search for: zerozone

...ogram received signal SIGSEGV, Segmentation fault. [Switching to Thread 49156 (LWP 5719)] 0x405195cc in speex_encode (state=0x7ec30d93, in=0xbf3ff624, bits=0x80955bc) at modes.c:575 in modes.c (gdb) i'm using speex 1.0.5. Someone can help me ? Thanks ! Oz -- ---- O-Zone ! No (C) 2005 www.zerozone.it -- ---- O-Zone ! No (C) 2005 www.zerozone.it

...============================== As you can image i use speex to clean and elaborate audio from a voice modem. All work correctly BUT i've a ugly "noise" in the output sound data. There's some hints that i've missed ? Thanks ! Oz -- ---- O-Zone ! No (C) 2005 WEB @ http://www.zerozone.it HOBBY @ http://peggy.altervista.org Call me with FWD: 692329 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20050922/c59ab93d/attachment.htm

...ogram received signal SIGSEGV, Segmentation fault. [Switching to Thread 49156 (LWP 5719)] 0x405195cc in speex_encode (state=0x7ec30d93, in=0xbf3ff624, bits=0x80955bc) at modes.c:575 in modes.c (gdb) i'm using speex 1.0.5. Someone can help me ? Thanks ! Oz -- ---- O-Zone ! No (C) 2005 www.zerozone.it

...<< 8) - 32640; tbc++; if(tbc > NN) { ret = speex_preprocess(speex_pp_state, TEMP_Buffer, NULL); } at this point TEMP_Buffer contain original sound *with* noise. Ideas ? Suggestions ? Thanks ! Oz -- ---- O-Zone ! No (C) 2005 WEB @ http://www.zerozone.it HOBBY @ http://peggy.altervista.org Call me with FWD: 692329

...one number) and we plain to install an Asterisk server as PBX. We need to have 15 PSTN devices (phones, fax, etc) in opur office. I've seen FXS and FXO but i'm not sure: we need 7 FXO and 15 FXS ?!?!?!?!? There's a smarter solution ? Thanks ! Oz -- ---- O-Zone ! No (C) 2005 www.zerozone.it

...erent action based on incoming call. For example, if call is on Line 1 i want to redirect it to extension 203, on line 2 to extension 201 etc etc it's possible ? How ? Looking in AMP i'm using AMP to manage Asterisk) this is not possible... Thanks ! Oz -- ---- O-Zone ! No (C) 2005 www.zerozone.it

Hi all, i'm using Asterisk with several extensions with 7 PSTN lines. Is possible, for a caller, to dial directly an extensions ? For example, dial something like [PSTN number]*[ext number] ? Thanks ! -- ---- O-Zone ! No (C) 2005 www.zerozone.it

...However i've done some tests on .tif file created by asterisk and i've noticed that it truncates my fax almost after 5-6 seconds. As results my pdf are corrupted and i receive a mail with empty pdf :-( someone can help me ? Thanks !!! Oz -- ---- O-Zone ! No (C) 2005 WEB @ http://www.zerozone.it HOBBY @ http://peggy.altervista.org Call me with FWD: 692329

...Sep 8 13:28:55 WARNING[10750]: Unable to allocate channel structure Sep 8 13:28:55 NOTICE[10750]: Unable to request channel Zap/g1/2430 Sep 8 13:28:55 NOTICE[10750]: Call failed to go through, reason 0 please help me !!!! i'm desperate !!! Oz -- ---- O-Zone ! No (C) 2005 WEB @ http://www.zerozone.it HOBBY @ http://peggy.altervista.org Call me with FWD: 692329

...on 1, with 0 conference users Sep 15 13:07:27 VERBOSE[14085]: -- Hungup 'Zap/1-1' Sep 15 13:07:27 NOTICE[14085]: Call completed to Zap/g0/2479 or remote FAX machine say "Error receiving fax" and line remaing stale. Any help ? -- ---- O-Zone ! No (C) 2005 WEB @ http://www.zerozone.it HOBBY @ http://peggy.altervista.org Call me with FWD: 692329

...19521 2 mptscsih 36605 0 mptbase 47457 1 mptscsih sd_mod 20545 7 scsi_mod 116429 2 mptscsih,sd_mod [root@srvoip ~]# It's a RH ES 4.0..someone can help me ? Thanks ! Oz -- ---- O-Zone ! No (C) 2005 www.zerozone.it

...}; so i call SPEEX_Init before start encoding and SPEEX_Encode on every 16000 Bits of the PCM stream from the modem. The result is a strange file that contains a lot of zeros and no audio data :-( Someon can help me pointing to the right direction ? Thanks ! -- ---- O-Zone ! No (C) 2005 www.zerozone.it -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20050622/ade2cd2d/attachment.pgp

...oying call '499575437@192.167.125.9' moloch*CLI> Sip read: 0 headers, 0 lines moloch*CLI> ===========================8<=================================== and i get the VoiceMail apps instead of 201. Why ? -- ---- O-Zone ! No (C) 2005 www.zerozone.it -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050517/f766d928/attachment.pgp

...3,204): [20x] username=20x type=friend seret= qualify=200 port=5060 pickupgroup= nat=never mailbox=20x@default host=dynamic dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no callgroup= callerid="djdjdj" <20x> allow= Help !!!!!!!!!!! -- ---- O-Zone ! No (C) 2005 www.zerozone.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050518/a6d36f9c/attachment.htm

Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. That is why I will post my questions here: 1- Transcoding: is this when

...E, value: OFF(0) on Zap/1-1 Sep 6 11:06:14 DEBUG[10750]: Updated conferencing on 1, with 0 conference users Sep 6 11:06:14 VERBOSE[10750]: -- Hungup 'Zap/1-1' but pdf is illeggible and TIFF corrupted ! Please someone can help me ? Oz -- ---- O-Zone ! No (C) 2005 WEB @ http://www.zerozone.it HOBBY @ http://peggy.altervista.org Call me with FWD: 692329