[asterisk-bugs] [JIRA] Commented: (ASTERISK-20333) 'directmedia' not working any more
Maciej Krajewski (JIRA)
noreply at issues.asterisk.org
Mon Oct 1 10:09:27 CDT 2012
[ https://issues.asterisk.org/jira/browse/ASTERISK-20333?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=197784#comment-197784 ]
Maciej Krajewski commented on ASTERISK-20333:
---------------------------------------------
test001
{code}sip show peer test001
* Name : test001
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : asterisk
Subscr.Cont. : asterisk
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox : 1234 at default,1233 at default
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "Jane Smith" <5678>
MaxCallBR : 1024 kbps
Expire : 23
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : Yes
T.38 EC mode : FEC
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 10.0.0.2:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: test001
SIP Options : (none)
Codecs : 0xe (gsm|ulaw|alaw)
Codec Order : (gsm:20,ulaw:20,alaw:20)
Auto-Framing : No
Status : Unmonitored
Useragent : Zoiper rev.11137
Reg. Contact : sip:test001 at 10.0.0.2:5060;rinstance=838af85f46405011;transport=UDP
Qualify Freq : 40000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
{code}
test 002
{code}
sip show peer test002
* Name : test002
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : asterisk
Subscr.Cont. : asterisk
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox : 1234 at default,1233 at default
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "Jane Smith" <5678>
MaxCallBR : 1024 kbps
Expire : 55
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : Yes
T.38 EC mode : FEC
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 172.16.0.84:5062
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: test002
SIP Options : (none)
Codecs : 0xe (gsm|ulaw|alaw)
Codec Order : (gsm:20,ulaw:20,alaw:20)
Auto-Framing : No
Status : Unmonitored
Useragent : Linksys/SPA962-6.1.5(a)
Reg. Contact : sip:test002 at 172.16.0.84:5062
Qualify Freq : 40000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
{code}
one channel
{code}
sip show channel 392ebbf11217952f067c18f44c0abd8f at 10.0.4.63:5060
* SIP Call
Curr. trans. direction: Outgoing
Call-ID: 392ebbf11217952f067c18f44c0abd8f at 10.0.4.63:5060
Owner channel ID: SIP/test002-00000005
Our Codec Capability: 0xe (gsm|ulaw|alaw)
Non-Codec Capability (DTMF): 1
Their Codec Capability: 0x4 (ulaw)
Joint Codec Capability: 0x4 (ulaw)
Format: 0x4 (ulaw)
T.38 support No
Video support No
MaxCallBR: 1024 kbps
Theoretical Address: 172.16.0.84:5062
Received Address: 172.16.0.84:5062
SIP Transfer mode: open
Force rport: Yes
Audio IP: 10.0.4.63 (local)
Our Tag: as1772de95
Their Tag: 39f7c1126c2d0a2ei2
SIP User agent:
Username: test002
Peername: test002
Original uri: sip:test002 at 172.16.0.84:5062
Caller-ID: 5678
Need Destroy: No
Last Message: Tx: ACK
Promiscuous Redir: No
Route: sip:test002 at 172.16.0.84:5062
DTMF Mode: rfc2833
SIP Options: (none)
Session-Timer: Inactive
{code}
and the second
{code}
sip show channel ZDFmODRiZGIyYzg2MWJlM2IzZDczNGZkNmVjY2VlYzQ.
* SIP Call
Curr. trans. direction: Incoming
Call-ID: ZDFmODRiZGIyYzg2MWJlM2IzZDczNGZkNmVjY2VlYzQ.
Owner channel ID: SIP/test001-00000004
Our Codec Capability: 0xe (gsm|ulaw|alaw)
Non-Codec Capability (DTMF): 1
Their Codec Capability: 0x60c (ulaw|alaw|speex|ilbc)
Joint Codec Capability: 0xc (ulaw|alaw)
Format: 0x4 (ulaw)
T.38 support No
Video support No
MaxCallBR: 1024 kbps
Theoretical Address: 10.0.0.2:5060
Received Address: 10.0.0.2:5060
SIP Transfer mode: open
Force rport: Yes
Audio IP: 10.0.4.63 (local)
Our Tag: as167300ab
Their Tag: 802d1a03
SIP User agent: Zoiper rev.11137
Username: test001
Peername: test001
Original uri: sip:test001 at 10.0.0.2:5060
Caller-ID: 5678
Need Destroy: No
Last Message: Rx: ACK
Promiscuous Redir: No
Route: sip:test001 at 10.0.0.2:5060;transport=UDP
DTMF Mode: rfc2833
SIP Options: norefersub replaces replace
Session-Timer: Inactive
{code}
> 'directmedia' not working any more
> ----------------------------------
>
> Key: ASTERISK-20333
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-20333
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General
> Affects Versions: 1.8.15.0
> Reporter: Maciej Krajewski
> Attachments: directmedia.cap, full, full+sipdebug.log, sip.conf, sippeers.csv, sippeers_propper.txt
>
>
> I've updated me asterisk today from 1.8.13.1 to 1.8.15.0,, what I've noticed is that directmedia has stopped working.
> I'm using realtime peer configuration, I have directmedia set to yes on both friends calling each other (simple dial without any options).
> {code}
> exten => 1,1,Dial(SIP/test002)
> same => n,Hangup()
> {code}
> Configured peers has dtmfmode set to "info" (I know this is an issue that direct media ins not working if set to rfc2833 or inbound).
> When I've downgraded asterisk to 1.8.13.1 the directmedia has worked again. I've done some investigation and what I have found is that some change between 1.8.13.1 and 1.8.14.0 has done that.
> I've attached the logs from asterisk 1.8.15.0
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