Archives : January-2014
peopleIm just mailing to see what people are using for CTI solutions with asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce?Thanks in ..
On a Asterisk 1.8.12 system working OK for months (>100k calls proceed), users are complaining for bad audio.My setup is:PSTNAsteriskSiemensHiPathPSTNasterisk -rx dahdi show versionDAHDI Version: SVN-trunk-r10414 Echo Canceller: HWECasterisk -rx ..
http://stackoverflow.com/questions/21015596/failed-to-get-160-samples-from-read-factory-asterisk-11-5-1-app-co..
i noticed in asterisk 10.12.3, i get messages like this:[2014-01-08 19:03:59] NOTICE[2987]: chan_sip.c:23900 handle_request_invite:Failed to authenticate device 305;tag
516e63but not mentioning attacker ip (to be used for fail2ban)is this..
I recently purchased the Cepstral 6 text-to-speech engine (swift), and am now wondering if I should have bought something else.I would like to use Cepstral text to speech like some people use the Festival() or Flite() applications.For example, whe..
allSorry for null subject last mail.I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it Asterisk11. I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Lo..
I see the strange behaviour that outgoing calls end after 15 minutes.I didnt knew there is some kind of call duration limit that can be set ?Is there ?Using Asterisk 1.8.12.2Kind rega..
allSorry that forgot add mail subject last one.I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it Asterisk11. I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial ..
The issue you reference shows that the patch was committed to the v1.6.2branch with SVN revision -r261498. The commit message mentions the patch file but it does not appear to be attached to the issue. You can use SVN to generate a diff file of the commit..
Im asking about this scenario:Asterisk(public IP)InternetRouter (public IP)SIP client (private IP and NAT)What settings in sip.conf will give this the best fighting chance of working?We already have nat=force_rpo..