similar to: ERROR[8192]

Displaying 20 results from an estimated 500 matches similar to: "ERROR[8192]"

2004 Jun 16
2
embedded Asterisk
Hi All, I have a thin cliente here that i want to run asterisk: - National Semicondudor Geode GX1 266MHz Geode 266MHz single chip - NS Cx5530a Southbridge National Semiconductors SC2200 - NS PC97317 in chipset - 32MB Compact Flash - 64MB Ram - 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816 Some tip? I have a ide>flash
2003 Aug 04
14
Mysql CDR
hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end
2003 Oct 31
2
MOH problem
Hi all! Every time i receive a sip call MOH begin to play and i can?t talk to the caller. My setup is the default. Someone knows what is the problem? thanks Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 FWD 64662 ICH 31451543 www.ipfone.com.br info@ipfone.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL:
2003 Sep 23
2
error message playing .mp3
> -----Original Message----- > From: listas iPfone [mailto:listas@ipfone.com.br] > > Somebody knows why asterisk gives me that error wile playing .mp3 files? > > The files play well but the message aperas any way: > WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of > 4 > bytes) (No such file or directory)! Listas, You might try down-sampling
2005 May 26
1
VIDEO ON 1.0.7 stable
--- listas iPfone <listas@ipfone.com.br> wrote: > Hi all > > I need to know if the video support for h.263 is > active in version stable > 1.0.7 to use with eyeBeam in asterisk it works for me... [2222] type=friend secret=xxxx auth=md5 callerid="myCallerId" <2222> canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex
2003 Dec 08
2
snom X MOH
Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:
2008 Dec 22
1
Web-driven SIP call thru Asterisk IPBX
Hi, I think that the web-driven SIP Phone (free) doddle (beta version) can be useful with your Asterisk applications. You can pre-fill it with your sip settings (Asterisk host name or IP /?realm / sip user), you just need to setup the HTML link as that: (Attached is the HTML page example) ? /**************************/ simple HTML code example: /*************************/ <html> <head>
2004 Jan 23
6
rc.local dont works
Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don?t in the initialization... I have in my file that comands: touch /var/lock/subsys/local modprobe zaptel modprobe wcfxo safe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening Somebody
2003 Nov 27
13
Asterisk behind NAT << How to do it.
Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c
2003 Oct 08
7
chan_capi and latest Debian package
After apt-get update && apt-get upgrade -y wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz tar xfvz chan_capi.02.5c.tar.gz cd chan_capi-0.2.5c make && make install shutdown -r now asterisk seg faults upon calling in via ISDN. Any ideas are greatly appreciated. rgds pos
2003 Oct 06
6
Alternatives to FXS cards?
Hi everyone, I know someone makes a product that's a POTS phone to SIP converter, where you just plug your POTS phone in one side and the network cable in the other. Has anyone successfully used any of these with Asterisk, and if so how expensive were they? I ask partly out of frustration with the FXS cards but mostly because it would make installation MUCH easier for what we're
2004 Jan 26
0
Digium FXO Card
Hi, I wish to know if GNUGk can work with * running as a gateway with the Digium FXO card. Kindly share your experiences in case there are some issues which one must know before going in for such a setup. Also, I've been reading about the DialTone detection capability by the hardware in different countries. What are the issues with it? Thanks & Regards, Deepak ----- Original Message
2003 Oct 02
3
Xten Lite Build 1079
I've just down loaded Xten Lite and it is now build 1079. It now finds the NAT firewall type and has loads more to configure. But it doesn't work on my poor W95 tablet PC. -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene 04 90 23 30 81 "Internet Sheriff Technology" revendeur en France <http://www.linuxautrement.com> IAX 17004902330
2004 Jan 13
1
Symbol NetVision Phone
Hi List ! I received an unit of the Symbol NetVision Phone and i will test it with asterisk using H.323 or Skinny , somebody tested this phone with asterisk and can share experience? Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040113/2ea43296/attachment.htm
2005 May 26
5
Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages
I'm wanting to have a phone at home next to the garage door that when my bride comes home, she can see that there is a new message, push a button and have the messages played to her. Otherwise, she will not let me install asterisk on my home line. Can someone suggest relatively inexpensive hardware that will do this for me (us)? Thanks, -Peter
2004 Aug 06
3
[PATCH] Make SSE Run Time option.
Le jeu 15/01/2004 à 15:30, Daniel Vogel a écrit : > Unrelated, but please use SSE/MMX/... intrinsics on Windows instead of using > inline assembly so you also get the speed benefit on Win64. OK, so here's a first start. I've translated to intrinsics the asm I sent 1-2 days ago. The result is about 5% slower than the pure asm approach, so it's not too bad (SSE asm is 2x faster
2007 Dec 21
1
Asterisk SIP handling - why 491 Request Pending response
Hi, I have the following situation I use asterisk as o gateway between networks. What is the reason for such response? What are the criteria for such evaluation? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP
2008 Feb 14
4
domain name display issue in linux pc
Hi, Thanks for your response on the kernel switching.I was away and could not reply immediately. Right now, I am facing a differentissue. I have to set up DNS server using BIND on Centos 4.3. When Itype the hostname on Centos, it shows: sipserver.vodcalocal.com But the cli prompt has root at sipserver~ meaning only the sipserver part of the hostname is displayed. whyis this so? What is the
2006 Dec 18
2
Digium TE405P with French E1 => Red Alert
Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes
2005 Oct 09
1
Problem setting SIP incoming/outgoing
Hi I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search. Simply speaking, I have an external SIP proxy server which I am trying to configure for incoming and outgoing calls from my asterisk installation. So here is my configuration in sip.conf [general] register =>