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Please note that we are available to fix the current REMOTE crash that affects Asterisk/openpbx/trixbox and crashes these systems via a malformed packet please contacts use if you need a hand to patch your systems. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part -------------- An HTML attachment was scrubbed... URL:

...erminal or other services. The lowest-performance device that plugs into the USB, with its own Linux instance. In OEM quantity, under $50? Under $100? On Sun, 2007-04-01 at 02:51 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Sat, 31 Mar 2007 16:02:06 -0500 > From: "Mike Lynchfield" <theclubvoip@gmail.com> > Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone > To: michael@vandonselaar.org, "Asterisk Users Mailing List - > Non-Commercial > Discussion" <asterisk-users@lists.digium.com> > Message-ID: >...

We have decided to allow our tech's to do support for non-clients of voicemeup.com You can head to http://support.voicemeup.com/ and one will be in touch 8 to 6pm business hours. 3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part

I'm trying to get an IAX trunk to failover to a local trunk it the trunk is down. This is what I've been working on: [macro-forward1]; exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20) exten => s,2,Goto(call-${DIALSTATUS},1) exten => s-CONGESTION,1,Dial(LOCAL/${ARG2},20) exten => s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20 ;end macro-forward1 exten =>

Hello eveybody, I've been looking for a way to share trunks between two asterisk servers. I guest I have to use Dundi, but I've not found the exact method yet. I need a way to allow users registered in one server to use the another server's trunks in the case the first server's trunks were busy and vice versa. Is this possible? Thank you so much, any comment will be useful.

Hello, I created a new field named pre_dst of type varchar(80) exactly like dst field in cdr table. In the dialplan I put: exten => _7.,1,Set(CDR(pre_dst)=${EXTEN:1}) and when I call, all goes fine except that pre_dst has always NULL value in cdr. Do you know why? Is something wrong I did? I know that original fields in cdr are only readable, but in this cas pre_dst is one I created

I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s -- Abra?os Luis Claudio Mobile + 55 21 9215 2888 Mobile +55 15 9141 8402 Office +55 15 2102 5859 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/b8593cb1/attachment.htm

Kinda confused by this... I have a Cisco 7960 configured with a couple SIP extensions configured on the phone. Just trying to dial one extension from the other on the same phone, but when I do, I get: -- Remote UNIX connection -- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new stack -- Called 2002 -- Got SIP response 486 "Busy here" back

Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got

Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate

Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? ? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't