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:0049351xxxxxxx@192.168.200.10:25572;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 9eff88f7-c0a80101-0-22c911@192.168.200.10 * SIP Call Curr. trans. direction: Incoming Call-ID: 9eff88f7-c0a80101-0-22c911@192.168.200.10 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:0049351xxxxxxx@192.168.200.10:25572 Caller-ID: 0049351xxxxxxx Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: <sip:0049351xxxxxxx@192.168.200.10:25572;user=phone> DTMF Mode: rfc2833 SIP Options: replaces replace timer Session-Timer: Inactive Transport: UDP Media: RTP bpi*CLI> sip show channel 453efbcb7a04f33e1e0de7ef461f9...@tel.t-online.de * SIP Call Curr. trans. direction: Outgoing Call-ID: 453efbcb7a04f33e1e0de7ef461f9...@tel.t-online.de 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:sg...@217.0.27.xx 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 11b86bd612b71ae0f06c62d53ecf08c6@192.168.10.12 * SIP Call Curr. trans. direction: Incoming Call-ID: 11b86bd612b71ae0f06c62d53ecf08c6@192.168.10.12 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:0049177xxxxxxx@192.168.10.12:37210 Caller-ID: 0049177xxxxxxx Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: <sip:0049177xxxxxxx@192.168.10.12:37210;transport=udp> DTMF Mode: rfc2833 SIP Options: (none) Session-Timer: Inactive Transport: UDP Media: RTP bpi*CLI> sip show channel 5647efe41d746b4d67ad5c576b67b...@tel.t-online.de * SIP Call Curr. trans. direction: Outgoing Call-ID: 5647efe41d746b4d67ad5c576b67b...@tel.t-online.de 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:sg...@217.0.27.xx 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 (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