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 at 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 at 192.168.200.10 * SIP Call Curr. trans. direction: Incoming Call-ID: 9eff88f7-c0a80101-0-22c911 at 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 at 192.168.200.10:25572 Caller-ID: 0049351xxxxxxx Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: <sip:0049351xxxxxxx at 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 453efbcb7a04f33e1e0de7ef461f9b38 at tel.t-online.de * SIP Call Curr. trans. direction: Outgoing Call-ID: 453efbcb7a04f33e1e0de7ef461f9b38 at 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:sgc_c at 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 at 192.168.10.12 * SIP Call Curr. trans. direction: Incoming Call-ID: 11b86bd612b71ae0f06c62d53ecf08c6 at 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 at 192.168.10.12:37210 Caller-ID: 0049177xxxxxxx Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: <sip:0049177xxxxxxx at 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 5647efe41d746b4d67ad5c576b67beba at tel.t-online.de * SIP Call Curr. trans. direction: Outgoing Call-ID: 5647efe41d746b4d67ad5c576b67beba at 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:sgc_c at 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 (lucabert at lucabert.de)
So the call used Alaw as Codec.> Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > 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 at 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 at 192.168.200.10 > > * SIP Call > > > Curr. trans. direction: Incoming > > > Call-ID: 9eff88f7-c0a80101-0-22c911 at 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 at 192.168.200.10:25572 > Caller-ID: 0049351xxxxxxx > Need Destroy: No > Last Message: Rx: ACK > Promiscuous Redir: No > Route: > <sip:0049351xxxxxxx at 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 453efbcb7a04f33e1e0de7ef461f9b38 at tel.t-online.de > > * SIP Call > Curr. trans. direction: Outgoing > Call-ID: 453efbcb7a04f33e1e0de7ef461f9b38 at 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:sgc_c at 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 at 192.168.10.12 > > * SIP Call > Curr. trans. direction: Incoming > Call-ID: 11b86bd612b71ae0f06c62d53ecf08c6 at 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 at 192.168.10.12:37210 > Caller-ID: 0049177xxxxxxx > Need Destroy: No > Last Message: Rx: ACK > Promiscuous Redir: No > Route: > <sip:0049177xxxxxxx at 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 5647efe41d746b4d67ad5c576b67beba at tel.t-online.de > > * SIP Call > Curr. trans. direction: Outgoing > Call-ID: 5647efe41d746b4d67ad5c576b67beba at 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:sgc_c at 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 > (lucabert at 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-usersMichael http://www.mksolutions.info
On Saturday 13 June 2020 at 18:06:23, Michael Keuter wrote:> So the call used Alaw as Codec....which should be excellent quality. PS: Michael: thanks for the tips regarding "sip show channels" and "sip show channel <ID>" - I was aware of these for some details, but hadn't realised they showed the codecs as well. Very useful :) Antony. -- Your work is both good and original. Unfortunately the parts that are good aren't original, and the parts that are original aren't good. - Samuel Johnson Please reply to the list; please *don't* CC me.
On Saturday 13 June 2020 at 17:23:14, Luca Bertoncello wrote:> Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > > Try "sip show peer <peername>" for a phone.> bpi*CLI> sip show peer 0049177xxxxxxx > 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|t > estlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk > |silk|silk)That strikes me as somewhat unlikely.> bpi*CLI> sip show peer 0049351xxxxxxx > Codecs : (alaw|ulaw|ilbc|g729|g723|gsm)That looks a little more standard. Regards, Antony. -- I just got a new mobile phone, and I called it Titanic. It's already syncing. Please reply to the list; please *don't* CC me.
Am 13.06.2020 um 18:20 schrieb Antony Stone: Hi>> bpi*CLI> sip show peer 0049177xxxxxxx >> 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|t >> estlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk >> |silk|silk) > > That strikes me as somewhat unlikely.Too much things, isn't it?>> bpi*CLI> sip show peer 0049351xxxxxxx >> Codecs : (alaw|ulaw|ilbc|g729|g723|gsm) > > That looks a little more standard.The questions are: 1) why the mobile phone, with "too many things" has a better quality 2) where can I change these settings? Thanks Luca Bertoncello (lucabert at lucabert.de)
Am 13.06.2020 um 18:06 schrieb Michael Keuter:> So the call used Alaw as Codec.Yes, so seems it to be... It should has the better quality... But the calls done using my mobile phone in VoIP with the Asterisk have better quality as the calls done using the normal VoIP-telefon... I'm really puzzled... Luca Bertoncello (lucabert at lucabert.de)