[asterisk-users] Voice "broken" during calls

Luca Bertoncello lucabert at lucabert.de
Sat Jun 13 10:23:14 CDT 2020


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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