Am 03.12.2019 um 19:57 schrieb Antony Stone: Hi Antony,
thank you for your answer. > I would firstly look at whether your Asterisk box is doing transcoding - > converting from oe codec (supported by your phones) and another codec > (supported by the provider) because no codec can be found in common between > the two. > > Secondly I would put a full packet sniffer (by which I mean collect all the > RTP > data as well as SIP) on each of your interfaces (internal and external) to > see > whether the delay really is happening inside your Asterisk server - if you > see > RTP data on your internal interface, then appearing 1-1.5 seconds later on > the > external interface, and vice versa, then you know the delay is inside your > system. I'm really not an expert on Asterisk... Could you please say me HOW can I check the codecs? I tried to get the information of the channel: bpi*CLI> sip show channel p65551t1575398506m6025c4749452s2 * SIP Call Curr. trans. direction: Incoming Call-ID: p65551t1575398506m6025c4749452s2 Owner channel ID: SIP/pbxanika-0000021e Our Codec Capability: (alaw) Non-Codec Capability (DTMF): 1 Their Codec Capability: (ulaw|gsm|alaw|amr) Joint Codec Capability: (alaw) Format: (alaw) T.38 support No Video support No MaxCallBR: 384 kbps Theoretical Address: 217.x.x.x:5060 Received Address: 217.x.x.x:5060 SIP Transfer mode: open Force rport: Auto (No) Audio IP: 217.y.y.y (local) Our Tag: as45e11359 Their Tag: h7g4Esbg_p65551t1575398506m6025c4749452s1_206873930-910452977 SIP User agent: Username: 550293777072-0001 Peername: pbxanika Original uri: sip:[email protected] Caller-ID: +4917711111111 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: <sip:217.x.y.z;transport=udp;lr> DTMF Mode: rfc2833 SIP Options: timer Session-Timer: Inactive Transport: UDP Media: RTP Maybe it helps to find the problem? Thanks Luca Bertoncello ([email protected]) -- _____________________________________________________________________ -- 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
