So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <[email protected]>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name : 0049177xxxxxxx > > > Description : > > > Secret : <Set> > > > MD5Secret : <Not set> > > > Remote Secret: <Not set> > > > Context : default > > > Record On feature : automon > > > Record Off feature : automon > > > Subscr.Cont. : <Not set> > > > Language : de > > > Tonezone : <Not set> > AMA flags : Unknown > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup : 1 > Pickupgroup : 1 > Named Callgr : > Nam. Pickupgr: > MOH Suggest : > Mailbox : > VM Extension : asterisk > LastMsgsSent : 0/0 > Call limit : 2147483647 > Max forwards : 0 > Dynamic : Yes > Callerid : "0049177xxxxxxx" <> > MaxCallBR : 384 kbps > Expire : -1 > Insecure : no > Force rport : Yes > Symmetric RTP: Yes > ACL : No > DirectMedACL : No > T.38 support : Yes > T.38 EC mode : FEC > T.38 MaxDtgrm: 4294967295 > DirectMedia : No > PromiscRedir : No > User=Phone : No > Video Support: No > Text Support : No > Ign SDP ver : No > Trust RPID : No > Send RPID : Yes > Path support : No > Path : N/A > TrustIDOutbnd: Legacy > Subscriptions: Yes > Overlap dial : No > DTMFmode : rfc2833 > Timer T1 : 500 > Timer B : 32000 > ToHost : > Addr->IP : (null) > Defaddr->IP : (null) > Prim.Transp. : UDP > Allowed.Trsp : UDP > Def. Username: > SIP Options : (none) > Codecs : > (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk) > Auto-Framing : No > Status : UNKNOWN > Useragent : > Reg. Contact : > Qualify Freq : 60000 ms > Keepalive : 0 ms > Sess-Timers : Refuse > Sess-Refresh : uac > Sess-Expires : 1800 secs > Min-Sess : 90 secs > RTP Engine : asterisk > Parkinglot : > Use Reason : No > Encryption : No > > VoIP-phone (Thomson ST2022): > bpi*CLI> sip show peer 0049351xxxxxxx > > > > > * Name : 0049351xxxxxxx > > > Description : > > > Secret : <Set> > > > MD5Secret : <Not set> > > > Remote Secret: <Not set> > > > Context : default > > > Record On feature : automon > > > Record Off feature : automon > Subscr.Cont. : <Not set> > Language : de > Tonezone : <Not set> > AMA flags : Unknown > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup : 1 > Pickupgroup : 1 > Named Callgr : > Nam. Pickupgr: > MOH Suggest : > Mailbox : > VM Extension : asterisk > LastMsgsSent : 0/0 > Call limit : 2147483647 > Max forwards : 0 > Dynamic : Yes > Callerid : "0049351xxxxxxx" <> > MaxCallBR : 384 kbps > Expire : 3111 > Insecure : no > Force rport : Yes > Symmetric RTP: Yes > ACL : Yes > DirectMedACL : No > T.38 support : Yes > T.38 EC mode : FEC > T.38 MaxDtgrm: 4294967295 > DirectMedia : No > PromiscRedir : No > User=Phone : No > Video Support: No > Text Support : No > Ign SDP ver : No > Trust RPID : No > Send RPID : Yes > Path support : No > Path : N/A > TrustIDOutbnd: Legacy > Subscriptions: Yes > Overlap dial : No > DTMFmode : rfc2833 > Timer T1 : 500 > Timer B : 32000 > ToHost : > Addr->IP : 192.168.200.10:25572 > Defaddr->IP : (null) > Prim.Transp. : UDP > Allowed.Trsp : UDP > Def. Username: 0049351xxxxxxx > SIP Options : (none) > Codecs : (alaw|ulaw|ilbc|g729|g723|gsm) > Auto-Framing : No > Status : OK (17 ms) > Useragent : THOMSON ST2022 hw2 fw3.56 00-26-44-31-10-23 > Reg. Contact : sip:[email protected]:25572;user=phone > Qualify Freq : 60000 ms > Keepalive : 0 ms > Sess-Timers : Refuse > Sess-Refresh : uac > Sess-Expires : 1800 secs > Min-Sess : 90 secs > RTP Engine : asterisk > Parkinglot : > Use Reason : No > Encryption : No > > >> Then "sip show channels" during an existing call. > > Call from normal phone: > bpi*CLI> sip show channels > Peer User/ANR Call ID Format Hold > Last Message Expiry Peer > 192.168.200.10 0049351xxxxxxx 9eff88f7-c0a801 (alaw) No > Rx: ACK 0049351xxxxxxx > 217.0.27.53 03501xxxxxxx 453efbcb7a04f33 (alaw) No > Tx: ACK pbxluca > 2 active SIP dialogs > > Call from mobile phone (via VoIP registered in Asterisk): > > bpi*CLI> sip show channels > Peer User/ANR Call ID Format Hold > Last Message Expiry Peer > 192.168.10.12 0049177xxxxxxx 11b86bd612b71ae (alaw) No > Rx: INVITE 0049177xxxxxxx > 217.0.27.53 00493501xxxxxxx 5647efe41d746b4 (alaw) No > Tx: INVITE pbxluca > 2 active SIP dialogs > > >> And "sip show channel <Call-ID>" for more info. > > Call from normal phone: > > bpi*CLI> sip show channel [email protected] > > * SIP Call > > > Curr. trans. direction: Incoming > > > Call-ID: [email protected] > > > Owner channel ID: SIP/0049351xxxxxxx-000000a7 > Our Codec Capability: (alaw|ulaw|ilbc|g729|g723|gsm) > > > Non-Codec Capability (DTMF): 1 > > > Their Codec Capability: (ulaw|g723|alaw|g729) > > > Joint Codec Capability: (alaw|ulaw|g729|g723) > Format: (alaw) > T.38 support No > Video support No > MaxCallBR: 384 kbps > Theoretical Address: 192.168.200.10:25572 > Received Address: 192.168.200.10:25572 > SIP Transfer mode: open > Force rport: Yes > Audio IP: 192.168.200.1 (local) > Our Tag: as12e44b1b > Their Tag: c0a80101-d3c8cef7 > SIP User agent: THOMSON ST2022 hw2 fw3.56 00-26-44-31-10-23 > Username: 0049351xxxxxxx > Peername: 0049351xxxxxxx > Original uri: sip:[email protected]:25572 > Caller-ID: 0049351xxxxxxx > Need Destroy: No > Last Message: Rx: ACK > Promiscuous Redir: No > Route: > <sip:[email protected]:25572;user=phone> > DTMF Mode: rfc2833 > SIP Options: replaces replace timer > Session-Timer: Inactive > Transport: UDP > Media: RTP > > bpi*CLI> sip show channel [email protected] > > * SIP Call > Curr. trans. direction: Outgoing > Call-ID: [email protected] > Owner channel ID: SIP/pbxluca-000000a8 > Our Codec Capability: (alaw|ulaw) > Non-Codec Capability (DTMF): 1 > Their Codec Capability: (alaw) > Joint Codec Capability: (alaw) > Format: (alaw) > T.38 support Yes > Video support No > MaxCallBR: 384 kbps > Theoretical Address: 217.0.27.xx:5060 > Received Address: 217.0.27.xx:5060 > SIP Transfer mode: open > Force rport: Yes > Audio IP: 91.49.50.x (local) > Our Tag: as29bbbfb6 > Their Tag: > h7g4Esbg_p65551t1592060241m195254c7230720s1_1763914935-920913141 > SIP User agent: > Username: 03501xxxxxxx > Peername: pbxluca > Original uri: sip:[email protected] > Need Destroy: No > Last Message: Tx: ACK > Promiscuous Redir: No > Route: <sip:217.0.27.xx;transport=udp;lr> > DTMF Mode: rfc2833 > SIP Options: (none) > Session-Timer: Inactive > Transport: UDP > Media: RTP > > Call from mobile phone (via VoIP registered in Asterisk): > > bpi*CLI> sip show channel [email protected] > > * SIP Call > Curr. trans. direction: Incoming > Call-ID: [email protected] > Owner channel ID: SIP/0049177xxxxxxx-000000a9 > Our Codec Capability: > (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk) > Non-Codec Capability (DTMF): 1 > Their Codec Capability: (ulaw|gsm|alaw|amr) > Joint Codec Capability: (alaw|ulaw|gsm|amr) > Format: (alaw) > T.38 support No > Video support No > MaxCallBR: 384 kbps > Theoretical Address: 192.168.10.12:37210 > Received Address: 192.168.10.12:37210 > SIP Transfer mode: open > Force rport: Yes > Audio IP: 192.168.10.1 (local) > Our Tag: as339b5367 > Their Tag: 1910565801 > SIP User agent: > Peername: 0049177xxxxxxx > Original uri: sip:[email protected]:37210 > Caller-ID: 0049177xxxxxxx > Need Destroy: No > Last Message: Rx: ACK > Promiscuous Redir: No > Route: > <sip:[email protected]:37210;transport=udp> > DTMF Mode: rfc2833 > SIP Options: (none) > Session-Timer: Inactive > Transport: UDP > Media: RTP > > > bpi*CLI> sip show channel [email protected] > > * SIP Call > Curr. trans. direction: Outgoing > Call-ID: [email protected] > Owner channel ID: SIP/pbxluca-000000aa > Our Codec Capability: (alaw|ulaw) > Non-Codec Capability (DTMF): 1 > Their Codec Capability: (alaw) > Joint Codec Capability: (alaw) > Format: (alaw) > T.38 support Yes > Video support No > MaxCallBR: 384 kbps > Theoretical Address: 217.0.27.xx:5060 > Received Address: 217.0.27.xx:5060 > SIP Transfer mode: open > Force rport: Yes > Audio IP: 91.49.50.xx (local) > Our Tag: as148b6300 > Their Tag: > h7g4Esbg_p65551t1592060364m136229c7238384s1_1886856096-203650581 > SIP User agent: > Username: 00493501xxxxxxx > Peername: pbxluca > Original uri: sip:[email protected] > Need Destroy: No > Last Message: Tx: ACK > Promiscuous Redir: No > Route: <sip:217.0.27.xx;transport=udp;lr> > DTMF Mode: rfc2833 > SIP Options: (none) > Session-Timer: Inactive > Transport: UDP > Media: RTP > > So, I'd say, the codecs are the same... > Do you see something strange that I should check/change? > > Thank you very very much for your help! > 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
Michael http://www.mksolutions.info -- _____________________________________________________________________ -- 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
