similar to: IPSec on asterisk

Hi, We have free pbx installed on asterisk 1.4.25.1. Mysql is installed and asterisk is connecting to it. CDR modules are all loaded as well. For some reason, it is not creating master.csv and no cdrs are generated. Can anyone help please. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

Hi, On some SIP interconnects with devices like Cisco, Dialogic we get SIP invite from different source port every time and asterisk rejects that INVITE. Does anyone knows solution for this? --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijhawan at oxygen8.com Skype:

Hi, Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/6114bf1d/attachment.htm

Hi, I am doing failover routing based on 2 dial commands. First route sends back 4xx response and I don't want it to try 2nd route when it is 4xx response. Can we do failover routing based on SIP 5xx response only ? Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it's busy or unallocated number or failed with some other reason. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi status is not showing alarms but channels are not coming up. It is not showing any channels when i run 'dahdi show channels'. Could anyone help pls. Thanks Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an incomming call but when I receive a call by a 2nd line in my softphone, lost the first call. Sometimes the first call is dropped, and sometimes the call is active, but I can't hear the caller. It's an asterisk Bug? I have asterisk 1.4.22. Please help!!! Thanks -- Carem Gyssell Nieto Garcia -------------- next part -------------- An HTML attachment was

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

hi, all i want to use g729 codec for set up a call. so i donwloaded the so file from web site: http://asterisk.hosting.lv/#bin and install it properly. *CLI> *CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin

Hi, Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about half an hour ago. IAX Friends (Zoiper Softphones) don't stay registered for more than a few seconds they start out with status unknown and quickly become unreachable, I am using realtime with postgresql, with tables and configuration that have worked fine for IAX in 1.6 and a trunk release from a few months

Hi all, We have an asterisk version v1.6.1.20 with a TDM400 board (2 FXS and 2 FXO). We want to do a transfer "blind" and "attended" from a line external connected to one FXO. We have made configuration, and transfers from internal lines (FXS) work fine but from (FXO) not. We have made 2 test, one work fine from FXS and the other form FXO no. Test 1, work fine: 1) A

SIP user => Asterisk 1.6 server => SIP Trunk => external destination: works AMI script => Asterisk 1.6 server => SIP Trunk => external destination: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@(ipaddr)>;tag=alphanumeric' I?ve tried doing things like ?include => contextwithtrunk" in various places, googling, re-reading relevant

Hi I hope someone can answer something I'm sure is quite basic. I am following the instructions at https://www.centos.org/docs/5/html/Deployment_Guide-en-US/ch-vpn.html On setting up a VPN The part I am having trouble with is when it show the /etc/racoon/racoon.conf file. But it doesn't say whay you have to do with this file. When I bring up my connection ifup bicester I get RTNETLINK

I've been using IPSec and racoon alot lately creating tunnels between FreeBSD machines. Everything works as it should once I've got it running. I do however seem to get delays when one, or both ends of the tunnel drop or are rebooted. On reboot, once the machine starts racoon, it takes two or three minutes for the tunnel to come back up. If I stop and restart racoon, it takes only 60

Centos 5 is still soon end of life. Using it as ipsec gateway is .. Eero 21.3.2016 7.25 ip. "Mike - st257" <silvertip257 at gmail.com> kirjoitti: > On Mon, Mar 21, 2016 at 1:17 PM, Mike - st257 <silvertip257 at gmail.com> > wrote: > > > I second Eero's comment, use a new IPSec daemon. > > > > Openswan was forked and became Libreswan. Paul, now

I second Eero's comment, use a new IPSec daemon. Openswan was forked and became Libreswan. Paul, now a RH employee, was a main developer for the Openswan project before he and others created the Libreswan fork. https://libreswan.org/ EL6 has Openswan EL7 has Libreswan Racoon isn't all that fun to work with. If you have the option, ditch it and EL5 and move to a newer platform

Err. Sounds like security nightmare. 21.3.2016 7.47 ip. "Glenn Pierce" <glennpierce at gmail.com> kirjoitti: > Will ask my boss :) We are hosted on memset so not so easy to update > > Thanks > > On 21 March 2016 at 17:36, Eero Volotinen <eero.volotinen at iki.fi> wrote: > > Centos 5 is still soon end of life. Using it as ipsec gateway is .. > >

Hi Running racoon on a Freebsd-4.11 machine gives a kernel panic. I am using the racoon from ports directory which comes with the freebsd installation. Steps followed are as shown below: racoon -f /usr/local/etc/racoon/raccon.conf setkey -f ipsec.conf ping -c 1 <ip_of_the_other_gw> The ping will lead into a crash. The crash dump looks like for th ping packet it