Re: [asterisk-users] RTP not being switched between both SIP endpoints

2013-09-18 Thread Kenny Watson
Hi, Since opensips is not handling media (i presume) is the audio not already going direct from asterisk to the other endpoint? Thanks Kenny From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Gareth

Re: [asterisk-users] Voiceprompts i.e. voicemail and conferencing in multiple codecs

2010-07-06 Thread Kenny Watson
codecs On Fri, 2 Jul 2010, Kenny Watson wrote: for i in `ls -R /var/lib/asterisk/sounds/uk/*wav`; # do recursive ls and only list wav files and loop through each one do # start do loop CONV=`echo $i|sed 's/.wav/.g729/g'` # set CONV variable as filename with wav swapped for G729

Re: [asterisk-users] Voiceprompts i.e. voicemail and conferencing in multiple codecs

2010-07-02 Thread Kenny Watson
in multiple codecs On Tue, Jun 29, 2010 at 12:51 PM, Kenny Watson kwat...@geniusgroupltd.com wrote: Is it simply a case of converting the prompts into other codecs and asterisk will pick these up? Yes, install both g729 and ulaw/alaw prompts to avoid trans-coding altogether. -- Paul Belanger | dCAP

[asterisk-users] Adding Congestion to CDR logs

2010-06-30 Thread Kenny Watson
that this is possible! Thanks Kenny Watson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

Re: [asterisk-users] Adding Congestion to CDR logs

2010-06-30 Thread Kenny Watson
Discussion asterisk-users@lists.digium.com Sent: Wednesday, 30 June, 2010 11:44:58 AM Subject: Re: [asterisk-users] Adding Congestion to CDR logs Kenny Watson wrote: Hi, I had a breif telco outage with one of my sip providers. Is there a way to add failed calls to the cdr aswell

Re: [asterisk-users] Adding Congestion to CDR logs

2010-06-30 Thread Kenny Watson
the weighting or even the shift between good and bad! The db would be held on a separate server, all I want this box to do is handle a tonne of calls and do some transcoding on harder (I have recently bought a howler screamer card). Thanks Kenny Watson - Original Message - From: Gareth

Re: [asterisk-users] Dial options not working

2010-06-30 Thread Kenny Watson
Hi, Have you tried sending the dtmf inband? I've had more success interoping betwen different vendors with inband DTMF. Thanks Kenny Watson Kenny Watson From: Anahi LudueƱa a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Sent: Wednesday, 30 June, 2010 12:50:23

Re: [asterisk-users] Find a way to block brute force attacks.

2010-06-29 Thread Kenny Watson
Hi, you can use fail2ban http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk Which works well, when a pattern is found in a log file it addes in an iptables rules to block the traffic for a period. On debian you can apt-get install fail2ban and on centos/redhat yum -i

[asterisk-users] Voiceprompts i.e. voicemail and conferencing in multiple codecs

2010-06-29 Thread Kenny Watson
Hi, I am running asterisk 1.6.1.6 with a howler screamer card. I have g729 and alaw trunks from a pbx /sip providers. The howler screamer will only transcode from g729 to alaw / ulaw but my voice prompts are in SLIN and throws errors when i try and access these applications. Is it