similar to: Re: Complete newbie seeks start

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

Hi .. I've just received in the post my Wildcard card with a single FXO and three FXS daughter cards. I've identified a dedicated PC to function as the * machine and installed the card. I've installed Fedora Core 2 on that machine. I've downloaded the * software and the zaptel drivers. And now, to be quite honest, I haven't got much of a clue what to do next! I've

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that should match NXXNX. Right? I built another route 01144[0-9]* that I thought would match 01144X. and send the call to the UK but the script is matching 01144207108???? With the first route. Can someone smarter than me help with some samples? Please? If I can get one for 1NXXN. and 01144. I should be able to figure the rest

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical

I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami

----- Original Message ----- > From: Greg Hill <gregh-asterisk@hillnet.us> > Subject: Re: [Asterisk-Users] New to * > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <Pine.LNX.4.44.0409031231070.1975-100000@hillnet.us> > Content-Type: TEXT/PLAIN; charset=US-ASCII > > On Fri, 3 Sep 2004, Bill

----- Original Message ----- > Message: 3 > Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT) > From: David Troy <dave@popvox.com> > Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over > National or DMS100 PRI to a Norstar MICS? > snip> > > I have a PRI link up and running between Asterisk and a Nortel Norstar MICS > > v4.1 . I'm having a

Back in the office post-astricon. 1.0.0 running in the lab. YIIIIIIIHAAAAAAAAAA! THIS GUY!!!! rocks. Thanks to Mark for *, Steve and Olle for the conference and to ALL community members. Everyone using * is contributing in one way or another. See y'all next year Jason Kawakami www.optellabs.com

Hello all- Playing around with res_odbc (thanks bkw) and have successfully gotten sip.conf to run but am having difficulty with voicemail.conf and extensions.conf. I used the load_res_config.pl script for each one and all of the data seems to be in the DB but * doesn't seem to see anything after a reload even though it is acknowledging a load of x.conf (where x is extensions/voicemail)

-----Original Message----- > -this is very true, however, the current version of the Axxess software > (9.0) supports SIP trunking natively on the IPRC. I just got my Axxess > upgraded and am salivating to get * connected to it. Hmm, so 9.0 is out and it supports SIP natively. How did you plan to integrate the 2? -The Axxess will see the * as it would see an IP service provider.

Sorry everyone, I know this has been hashed over a bunch of times but I can't find anything that pertains to specific cracking and popping on the FXO modules of a TDM04. This happens on inbound or outbound calls. This is the first install I have done with a TDM card for FXO modules so please, be kind if I am missing something really simple. Damn I wish everyone wanted t-1's or

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

----- Original Message ----- > Subject: Re: [Asterisk-Users] Analog Phones with Status Light Indicators > From: Adam Goryachev <mailinglists@websitemanagers.com.au> > To: asterisk-users@lists.digium.com > Organization: Website Managers > Date: Thu, 12 Aug 2004 14:53:02 +1000 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-08-11 at 20:42, Steven

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

Hi I have recently installed samba on a RH7.0 box. This box has win98 + linux rh7 installed and both os's are running well. The F drive on windows is FAT32 and is mounted r/w on linux RH7. Works fine. I also have a Win98se machine that needs to have full read/write access to some partitions on the RH7 box. Samba was found ideal for such setup. The problem I have is that I can't write to

After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have

[Moving this thread over to the public mailing list] On Mon, Dec 02, 2013 at 01:17:09AM +0100, Pino Toscano wrote: > On Friday 29 November 2013 18:53:38 you wrote: > > > > I'm marking options with NONE (not used at all), or M (used in > > > > master) and/or RH7 (used in rhel-7.0). > > > > > > > > > * -nographic > > > > >