similar to: rc.local dont works

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

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

Hi all! I get this error when trying to start asterisk: ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk What can be the problem? Thank you! Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 UK +44 870 - 3403539 FWD 64662 sip:ipfone@sipserver.com.br www.ipfone.com.br info@ipfone.com.br -------------- next part

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

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:

> -----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

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

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

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

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

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

--- 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

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

I have installed RedHat ES 3.0 It came with samba 3.0.0-14.3E.i386 I have no SWAT. Which one can I use? I downloaded samba-swat-3.0.0-1.test.rh90.i386.rpm and an rpm -i said it needs: samba 3.0.0-1.test.rh90 I downloaded samba-swat-2.2.7a-8.9.0.i386.rpm and an rpm -i said it needs: samba 2.2.7a Which SWAT do I need and where can I download it? thanks, in advance. Dean

> On Dec 3, 2007 5:19 AM, Miklos Szeredi <miklos at szeredi.hu> wrote: > > > Yes it is weird. I hope you can help me with this problem as sshfs is > > > so much simpler than nfs. > > > > Can you please do one more thing: start up sshfs normally, then on the > > server do a strace of the sftp-server process: > > > > strace -o /tmp/strace.log

https://bugzilla.mindrot.org/show_bug.cgi?id=1555 Summary: add hard link and attribute extensions to sftp-server Product: Portable OpenSSH Version: 5.1p1 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P2 Component: sftp-server AssignedTo: unassigned-bugs at mindrot.org

Hi, I'm having a problem rsyncing one file (since I signed it). It seems that the content of a file is able to cause problems in the protocol. building file list ... 28820 files to consider apt/packages/avifile/ apt/packages/avifile/avifile-0.7.34-1.dag.rh90.i386.rpm rsync: error writing 4 unbuffered bytes - exiting: Broken pipe rsync error: error in rsync protocol data stream (code

https://bugzilla.mindrot.org/show_bug.cgi?id=1399 Summary: add statfs extension to sftp-server Classification: Unclassified Product: Portable OpenSSH Version: -current Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: sftp-server AssignedTo: bitbucket at

In certain situations sftp download speed can be much less than that of scp. After many days of trying to find the cause finally I found it to be the tcp nagle algorithm, which if turned off with TCP_NODELAY eliminates the problem. Now I see it being discussed back in 2002, but it still unresolved in openssh-4.2 :( Simple solution would be to add a NoDelay option to ssh which sftp would set.

https://bugzilla.mindrot.org/show_bug.cgi?id=1400 Summary: add atomic rename extension to sftp-server Classification: Unclassified Product: Portable OpenSSH Version: -current Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: sftp-server AssignedTo: bitbucket at