similar to: Grandstream Source?

any way the you can get * to send a NOTIFY SIP message to all SIP phones? to have the SIP sets recheck thier configs etc?? Like this? NOTIFY sip:sip@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP ipaddress:5060;branch=1 Via: SIP/2.0/UDP ipaddress From: <sip:webadim@192.168.0.1> To: <sip:sip@192.168.0.3> Event: check-sync Date: Mon, 10 Jul 2000 16:28:53 -0700 Call-ID: test@192.168.0.1

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of David Burr > Sent: Sunday, January 11, 2004 4:31 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive > > > We have a new contest starting today! > > The first three

Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Hi again, I found chan_skinny and that seems to work pretty good. the SCCP one filled out all the buttons really nice, but skinny seems to be working. How do I fill out the second line button on the phone with skinny.conf? Thanks much! ...Jeff

So far in January, we've had 726 messages on -users. December 2003: 2.978 messages November 2003: 3.410 messages October 2003: 3.177 messages December 2002: 741 messages December 2001: 67 messages ...the project is growing. /Olle

Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy

Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul.

Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default &password=000012&msgid=0000&format=gsm&dontcasheme=4624.gsm HTTP/1.1" 200 9438

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

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Gary Franczyk > Sent: Thursday, January 15, 2004 10:37 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question > > > Whaaa?? So, to allow 24+ lines of dial in access, how

Hi, I'm trying to install OpenShift Origin on a CentOS 7 host (just for initial testing), and I'm trying to follow the instructions from here: https://wiki.centos.org/SpecialInterestGroup/PaaS/OpenShift-Quickstart On that page we need to run: "atomic-openshift-installer install" to configure OpenShift... after run the script it throws this errors: Failure summary:

Anyone know what happened to netxusa? Seemed like they dropped off the web overnight. -Matt

I am trying to use chan_skinny but when loading the module I get: [ Booting....../usr/lib/asterisk/modules/chan_skinny.so: undefined symbol: ast_pickup_call I am using CVS 07/23 I can't get chan_sccp2 to compile, it gives me parse errors, or I'd be using that. :-/

>how does the phone know where to find the TFTP server..? Dude, option 150 in your DHCP server: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186 a00800942f4.shtml We use the same option for our Mitel phones. HTH.

Guys. Anybody has developed and asterisk brochure for commercial purposes (consultant, etc) that I might be able to take a look at?

Have Asterisk us at running fine, but have run into a small snag. It's not emailing the voicemails to the users. I have attach=yes set, sendmail is configured and works from from the commandline (sent an email to myself). Unless I'm wrong, or missing something, asterisk is configured by default to send an email when a users receives a voicemail, correct? Thanx A

Our main asterisk box peers with that of a customer. We are trying to assign DID's to their extensions but get this error. What are we doing wrong? Client side Mar 25 18:49:47 NOTICE[1369]: chan_iax2.c:6545 socket_read: Rejected connect attempt from 203.xxx.xxx.16, who was trying to reach 's@' Our side Mar 25 18:56:15 WARNING[705]: chan_iax2.c:5546 socket_read: Call rejected by

Hi all, I know Asterisk is able to send a waiting message (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour