Re: [asterisk-users] memory issues

https://issues.asterisk.org/jira/browse/ASTERISK-28695 On Fri, Sep 25, 2020 at 2:32 PM hw wrote: > > > Hi, > > > > ever since I have switched my server from Centos 7 to Fedora 32, asterisk > > is showing memory issues and no calls are possible. I'm using the asterisk > > that comes with

Re: [asterisk-users] call an IP camera?

On Thu, 2020-09-24 at 18:45 +0200, Antony Stone wrote: > On Thursday 24 September 2020 at 18:28:13, hw wrote: > > > On Thu, 2020-09-24 at 16:57 +0200, Antony Stone wrote: > > > I would start with something like > > > https://www.voip-info.org/asterisk-config-musiconholdconf/ > > >

[asterisk-users] PJSIP - Forcing codec preference?

Hi, We're holding ourselves back from moving to PJSIP as we don't appear to have figured out how to force codec preference in a dial plan. The 'PJSIP Advanced Codec Negotiation' document (https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation) appears to ultimately be

Re: [asterisk-users] Negotiates g729 but RTP contains g711

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call

Re: [asterisk-users] Negotiates g729 but RTP contains g711

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call

[asterisk-users] memory issues

Hi, ever since I have switched my server from Centos 7 to Fedora 32, asterisk is showing memory issues and no calls are possible. I'm using the asterisk that comes with Fedora; before that, I used a self-compiled version on Centos. The hardware is still the same. Asterisk shows the following

Re: [asterisk-users] Asterisk Drop call

Thanks Luciano. But there is no active ALG on the modem. Attached the call flow, including the ACK. Em 22/09/2020 14:41, Luciano Moreira escreveu: Roberto Check your router if ALG or similar feature is enabled. Disable and test. Also, on SNGREP check if both parties are getting ACK correctly

[asterisk-users] DPMA 3.5.5 Released

Greetings, We don’t normally announce DPMA releases on here but 3.5.5 was just released which resolves a compatibility issue between the latest versions of Asterisk (using PJSIP 2.10) and DPMA. For TCP or TLS traffic it was possible for a crash to occur. It’s recommended to update to 3.5.5.

Re: [asterisk-users] Negotiates g729 but RTP contains g711

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call

[asterisk-users] Directory Application

Hello all, Anyone know an easy way to have the Directory Application lookup all the voicemail contexts in the system. Like a global option John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: