similar to: New batch of phrases from Allison

I've been looking at the weather vocabulary in asterisk-sounds in CVS. I've run into a few hitches with words I can't seem to find. So far, I'm looking for 'point' (for constructing floating point numbers) and 'around' as in "high around 70" (don't I wish). Any chance of getting these? While I'm on the subject, I'd be very interested in a

I'm looking for an ISDN card that will work for me using either the i4l or capi with asterisk under Linux. I'm in the US, so I need an ISDN-U interface. Can anyone tell me if this card will work for me? http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=14922&item=6700879965&rd=1&ssPageName=WDVW There are a lot available, and for what seems to be a good price, if

Since I only seem to get questions, and no feedback, from the Wiki page, I'll ask here. There seems to be no lack of opinions here... I have a working wakeup call system on my home * system. The architecture is something I'm not perfectly happy with, though. There are two AGI scripts, written in Perl, which handle (a) scheduling, confirming, and cancelling a wakeup call, and (b) the

Just upgraded from 0.7.1 to the latest CVS version yesterday. This introduced a slew of warnings on startup. About 20 or 25 of the first, then 5 of the second: Jan 23 10:58:49 WARNING[8201]: chan_sip.c:446 __sip_xmit: sip_xmit of 0x80ef064 (len 461) to 0.0.0.0 returned -1: Invalid argument Jan 23 10:58:55 WARNING[8201]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call

Not having a whole lot of luck... I've decided I need to open up the search to cards with S/T interfaces and just find an NT1, too. Can someone with experience give me some pointers what would be easy to find, and easy to configure under Linux? I've stumbled across an Eicon card that's just labeled "DIVA T/A PCI", both on stickers and printed on the circuit board. Is this

I've seen people mention that they have fax reception working with Asterisk, spandsp, and app_rxfax. I'm using the latest Asterisk from CVS, SpanDSP 0.0.1k, and the latest app_rxfax.c (as mirrored by friendly list members recently), and libtiff 3.5.7. Asterisk is detecting the fax signal properly, and executing the fax extension in the dialplan. The fax part of the dialplan is pretty

I am getting ready to submit a list of department names to be recorded. This is what I have so far: Accounting Accounts payable Accounts receivable Administration Billing & Collections Complaint Customer Service Engineering Facilities Help desk Human Resources Information Technology Inside Sales Investor Relations Legal Mail room Marketing Printing Projects Public Relations Purchasing

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

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

http://bugs.digium.com/bug_view_page.php?bug_id=0001429 * Support for other language syntaxes in saynumber Accidentally I opened this can of worms to see if we can add support for other language syntaxes for saying numbers. Seems like Swedish, english and norwegian follow the same syntax. I've integrated existing patches for french, danish and soon portuguese syntax. The steps we're

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

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

Here's the latest batch of words to get shipped out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket. Please send to paypal address "jtodd@loligo.com". I did not include all

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

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: