests (2): XRay-x86_64-linux :: TestCases/Linux/argv0-log-file-name.cc XRay-x86_64-linux :: TestCases/Linux/fixedsize-logging.cc I'll investigate these failures. Otherwise looks ok. I've uploaded the binaries. 9d5a389c20eb5b3071e6a0504b7cf87d clang+llvm-4.0.0-rc1-mipsel-linux-gnu.tar.xz

Hi All, Can I have the release plan of OCFS R1.0.15? Our big Customer, *** BANK, has hit BUG:4590449 (fixed in R1.0.15) and we need to show our ETA of R1.0.15 now. # Customer's system is running on RH 2.1 x86. Since OCFS R2.0 has already been released, I'm afraid that R1.0.15 will be released or not. (R1.0.14 is terminal release?) Any information will be helpful for us. Regards,

Hi All, Can I have the release plan of OCFS R1.0.15? Our big Customer, *** BANK, has hit BUG:4590449 (fixed in R1.0.15) and we need to show our ETA of R1.0.15 now. # Customer's system is running on RH 2.1 x86. Since OCFS R2.0 has already been released, I'm afraid that R1.0.15 will be released or not. (R1.0.14 is terminal release?) Any information will be helpful for us. Regards,