similar to: Asterisk 1.8.14.0 Now Available

Thanks Paradise, this seems to have worked a treat!!! I commented out the: exten => 110,hint,SIP/110 lines which were in extensions_additional.conf for each sip extension I had. This seems to have stopped the crashes which were previously 3-5 times a day, now: System uptime: 1 day, 18 hours, 10 minutes, 3 seconds Interestingly it had the knock on effect of fixing another problem I had

> -----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

Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be

I have just setup asterisk on a debian sarge box. I am running Asterisk 1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz) ISDN card, connected to a BT ISDN2e line. Currently we have 6 extensions (SIP) configured all using CounterPath(Xten) eyebeam softphone. After many hours of Googling I have finally got it all setup and working. We can transfer calls internally and make and

The Asterisk Development Team has announced the third release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.0.0-rc3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the

The Asterisk Development Team has announced the third release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.0.0-rc3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

I know I'm getting tired of looking at code, but why is the function register_verify defined in 2 different files? chan_iax2.c line 3860 static int register_verify(int callno, struct sockaddr_in *sin, struct iax_ies *ies) chan_sip.c line 4869 /*--- register_verify: Verify registration of user */ static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req,

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So

Anyone know how can I fix Half-Life problem? : "Half Life need at least a 16bit color depth to run, usually this is "32768" or "65535" colors" I tried many settings in config, and it doesn't work. I have Wine 20010809 from Debian binaries ( other versions don't work too ) and X 4.1.0 with TrueColor (24bpp) color depth on Nvidia card. -- Pozdrawiam Jakub

Hello I've discovered, that sending a SIGUSR1 signal to ices causes it's shutdown when recording from an alsa dsnoopped device. > [2004-11-01 18:51:51] INFO signals/signal_usr1_handler Metadata update requested > [2004-11-01 18:51:51] DBUG metadata/metadata_thread_signal meta thread wakeup > [2004-11-01 18:51:51] DBUG stream-shared/stream_wait_for_data Shutdown signalled:

Hi there, I'll need some help with this: Trying to establish an IAX2 link between two servers works in one direction (SIP client with ulaw), but not in the other (SIP client with GSM). The client used for this is X-Lite behind NAT while both servers have a public IP (I playback an anouncement before trying to connect to the second *). Error on the originating * server:

The latest chan_sip.c works for my budgetones with the following lines removed. YMMV. I haven't bothered to dig in and see what those lines actually do. Did soneone just get wacky with cut and paste from the peer while loop? Or am I breaking something else. Jon --- chan_sip.c.broken Fri Nov 7 02:17:47 2003 +++ chan_sip.c Fri Nov 7 02:16:23 2003 @@ -3928,8 +3928,8 @@ static int

Hi, The Polycom 600 phones do not natively bridge with Asterisk. I've solved the problem, but I'm not sure how general it is, so I thought I'd ask this list for advice. It's necessary to use a recent Asterisk CVS for this, since there was a problem with session versions in earlier CVS builds. The problem now is the Via field. When the reinvite goes out, the branch number

Wow, thanks Brian! Everything I saw said the patch was only needed on older releases. I've updated several times over the last week. I patched two systems today, one 3/11/05 and one 3/17/05 and now they both work. Should have posted here sooner! Brian G. On Thu, 2005-03-17 at 13:28, Brian Buhrow wrote: > Hello. I'm writing in response to your message to the ASterisk-users >

Greetings. Attempting to connect Asterisk to LDAP database using res_config_ldap module. While trying to register sip client (Ekiga softphone), according to slapd.log, asterisk connects to LDAP server, asks for some attributes to modify (they do exist, and asterisk user has all permissions to do that, etc). And then asterisk application just crashes. Without ldap (using just static users'

Hi, A brief 6-step guide on how to hardcode a change in the Asterisk source that will allow it to work from behind a nat device. I know it?s messy, but it may prove useful to some people. 1. First punch a whole in your nat device. I just forwarded the port 5060 (for sip) and all ports between 10000 to 10020 (for rtp) to my asterisk gateway. 2. Now make sure your /etc/asterisk/rtp.conf correctly

hello, I read http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large anybody could tell me more about this ? Is it available with ARA ? Regards Harry Method 3 Q: If you have your SIP phones registered with SER but your voicemail is handled by asterisk, how do you get the MWI (Message Waiting Indicator) light to function on the phone? A: In sip.conf create a section pointing at your

The Asterisk Development Team has announced the release of Asterisk 1.8.27.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.27.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 1.8.27.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.27.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs