similar to: Dahdi/callerid issue

Hi, I’m currently using 32 CENTOS 5 but it’s now unsupported. I only have a 32 bit processor and CENTOS no longer supports 32 bit so I need to move on. I’ve installed the current version of 32 bit Fedora and I can’t get the latest Dahdi to build. Even tried downloading the early release DAHDI from github with no luck. Any recommendation for which 32 bit LINUX to use going forward? Or

To whom it might concern, The Gesko Ikarus 1200S analog telephone has advanced callerid capabilities. When used with an ATA186, it show the username and the phonenumber of the caller. (or whatever you let * tell it) http://www.gesko.be/idgg004.htm Price is 77 euro something and available with Telec. (NL) Met vriendelijke groet, Pauline Middelink -- GPG Key fingerprint = 2D5B 87A7

I finally decided it's been long enough using my ancient HP junker and I built a Atom 330 based machine to replace it. I've installed Centos 5, Dahdi and Asterisk 1.6.2. After a bit of struggles getting the 1.2 version files converted to 1.6 almost everything seems to be working. All my SIP lines seem to work and calls come in as expected and all my DAHDI lines work with incoming

Hi I have a TDM400 and 4 channels of HPEC. I don't use the POTs lines much so I didn't realize it wasn't working. This morning I was watching the console and noticed that the echo canceller didn't load when a call came in. /etc/dahdi/system.conf showed mg2 for all 4 channels. I changed them all to hpec and then restarted dahdi and Asterisk and suddenly HPEC was working. Is

Hi to all, I have a question about the callerid (msn number) off calls comming FROM a queue. This is my setup: - ISDN using zap (zaphfc) - Incomming calls arrive in a queue - One of the members of the queue is my cell phone... member => Zap/g1/XXX73XX19 (X to protect my privacy ;-)) The problem is the call from the queue to my cell phone. In a dailplan i can set the outgoing msn using

Hi All, After searching and didnt found it, im just sending my situation here, maybe someone knows where i should look. Im using Asterisk 1.6.1.10 Internally the user with a sip phone dials a number for instance 0623456789 It goes fine to the specific dial rule: which is: exten => _0[6].,2,Dial(SIP/0${EXTEN:1}#@xs4all-out,60,tTwWkK) This works fine without a charm, but the situation is that

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.7.0.2 DAHDI-Tools-v2.7.0.2 dahdi-linux-complete-2.7.0.2+2.7.0.2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete *** THIS RELEASE FIXES

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.7.0.2 DAHDI-Tools-v2.7.0.2 dahdi-linux-complete-2.7.0.2+2.7.0.2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete *** THIS RELEASE FIXES

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.8.0-rc2 DAHDI-Tools-v2.8.0-rc2 dahdi-linux-complete-2.8.0-rc2+2.8.0-rc2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In version 2.8 we

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.8.0-rc2 DAHDI-Tools-v2.8.0-rc2 dahdi-linux-complete-2.8.0-rc2+2.8.0-rc2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In version 2.8 we

Hi all! I need a simple plan for the following: *answer call *wait for 4 digit extension *send call to 4-digit extension entered. I tried the following, but that doesn't work... exten => 998,1,Answer() exten => 998,2,Background(agent-newlocation) exten => 998,n,WaitExten(20) exten => 998,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) WaitExten obviously does not fill EXTEN with

Hi everyone! situation: Asterisk-server A: 192.168.11.6 Asterisk-server B: 192.168.2.44 server B contains a register => username:password@192.168.11.6 But... when I boot it, I get: Registered to '192.168.11.6', who sees us as 10.138.3.2:4569 Why doesn't server A see server B as 192.168.2.44?? All other traffic going over these lines has no problems with this. The

On 12/14/19 11:29 AM, Greg Troxel wrote: > sean darcy <seandarcy2 at gmail.com> writes: > >>> There is also the ObiHai OBi202 with an OBiLine, which provides an FXO >>> port remoted over SIP. (I am not sure if this is discontinued.) >> >> "FXO port remoted over SIP"? >> >> I have an analog phone system. I can use the obi202 to

Greetings! Being a Perl hacker for some time, and wanting to leverage what R provides, I've been trying to work with Statistics::R and RSPerl. The former has a race condition that breeds some unreliability and the latter seems to have issues all around, and neither has been updated in some time. Are these projects are abandoned, or is there some effort currently being undertaken to

Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

Hi all, I have been looking at various alternatives for a system that runs several Linux & Windows guests. So far my favorite choice would be OpenSolaris+ZFS+RAIDZ+VirtualBox. Is this combo ready to be a host for Linux & Windows guests? Or is it not 100% stable (yet)? Greetings, Evert This message posted from opensolaris.org

Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf ***** [ip-incoming]

Hi everyone! I'd like to create the following: a user picks up the phone (gets a dial tone), dials '0' for an 'outside' line, gets a second (different?) dialtone, and is able to enter an external phone number. How do I implement this in extensions.conf...? Regards, Evert