Re: [asterisk-dev] Asterisk crash
On Mon, Jun 14, 2021 at 04:49:12PM +, Dan Cropp wrote: > Any suggestions of what I could try? Please read the wiki page "Getting a Backtrace". https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Advanced Codec Negotiation: Should we allow an endpoint with no codecs configured?
On Wed, Jul 08, 2020 at 02:01:18PM -0600, George Joseph wrote: > Today, if you create an endpoint configuration with no codecs, we create an > empty topology for it but otherwise accept it. I'm not sure what you'd use > it for though. Is there a situation where an endpoint with no codecs > _should_ be allowed or can we start rejecting those endpoint configurations? I have never used such a configuration. If the endpoint is only used for out of call messages this should be possible. But if the endpoint is not used for voip calls a list of codes is not a problem. -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Advanced Codec Negotiation: Need info and uses cases
On Sat, Jun 06, 2020 at 08:20:47AM +0200, Michael Maier wrote: > Transcoding only if there is an allowed and supported codec on both sides > which are not common. If there is one common codec on both sides - use > this codec and do not transcode. If Alice (OpenStage20, Local Net) offers G.722, alaw and ulaw and Bob (Remote Localtion) supports Opus, alaw and ulaw, then transcoding between G.722 and Opus results in better audio quality. Transcoding causes some CPU Load, but this might be acceptable. Kind regards -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Advice for having a bug fixed
On Thu, Jul 21, 2016 at 11:06:50PM +0200, Leandro Dardini wrote: > About replicating the problem at will, it is not easy ... I have some > servers where the locking is quite immediate, others running for days > without any problem. Could issue 25468 be related to 25388 ? The attached tar archive contains a simple configuration, which caused deadlocks on one server. Unfortunately I did not manage to reproduce the problem on other systems. Is the locking mechanism hardware dependend? Something not available on another server and another locking mechanism is used? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Re: sms sending and receiving
On Sun, Jun 27, 2004 at 11:10:56PM +0200, Hans Mueller wrote: > Hi Andeas, > > Andreas Bayer wrote: > >I heard, that sms over isdn (compatible with protocol used by "Deutsche > >Telekom" ) work. > > > IMHO there are two completely different SMS Protocoll: The first works on > any analog line. This is a known ETSI standard and is supported by app_sms. > > The second one, which Swisscom Fixnet uses (i guess this is the same stuff > as Deutsche Telekom), works ONLY on BRI. The Protocoll itself is very app_sms is adequate to send and receive short messages if you are a Deutsche Telekom customer. It is the same protocol BT uses (ETSI ES 201 912). Send outgoing messages to 0193010. Incoming messages can be identified by the callerId 01930100 if they are send by fixed line or D1 phones. Some information can be found here (German Language) http://www.telekom.de/dtag/downloads/S/Schnitt_1TR140_V2.1.pdf Sending messages is well documented at voip-info. For incoming messages I had to use something like this: Answer Wait,2 SMS(default,a) Hangup -- Stefan Tichy <[EMAIL PROTECTED]> ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
