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hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

I have Spandsp working fine. Asterisk sees a fax on the zap port and redirects the call to the fax-in area. This works if I have a simple dialing rules that goes answers first and waits 10 secs then goes to the next item. If it hears a fax it goes to the right place. Here is a sample that works. [incoming] exten => 2019,1,Goto(test,s,1) [test] exten => s,1,answer exten => s,2,wait(5)

I think you will find the functionality you are looking for is in * already. Here is an excerpt from the sample extensions.conf file that is included with the source: exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten

I don?t know if the last message was with content. So, I sent again. I have installed a Digium card TE210P and unicall for use MFC/R2. I think that it?s all right but I can?t make and receive calls. I?m using asterisk 2.1 with the patch made by Jos? P. Leit?o and the follow libs: libsupertone-0.0.2 libunicall-0.0.3 libmfcr2-0.0.3 zaptel 2.1 My number is 34318300. The Telco send me only 8300.

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

If receive a call on the IP Dialog SipTone II, and the other end hangs up first, the siptone immediately enters into the congestion tone. If I initiate the call from the siptone and the other end hangs up first, same thing -- congestion. The same thing happens if we make calls from the analog phones attached to the Mediatrix 1102. This does not happen on our Snom 200 phones, which have

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an

We have quadBRI configured 1 port in TE mode 2,3,4 ports in NE mode. We are trying to place a call from the phone connected to BRI card port #4 to city number through ISDN line connected to port #1. Number successfully dialed. Person on the other end answering the line. But conversation can't last more then 10 seconds. Below is a log of such call. Its not clear for me why we appear in

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

I'm trying to set up my extensions.conf file using some of the existing macros like stdexten, etc. while at the same time having two logically separate virtual PBX's (with no "default" context) and two trunks coming into separate contexts, i.e. one for residence and one for my at-home business. I noticed, however, that macro-stdexten depends on the "default" context:

SIP SECURITY WARNING Version: v1-0 (cvs today) Problem: sip context in general section ignored - goes to default - allowing unauthorized sip devices to place calls in default context Fix [workaround]: Remove or rename "default" context in extensions.conf Notes: I am not sure what other asterisk functionality may be affected by this - review your other config

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

I've been banging my head on a brick wall for about an hour now trying to understand why the following doesn't work (which is even provided as an example in the distribution!). The goal is to create a voicemail-only extension not associated with a phone. I'd rather not have an extension dedicated to VoicemailMain(), so I would like the user to be able to hit '*' during

I have a pretty standard setup with Asterisk acting as a PABX for a bunch of SIP handsets (in this case, SwissVoice IP10S). My users are complaining that when they forward their phones to their cellphones on unavailable (i.e. forward when no-answer), their cellphone only rings once or twice, and then Asterisk sends the call through to Voicemail. I'm using the standard extension Macro

I am using the sample (slightly modified) macro for dialing phones. My extensions are in the 2000 range. Each extension has it's own external DID. Is there any way to have the macro look up the DID to be used for the extension and set the DID as the callerid? Maybe from a flat file somewhere? Or is there a better way to do this??? I know I can use callerid in sip.conf, but I only want the

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

Hi all, Having fixed my problems with the call waiting ringing on the GS phones, I needed to extend that with a campon facility (available on some legacy systems - sort of callwaiting without phone ringing). I've managed to implement that by adding/modifying app_queue.c. Basically, when calling the SIP phone, and if busy, I can camp the call onto that extension, with MOH, allowing the caller