similar to: cheap ip phone with auto-answer

Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? Heck, it doesn't even need to be able to do calls :-) Leif

I'm running asterisk on a Freebsd with 2 Nic's. Inside NIC is 192.168.5.x where the phones are. Outside NIC used to be a public IP with the ISP's device set to bridging, but the new WiMAX router only offers me the public ip 94.18.x.x on the outside, and forwarding everything to 192.168.1.50 on the "Outside NIC" Some of the phones are being disconnected with Asterisk

I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? Leif

I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) return busy when just one extension is busy. http://www.neland.dk/app_dial.c.diff It works, but... I can't figure out setting/reading an option. It looks fairly easy, but the flag is always set. *** app_dial.c.org 2009-11-04 22:15:50.000000000 +0100 --- app_dial.c 2009-12-01 09:29:19.000000000 +0100

Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service....)

When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? Leif

In relation to our CRM-system I'd like to send a query to asterisk who is extension xxx talking to. When the operator enters the page with customer data, the crm should send a query to asterisk, to get the cli of the call the operator is having. If the number is matching the customers number in crm, a record will be made, if it is not, a popup "Are you talking with this customer

Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat

Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the

Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap

Hello,

Hi Everyone, I am accustomed to PUTTY and it's very nice as in it allows many many SSH profiles to be saved and allows tunneling etc....but it's not very good when it comes to scrolling up and down, colors, text size, and specially it doesn't give a title to the opened instance. Maybe giving the IP address as the title of the window would help a lot if you have many different servers

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Leif, Did You faced that with the exact same Chrome version? Since this version is the only one having this issue. I have had this kind of arrangement (intro + live stream) for decades. Technology changes but the idea is the same. I have tested a lot of hardware and combinations. I do have a fail over stream (with different specs) and that hasn’t been an issue at all. I do not know but I

> I'm pretty sure that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons