[asterisk-users] DTMF not working on incoming calls
What is the best way to debug DTMF on a PJSIP trunk? I have
cycled through all available options
('rfc4733','inband','info','auto','auto_info') but my IVR does not
recognize any options from the remote end. I have also tried changing
codecs from g729 to alaw or ulaw with the same result. Outgoing calls
do not seem to have this problem, just incoming. This is with Asterisk
13.29.2 but the problem started with 13.21 before I decided to upgrade
to the latest 13.x version. Any pointers?
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[asterisk-users] no video when dialing between extension
hi all im trying to call a door phone supporting video i hear the audio but dont get video i see this in the log why should it try to translate? Unable to find a codec translation path: (h264) -> (opus) asterisk version 13.26 thanks for any help -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
Am 04.12.2019 um 11:14 schrieb Antony Stone: > Hm, I was judging based on what you posted previously: > > Our Codec Capability: (alaw) > Their Codec Capability: (ulaw|gsm|alaw|amr) > Joint Codec Capability: (alaw) > > which suggested to me that if you offered GSM, that could be agreed with the > other side. I tried again right now: disallow=all allow=alaw allow=gsm If I call my phone I can see, alaw is used. If I allow just gsm I get the error: [Dec 4 11:23:17] NOTICE[14060][C-012e]: chan_sip.c:10798 process_sdp: No compatible codecs, not accepting this offer! So, back to alaw... :( > Ah, but SIP is not RTP :) OK, I forgot it... I privilege RTP, too... ;) Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
On Wednesday 04 December 2019 at 11:00:23, Luca Bertoncello wrote: > Am 04.12.2019 um 10:53 schrieb Antony Stone: > > Hi Antony! > > > 1. Try using codec GSM (which is pretty good quality but lower bandwidth > > than alaw, which is currently the only one you are offering). > > gsm seems to be unsupported from Deutsche Telekom... > Already tried, it does not work... :( Hm, I was judging based on what you posted previously: Our Codec Capability: (alaw) Their Codec Capability: (ulaw|gsm|alaw|amr) Joint Codec Capability: (alaw) which suggested to me that if you offered GSM, that could be agreed with the other side. > > 2. What is the bandwidth (upstream is more important than downstream) of > > your Internet connection? > > Down 50Mbps > Up 10Mbps Well, that should certainly be plenty for a single VoIP channel (which I usually estimate as 100kpbs each way for ulaw or alaw). > On my Router (Debian 9) I configured a traffic shaper that privileges > the SIP-Packets. Ah, but SIP is not RTP :) SIP is used to set up the call, tell the other end what number you want to dial, tell you that the phone needs to ring, etc. It's not the audio part of the call once it's set up. RTP is the audio part of the call, and that's what you're saying is not so good now you've disabled the jitter buffer. RTP is UDP packets normally sent on any port between 10,000 and 20,000, so you need to ensure that your router allows that through with as low latency (and very importantly, consistent latency, since inconsistent latency = jitter) as possible. Prioritising SIP is hardly ever needed - who cares about a few tenths of a second setting up or responding to a call? What needs prioritising, and QoS if you can do it, is RTP. Antony. -- Police have found a cartoonist dead in his house. They say that details are currently sketchy. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
Am 04.12.2019 um 10:53 schrieb Antony Stone: Hi Antony! > 1. Try using codec GSM (which is pretty good quality but lower bandwidth than > alaw, which is currently the only one you are offering). gsm seems to be unsupported from Deutsche Telekom... Already tried, it does not work... :( > 2. What is the bandwidth (upstream is more important than downstream) of your > Internet connection? Down 50Mbps Up 10Mbps On my Router (Debian 9) I configured a traffic shaper that privileges the SIP-Packets. Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
On Wednesday 04 December 2019 at 07:37:51, Luca Bertoncello wrote: > Am 03.12.2019 um 19:28 schrieb Luca Bertoncello: > > Hi again > > > This delay happens on every peer, Deutsche Telekom and Messagenet, so I > > think the problem is NOT by the Provider, but in my configuration... > > Maybe I got the solution... > I see, that I had the jitter buffer active. As I deactivated it, I have > no delay anymore. > Unfortunately is the audio quality now a little bad than with the jitter > buffer... > > Any suggestion how can I improve the audio quality without add the delay? 1. Try using codec GSM (which is pretty good quality but lower bandwidth than alaw, which is currently the only one you are offering). 2. What is the bandwidth (upstream is more important than downstream) of your Internet connection? Antony. -- Why is "dylexia" so difficult to spell, and why can I never remember "aphasia" when I want to? Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
