search for: astreisk

Hi Once I installed astrisk , how do we connect with SIP interface ? Can somebody guide me how to integrate SIP interface with asterisk ? I want to use Astrisk just for IVR purpose. Thank you Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130511/a6d32bfe/attachment.htm>

Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with "envelope=yes" and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full: File digits/afternoon does not exist in any format [Apr 10 14:09:08] WARNING[12955]: file.c:866 ast_streamfile: Unable to open digits/afternoon (format 0x2 (gsm)): No such file...

Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving...

Hi; It is my first time installing an asterisk PBX system . I do have a TDM400 wildcard with 4 FXO moduls on a PC with 3.0GHZ HT CPU and INTEL 915 moatherboard . Fedora C2 Linux as O.S. and I have the latest CVS astreisk , Zaptel and Libpri downloaded the zaptel drivers installation and configuration seems to be fine and the libpri but when I tried to compile and install the asterisk software the following error occurred : Chan_zap.c 2772 : error : "Zt_event_DTMFDIGIT" undeclared Can any body help why...

Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in "allow=gsm" line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being...

...didn't experience on my setup. Today I upgraded my asterisk 1.0.5 to 1.0.6 and suddenly I have an echo too on sip calls thru wengo!! I already verified wengo was not the source of the problems by switching accounts on the server so I'm pretty sure the problem appeared after installing astreisk 1.0.6. Is anyone else experiencing this problem and is there anything that can be done about it? Both setups are similar, an * box with SIP phones (Grandstream and Sipura SPA-2000's) and a wengo sip account to place calls. Thanks! Remco

...ssfully dial out using dial( SIP/{$EXTEN}@sipserverout) but now when I call my incoming number, I get a busy or invalid number signal. If I coment out sipserverout section, I could receive incoming calls again. So I turned on sip debug on CLI. and it appears to me that the following is happening. astreisk takes the incoming call and tries to match it with a section with the same hostname. Now the reverse IP lookup on 109.147.41.48<http://109.147.41.48>return sipserver.com <http://sipserver.com> (which is correct), so it is trying to send the call to sipserverout which is essentially back...

...ence to `std::__pad<char, std::char_traits<char> >::_S_pad(std::ios_base&, char, char*, char const*, int, int, bool)' collect2: ld returned 1 exit status make[1]: *** [ch_lab] Error 1 make: *** [main] Error 2 I have successfully compiled and used the same festival version with astreisk on redhat 7.2/3. Any help would be appreciated. Azher

hello list, i need your help please regarding send mail i use astreisk 1.4; i try to send mail when no response like below exten => 5xx,1,Dial(SIP/223, 10) exten => 5xx,n,system(echo test ${DNIS} Email| mail -s 'Call failed' myadresseemail at gmail.com) when i launch the CLI i found : You have new mail in /var/spool/mail/root i check the root and i...

Hi, I'm currently using chan_capi-cm-0.6, with the following capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de [ISDN1] msn=8304490 incomingmsn=8304490 isdnmode=msn group=1 controller=1 softdtmf=1 context=demo echosquelch=1 echocancel=yes echotail=64 callgroup=1 devices=2 Each user has a different numer, e.g. 83044910, 83044911, 83044912 and so

Hi List, Im planning on setting up asterisk for a large scale enviorment, with multiple sites. We will be doing quite a bit of inner office calling at each site, and want to place a smaller scale * box at each site with no PRI's, and have that connect to our main * servers at our data center that will have the PRI connections. Can this be done? I havent seen to much of this on the mailing