similar to: changed ip now * demo call not working.

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

The orginal did not make it to the list... Spam filter issue??? No repeat of the lockup yet. Lyle -------- Original Message -------- Subject: voicemail locked up Asterisk 1.4.13 Date: Thu, 01 Nov 2007 20:57:27 -0500 From: Lyle Giese <lyle at lcrcomputer.net> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> I am running Asterisk

You could accept as the "passcode" the caller punching in their own phone#, then checking that against your whitelist. Lets associates get past the challenge when using someone else's phone, without their remembering some arbitrary passcode. And strangers or barred old associates who abuse it can get an earful about how you're suing them for wire fraud. Preferably after you

I am putting together my first *. I had it running with two other pc's running xlite and setup voicemail and a couple of menus and submenus and had that running well. I had order a couple of oem x100p cards from digitnetworks. I installed them as they said with their voicepet2.2.zip drivers and did the modprobe on zaptel and wcfxo and then ran ztcfg -vv and got this: Zaptel Configuration

I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

I am not that good at regex and it's use in Asterisk. I am running Asterisk 1.4.13 Currently I have this in my extensions.conf for incoming calls on our house phone line: [housemenu] exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12); 815xxxxxxx is our home phone number, when caller id fails or is missing that is what is recorded. I want to expand this

I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect'

I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) But

Hello everyone.... Searching the archives and google always comes up with entries regarding the "dyn" dns option in the 7960, but I can't find answers to my specific question.... My 7960 is connected via cable modem and is NAT'ed (everything is working fine). On the 7960 under SIP configuration\NAT Address I have the public IP of my cable connection. Comcast gives me a

I am setting up my * for at home office and still have analog phones attached and answer from those analog phones and not necessarily through the pbx. I found that with the X100P cards, they see the 2nd ring and will be ready to answer the line. I used a Wait to pause and allow another 2 rings before * answers. But found that if we answer the line after the 2nd ring and before the 4th, * still

Not even sure how important this is considering the state of many of the online docs... I have doxygen installed as is noted for the requirements for 'make progdocs', but the make doesn't find dot. I have no idea where dot went, is or should have been... I am installing und Suse 9.0 and it's rough. If you forget something duringthe initial install, adding the package later

--Qz2CZ664xQdCRdPu Content-Type: multipart/mixed; boundary="BI5RvnYi6R4T2M87" Content-Disposition: inline --BI5RvnYi6R4T2M87 Content-Type: text/plain; charset=us-ascii Content-Disposition: inline Content-Transfer-Encoding: quoted-printable Hi! Unfortunately there is a very unpopular announcement to be made on this list: A netfilter security advisory. Phillipe Biondi has been

On Wed, 12 Aug 2015, Alexander Dalloz wrote: >> I have been working at trying to get cyrus to listen on 148.197.29.5 >> interface instead of the localhost - I have failed >> > No square brackets around the ip address. imap cmd="imapd" listen="imap" prefork=5 # imaps cmd="imapd -s" listen="imaps" prefork=1 imaps

Hello everyone, I have followed this guide to enable my Windows 10 guest to reach my Arch Linux host: https://access.redhat.com/documentation/en-US/Red_Hat_Enterprise_Linux/6/html/Virtualization_Host_Configuration_and_Guest_Installation_Guide/App_Macvtap.html My VM now has 2 NICs: - macvtap (VEPA) - virtual network 'isolated' I have statically assigned the IP 192.168.254.2 to my

I just want to make a SIP call from 192.168.1.3 to 192.168.1.4; or not even a call. Ring? Beep? Ping? Some sort of "hello world" connection. 192.168.1.1 netgear router 192.168.1.2 asterisk (vicidial) 192.168.1.3 ubuntu client 192.168.1.4 mac OSX client (not shown) Do I have a firewall problem which would impact a soft phone from establishing a connection?

Hello everybody. Recently i moved external interface to zone "external" on my home server/router. And something strange is hapening. From my router (chamber, CentOS7) everything is fine: [root at chamber ~]# firewall-cmd --list-all home (default, active) interfaces: enp3s0 tun0 virbr0 sources: services: dhcp dhcpv6-client dns http https imaps ipp-client mdns nfs samba