search for: litnimax

...called! Aug 31 08:07:28 NOTICE[1135618864]: app_mp3.c:91 timed_read: Selected timed out/errored out with 0 -- Timeout on IAX2/maxhome@maxhome/3 == CDR updated on IAX2/maxhome@maxhome/3 -- Executing Hangup("IAX2/maxhome@maxhome/3", "") in new stack == Spawn extension (litnimax-in, t, 1) exited non-zero on 'IAX2/maxhome@maxhome/3' -- Executing Hangup("IAX2/maxhome@maxhome/3", "") in new stack == Spawn extension (litnimax-in, h, 1) exited non-zero on 'IAX2/maxhome@maxhome/3' -- Hungup 'IAX2/maxhome@maxhome/3' new*CLI&gt...

Hi, I'm looking for some software to allow users to login and manage Asterisk conferences (define conference user lists, call users to join a conference, etc.). I'm aware of astconfman (https://github.com/litnimax/astconfman). Are there any others? Thanks in advance. Regards, IanG

Hello Does anybody who have experience in installing the h323 modules in asterisk. I try'd it many times an spend xxx hours to install it but didn't get lucky so far. I have asterisk 1.0.1 running with bristuffed 2.0 I'm willing to pay for this. Sjaak

Hello all. I am trying to use app_mysql. It works for selects and functions, but does not want to work with procedures. Pls have a look: Calling function: CREATE FUNCTION f_1(a VARCHAR(20)) RETURNS INTEGER RETURN (SELECT count(*) from peer where name = a); Result: -- Executing Macro("IAX2/100-3", "local|100") in new stack -- Executing MYSQL("IAX2/100-3",

Hi all, did not find much info in lists about subj. I have ztdummy working properly because I can use conferences without any errors. But when I try to use trunk=yes, I get the following: Sep 2 21:20:51 WARNING[1137720112]: chan_iax2.c:6422 build_user: Unable to support trunking on user home' without zaptel timing Sep 2 21:20:51 WARNING[1137720112]: chan_iax2.c:6246 build_peer: Unable to

Good time of day all! 1) I am trying to use as5300 and asterisk. As5300 sends calls to me. I get the following in * console: -- IAX2/magrathea/6 is making progress passing it to OH323/R27464 Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate: Ignoring PROGRESS indication. As5300 user does not hear anything, just silense instead of dial tones. My config is oh323.conf

Hello asterisk-users. I have the following dial-plan: [test] exten => 482,1,Dial(OH323/106@192.168.2.73,10) exten => 482,n,Dial(OH323/102@192.168.2.73,10) exten => 482,n,Dial(OH323/103@192.168.2.73,10) exten => 482,n,Dial(OH323/104@192.168.2.73,10) exten => 482,n,Dial(OH323/105@192.168.2.73,10) exten => 482,n,Dial(OH323/106@192.168.2.73,10) When I call exten =>

Hello all! I googled lists.digium.com and ser mailing list, but did not find any working configuration of asterisk used as voicemail for SER. This is my config if (uri==myself) { if (method=="REGISTER") { save("location"); log (1, "Registered\n"); break; };

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware

Hi all! I am intrested in the following scheme My mobile phone -> SMS to SOMETHING -> Redirect to FWD number -> FDW redirect to my * -> My * doing smtg There are companies like calluk.com that provide DIDs for free, but they do not support SMS. In http://www.voip-info.org/wiki-Asterisk+cmd+Sms they say "Works to ETSI ES 201 912 compatible with BT SMS PSTN service in UK"