[asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice)

Hi, I´ve compiled asterisk-1.8.5.0   on my Debian based distro (Pinguy) I also compiled iksemel (v1.4) with the option 2./configure --with-libgnutls-prefix=/usr As explained in this link (to avoid compilation error ) http://code.google.com/p/iksemel/issues/detail?id=29#c3 I configured

Re: [asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice)

and test it again. Thanks, --Warren Selby, dCAP On Aug 2, 2011, at 12:06 PM, neo haux neo.h...@gmx.com wrote: Hi, I?ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy) I also compiled iksemel (v1.4) with the option 2./configure --with-libgnutls-prefix=/usr As explained

[asterisk-users] Asterisk+internal phones+recorded messages

Hi I want to change my old answering phone machine and two wireless phones with asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel 9133i) + Wifi/SIP phone I am wondering if I´ll lost actual functionalities that are present in my old answering machine: 1) is it possible to

[asterisk-users] Phone numbers and asterisk

Hi, It is possible to save all the phones numbers on asterisk servers instead of doing so manually in each VoIP device ? Does SIP take care of such configuration ? Thanks -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Phone numbers and asterisk

, neo haux neo.h...@gmx.com wrote: Hi, It is possible to save all the phones numbers on asterisk servers instead of doing so manually in each VoIP device ? Does SIP take care of such configuration ? Thanks -- _ -- Bandwidth

Re: [asterisk-users] RTP stream when * and Xlite are both behind Nat devices.

Hi rcswebb, I had a problem like yours : Asterisk -NAT - internet - NAT - 3CX phone Without modifiyng Astrisk conf I could start a call from the client but without hearing a sound. The solution for me was to force Asterisk to modify the outgoing udp packet to insert it's public ip and not the

[asterisk-users] Delay before ringing from PSTN`s call

Hi I am testing a degium TDP400P (2fxo+2fxs) on my asterisk I configured incoming calls from pstn to ring my SIP phone (extension : 100) cat extensions.conf ... [from-pstn] exten = s,1,Dial(SIP/100,10) same = n,VoiceMail(100,u) root@PC-debian:/etc/asterisk# cat dahdi-channels.conf ... ...

Re: [asterisk-users] Delay before ringing from PSTN`s call

-users@lists.digium.com Cc: neo haux neo.h...@gmx.com Message-ID: 4e8b5553.2030...@stromberg-carlson.org Content-Type: text/plain; charset=iso-8859-1; Format=flowed You need to make sure your (DAHDI or ZAPTEL ) is set up properly for your country's CLID protocol In the US CLID is sent between

[asterisk-users] Can't make call with TDM410P

Actually I can start and receive SIP calls (PC client, iphone client) but I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I'm having this  error when making a call: *CLI   == Using SIP RTP CoS mark 5     -- Executing [9000@local:1]

[asterisk-users] Fwd: asterisk-users Digest, Vol 95, Issue 33

-Commercial Discussion        asterisk-users@lists.digium.com Message-ID:        cac8s5nrg34tqgqn+ff0kuqr2mpcs8wovf7yghwh_gbz67u7...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 On Sat, Jun 23, 2012 at 10:32 AM, neo haux neo.h...@gmx.com wrote: Actually I can start and receive SIP calls (PC

[asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU

Hi, I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2 800MHz CPU. A small box to play with PBX at home. I get this error when I start asterisk: root@pbx01:/usr/src/asterisk-11.1.0# /etc/init.d/asterisk start Illegal instruction Starting Asterisk PBX: asteriskIllegal

Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU

Message-ID: cahkv19d6j0jfs3lgyyxsarqckhk32mgn6ciih9v8uwz4fqt...@mail.gmail.com Content-Type: text/plain; charset=utf-8 Try this... In menuselect, uncheck BUILD_NATIVE under Compiler Flags and recompile. On Sun, Dec 30, 2012 at 4:44 PM, neo haux neo.h...@gmx.com wrote: Hi, I've

[asterisk-users] Build asterisk for VIA C3

contact debian dev team for that? Thanks OLD messages --- Message: 6 Date: Mon, 31 Dec 2012 12:08:40 -0500 From: neo haux neo.h...@gmx.com Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU To: asterisk-users@lists.digium.com Message-ID: CAHtT-j

[asterisk-users] weird RED alarm on FXO channel

I have a recurrent problem on my asterisk box. I have VIA Samuel 2 as a CPU. With asterisk v11.1.0 and dahdi-linux-complete-2.6.1+2.6.1 compiled from source. I get a RED alrm drom the port 1( FXO) two or three times per day: [Feb 4 15:54:57] WARNING[9991]: chan_dahdi.c:8018 handle_alarms:

[asterisk-users] How to show caller number ?

Hi, I am using asterisk 11.1.0. How to display the caller number (from asterisk -rvvv terminal) in the first step of the extension (before doing any action) ? Thanks -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Joining an astablished call

Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and

[asterisk-users] How to use Skype ?

Hi, I want to recieve calls to my Skype account and forward them to a SIP/FXS line. I searched for chan_skype for asterisk (v11), but found it only available for asterisk 10 I know that Digium gives no support for this module, but I am sure that someone somewhere did write some tool to allow

[asterisk-users] (no subject)

To Jonas: I have an asterisk box at home and I have this line in my rtp.conf file: rtpstart=1 rtpend=10100 And My FW is setup to forward all incoming ports of range 1-10100 to the asterisk PC. I've never had a problem since one year, but I have never received more than two simultaneous

[asterisk-users] No voice when the calls come from Internet

Hi, I have trouble establishing a call between between two SIP phones. One sip phone is, with asterisk server, at home behind a firewall. The second sip phone is an iPhone with 3G wireless connection. When I call from the SIP device at home the SIP account on the Internet (iphone + 3G) I can

[asterisk-users] Google Puts the Final Nail in the Google Voice Coffin

No tell me that's a jock ! I can't believe it: http://nerdvittles.com/?p=7940 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: