[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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