similar to: Hard Coded CLASS Codes (was 11 instead of Star)

I have several older telephones with rotary dials that I would like to use a working museum pieces. I have everything working well except for those hard-coded codes that start with *. In the traditional phone world, dialing 11 in place of * works fine, ie, someone could dial 1172 in place of *72 and so forth. I'm thinking that a simple entry in extensions.conf ought to do the trick, but

Source: http://www.nanpa.com/number_resource_info/vsc_assignments.html See also: http://bugs.digium.com/bug_view_page.php?bug_id=0000071 Some (which?) of the codes below are hardcoded into Zap channels only. Is there a European equivalent for this (or ITU / IETF)? Greetings, Philipp VERTICAL SERVICE CODES (US Standard) ASSIGNMENTS *00 - Inward Voice Activated Services (English) *01 -

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

This appears to have been the magic bullet for me. Thank you very much. So, the bottom line is that there is a bug that ends up making inbound calls use type=peer rather than type=user. Correct? > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paul Cheng > Sent: Tuesday, August 10, 2004 8:35

Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542

http://bugs.digium.com/bug_view_page.php?bug_id=0002945 Test it.. I couldn't sleep tonight... thought I would see if I could find and fix it... Also did this gem too for ya... http://bugs.digium.com/bug_view_page.php?bug_id=0002948 bkw

Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com

Will the FXS cards that work with asterisk handle rotary? Are there any channel banks that can convert rotary to touch tone (like some sorta bridge)? The goal is to be able to log input from rotary phones. Full PBX functionality would be nice but... (It's for a project, not for serious production). -- // Ethan O'Toole //

Looking to install asterisk for a client and was shopping around for prices for 6 POTS lines with or an integrated T1 with voice and data. I called up Sprint and I told the sales rep that there was going to be a Phone system she said that they would have to install "key" rotary lines and then I told the rep that I needed two more lines with hunt group and she told me that would be

Can someone take a look, tell me if this is a bug, a possible resources issue, or my own damn fault? http://bugs.digium.com/bug_view_page.php?bug_id=0001381 Thanks, Derek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040407/f8f4d79b/attachment.htm

Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on

I'm trying to use a couple of old Western Electric type 500 phones (desk model, rotary dial). These phones work fine, as tested with telco lines (they dial, receiver/transmitter works fine, etc). I'm running Asterisk 1.6.2.11. I can't get them to dial through Asterisk. They are connected to a Rhino channel bank which is connected to Asterisk via a Sangnoma card (T1 with echo

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

In response to the SIP and NAT discussion, I have updated the ticket on the subject that seemed to be getting the most attention: #104. There are enough clueful people here that perhaps someone can come up with a patch that handles NAT in the elegant way that I describe in the bugnotes, as I am but a mere integrator who has limited C skills. In the absence of such a patch being offered, we

So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=0000889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the

The recent addition of recursive mutexes to Asterisk is causing a lot of problems on FreeBSD servers. I need help from someone that knows mutexes on FreeBSD to make it work, otherwise the FreeBSD port of 1.0 will be useless. See bug report http://bugs.digium.com/bug_view_page.php?bug_id=0001411 for more details. Thank you for your help! /Olle

http://bugs.digium.com/bug_view_page.php?bug_id=0001589 Has anyone else heard an audible blip, break or garble between answer and the native bridge attempt using sip? If I change the usleep(500000); to usleep(5000); in rtp.c the proble totally goes away... even the note above it says it needs to be fixed. bkw -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Should this patch work against current cvs? Of the 3 files 2 are .patch and one is .diff - what's the difference between them, and how should I

Can anyone tell me how I can implement the features added in the following link for call transfer? The authors seem to feel they are finished but it doesn't appear to have been integrated into what everyone can download. It is referred to as a patch but I don't understand how it could be applied. Here is the link: http://bugs.digium.com/bug_view_page.php?bug_id=0002010 I guess I just

hi Username : 1000012 Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (gsm|g729|g726|alaw|ulaw) the above is from SIP SHOW PEER 1000012, and as it clearly shows, g.729 is preferred before alaw. If I dial this SIP - * - SIP from a phone with G.729 enabled, it uses G.729. However, if I dial from my cell phone - GSM - PSTN - * - SIP, the call uses ALAW, which I thought it