similar to: Regular Crashes - Partially Solved

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

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

Hi, I'm cross-posting here from serverfault.com in case anyone can help. I just found a similar question on askubuntu.com also without an answer. Switched recently from W2K3 to Samba4.0.9/CentOS6.4 for our fileshare for WinXP clients. Have an ancient (1995!) piece of software that uses 8.3 filename format. After the switch, long filenames became useless in the context 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

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 release of Asterisk 1.8.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.14.0 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 issues resolved in this

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,

Just thought I'd follow up on my post from around august.... it seems that the latest kernel has indeed fixed the cool 'n quiet issues with my Athlon X2 3800. basically the following messages were appearing; powernow-k8: ignoring illegal change in lo freq table-2 to 0x2 powernow-k8: transition frequency failed and both cores were going at full 2.0ghz speed. Now with

Hi, Does the passdb backend = ldapsam:"ldap://ldap.daa.com.au ldap://yaminon.daa.com.au", smbpasswd syntax actually do proper failover? I have a samba 3.0.9 server on FC2 that's been overheating (our aircon failed), and the ldap server doesn't start automatically. The logs said: [2006/01/10 08:55:47, 0] lib/smbldap.c:smbldap_open_connection(678) Failed to issue the

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

Hi all A slightly off topic post, but one that I think someone out there in the NUT community could probably help me with. Summer is coming, and my office at home is getting hot... helped in part by the APC 3KVA UPS I have sitting under my desk. It kicks out heat like a bar-heater, even when there is almost no load on it (idle load is one PC, one switch, one wireless AP, and thats it). It is

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

Hi, My Asetrisk restarted after to output following warning message. [Oct 16 15:59:58] WARNING[17102][C-00008e34]: chan_sip.c:4696 update_provisional_keepalive: Unable to cancel schedule ID 738278. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4696). This message has been output after a timeout occurrs in the Dial() application. Then, the Hangup() application is