search for: bours

Hi, I recently upgraded the firmware on my Blackberry 8700 to 4.2, this seems to give it the ability to play wav files. I wondered if anybody out there had managed to get their BB to play the wav files as attached to the Asterisk voicemail emails? Mine seems to ignore the attachment. I am using BES 4.1 for sending these emails out via Exchange 2003 if that makes a difference. thanks Mike

Has anyone tried to do a script to pickup an ITSP voicemail. Lesnet provides an option for an overflow mailbox in the event a caller can get to my * box. I'd like my * to poll it and dump any messages found into my general mailbox Any ideas Similarly, a telco mailbox. It at least has the advantage of having stutter dial tone as a trigger Any hints or suggestions welcome D Dave Bour

Hello, Looking the asterisk 1.8 API documentation (http://www.asterisk.org/astdocs/api/index.html), I see a lot of new fields for sip register uris: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] But the *peer* is not explained anywhere. What it is for ? Regards, Guillaume Bour. -- Guillaume Bour<gbour at proformatique.com>

Hello. Is it possible to "mix" musiconhold music and playback voices? What i want to do is something like this: A person calls a number, gets a playback voice while in background music is playing. The configuration i use at the moment don't do what i want. Someone knows how to do it? Thanks in advance. exten => 18,1,Answer exten => 18,n,Background() exten =>

network outages and recent tests have shown that works well, albeit the switch takes about 20 minutes to propagate the dns updates but otherwise flawless. =20 It's embarrassing and I'm losing credibility when clients are asking if I'm still in business as the phone has dropped way to often in the past few month. Interesting enough all outages to date have been Fridays or Mondays. =20

I've had slew of problems with my Bell Canada Single Number Reach (SNR) dropping in the past couple of months. Another outage Monday for several hours has me wondering if there's a way to 1. Make a call out of my system via a PSTN back to my SNR line, say every 30 minutes (this I'm sure is easy enough via the call file...however...) 2. Track the outgoing call and match to an

Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says "Request sent." The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but

Requests 1. Directories linked to their databases 2 weather broadcasts 3 local traffic info. 4 local news headlines 5 sms send / receive 6 alarm on the phone of calendar events - not a call back, simply a beep and notice without pulling out a pda or opening outlook Couple of other "one of's" too, some very esoteric like evening's TV lineup, Dave Bour Desktop Solution

Yes Dave Bour Desktop Solution Center 905.381.0077 dcbour@desktopsolutioncenter.ca For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) ----- Original Message ----- From: asterisk-users-bounces@lists.digium.com

Hello, I have been using Voicestick inbound (no outbound) successfully for the last few months. Noticed in my logfile that sip registration failed on 6/27/07 at 3am EDT and no successful registration since. Calls to my number eventually timeout as I don't have voicemail setup - as the first step in trouble shooting I tried to enable voicemail on the voicestick website but this fails also

Are there ATAs that allow different phone numbers from one network connection? Such as supporting multiple IP addresses so that each RJ11 has a different extension or some other way?

Hello, I would like to known how to use DUNDi with a Lua dialplan ? In extensions.conf, we should do like these: |[lookupdundi] switch => DUNDi/priv [internal] include => dundiextens include => lookupdundi exten => _XXXX,2,NoOp(calling ${EXTEN}) exten => _XXXX,n,Dial(SIP/${EXTEN}) exten => _XXXX,n,Hangup()| priority 1 is either defined in dundiextens (local registered

...sswd [share] path = /home/samba writable = yes valid users = @users create mask = 0777 directory mask = 0777 browseable = yes even guest ok = yes I have created users accounts on RH server (the same like on suse server) wbinfo gives me users and groups on domain. I can see samba RH on network neighbours and I can rich the RH server to see share, but I can't get in share because message box appears on the screen asks me to write login and password. So I can map this share from login script on PDC. Can anybody explain me what is wrong, where I made mistake. Help please Robert bours --------...

Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY

Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server.

Using trixbox (or a custom dialplan if needed) has anyone been able to convert a number dialled like +612421100000 to something like 024221100000 ie (remove the +61 and replace with 0) i just dont know how to set it up, there seems to be no dialplan wildcard i can use to match +. I was thinking of something like .61XXXXXXXXXX but that still seems wrong to me. it could match other numbers.

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

Currently I want to implement in-place editing directly on the index-page (for the sake of learning just xx products with a name to be edited). These are my favorite links from yesterday''s research (for the archives): *On the spot is a Rails3 compliant unobtrusive javascript in-place-editing plugin: http://rubygems.org/gems/on_the_spot

Hello, Could anyone please give some information on configuring asterisk as a gateway. What contents have to add in h.323 .conf and extensions.conf files ? Thanks & Regards Bincy K Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071102/53be51ce/attachment.htm

I am a new trixbox user. One of the things I'd like to get working is being able to tell if a user is calling me by directly dialing my extension, or if they pressed 1 for sales. When they press 1, it rings a group of phones, and the call is almost always for someone else. So I'm always looking at my phone when it rings, trying to recognize the incoming number and decide if I