Hi,
David J. wrote:
Ok, I am going to capture this in NGREP.
To isolate the problem, I would like to create a simpler test that I
dial the SEMS app server and test the DTMF from a few different clients,
I assume that the issue is with blink on calls that are originated by
SEMS; it may be that blink does not include DTMF payload in SDP
answer. check the SDP in the 200 OK sent by blink.
you can find some simpler examples in doc/dsm/examples
So I guess I just need to Answer and Wait for Digits and See if the
problem is with other clients as well or just blink.
How could I simplify the script to create a test scnario?
I dont think I understand the script lifecycle yet.
We have an Initial invite, then we have transitions based on events.
exactly.
if you set run_invite_events=no in dsm.conf or your application
config, you will not get an invite event, instead you will only get
one sessionStart evetn, which will be executed when the call is
already set up and you can start sending/receiving audio.
BR
Stefan
Thanks.
On 8/24/10 12:34 PM, Stefan Sayer wrote:
you need to look further up, where the SDP negotiation takes place.
the problem is that RTP dtmf is sent as ID 96, but it is apparently
not included in the SDP as supported payload as it should be. is this
blink? possibly we need to tell this to the blink devs. therefore, can
you send an ngrep of the full call flow to the blink?
stefan
David J. wrote:
[#4046d950/27306] [~AmSipDialog, AmSipDialog.cpp:52] DEBUG: callid
= [email protected]
[#4046d950/27306] [~AmSipDialog, AmSipDialog.cpp:53] DEBUG:
local_tag = 68580430-4C73F2AC000C3110-4132F950
[#4046d950/27306] [~AmSipDialog, AmSipDialog.cpp:54] DEBUG:
uac_trans.size() = 0
[#4046d950/27306] [~AmSipDialog, AmSipDialog.cpp:62] DEBUG:
uas_trans.size() = 0
[#4046d950/27306] [run, AmSessionContainer.cpp:134] DEBUG: Session
cleaner finished
[#4196d950/27307] [conceal_loss, AmRtpAudio.cpp:182] DEBUG: default
PLC (ts_diff = 160; s = 320)
[#4196d950/27307] [read, AmAudioFile.cpp:409] DEBUG: 0-stuffing
packet: adding 260 bytes (packet size=320)
[#4196d950/27307] [setCurrentPayload, AmAudio.cpp:71] ERROR: Could
not find payload <96>
[#4196d950/27307] [get, AmPlaylist.cpp:78] DEBUG: get: gotoNextItem
[#4196d950/27307] [gotoNextItem, AmPlaylist.cpp:62] DEBUG: posting
AmAudioEvent::noAudio event!
[#4196d950/27307] [postEvent, AmEventQueue.cpp:53] DEBUG:
AmEventQueue: trying to post event
[#4196d950/27307] [postEvent, AmEventQueue.cpp:64] DEBUG:
AmEventQueue: event posted
[#4196d950/27307] [setCurrentPayload, AmAudio.cpp:71] ERROR: Could
not find payload <96>
[#40ae9950/27319] [processEvents, AmEventQueue.cpp:79] DEBUG: before
processing event (12AmAudioEvent)
[#40ae9950/27319] [runEvent, DSMStateEngine.cpp:342] DEBUG: checking
transition 'got cancel from A leg'
[#40ae9950/27319] [runEvent, DSMStateEngine.cpp:342] DEBUG: checking
transition 'pressed a number'
[#40ae9950/27319] [runEvent, DSMStateEngine.cpp:342] DEBUG: checking
transition 'pressed hash or start'
[#40ae9950/27319] [processEvents, AmEventQueue.cpp:83] DEBUG: event
processed (12AmAudioEvent)
[#40ae9950/27319] [processingCycle, AmSession.cpp:446] DEBUG:
[email protected]/7F742D34-4C73F2AC000C7F30-4132F950:
Connected, running, 0 UACTransPending
[#4196d950/27307] [setCurrentPayload, AmAudio.cpp:71] ERROR: Could
not find payload <96>
[#4196d950/27307] [setCurrentPayload, AmAudio.cpp:71] ERROR: Could
not find payload <96>
[#4196d950/27307] [setCurrentPayload, AmAudio.cpp:71] ERROR: Could
not find payload <96>
[#4196d950/27307] [setCurrentPayload, AmAudio.cpp:71] ERROR: Could
not find payload <96>
[#4196d950/27307] [setCurrentPayload, AmAudio.cpp:71] ERROR: Could
not find payload <96>
[#4196d950/27307] [read, AmAudioFile.cpp:409] DEBUG: 0-stuffing
packet: adding 260 bytes (packet size=320)
On 8/24/10 12:21 PM, Stefan Sayer wrote:
-D 3
--
Stefan Sayer
VoIP Services Consulting and Development
Warschauer Str. 24
10243 Berlin
tel:+491621366449
sip:[email protected]
email/xmpp:[email protected]
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