[asterisk-users] Voice "broken" during calls
Michael Keuter
lists at mksolutions.info
Sat Jun 13 11:06:23 CDT 2020
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)
>
> --
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Michael
http://www.mksolutions.info
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