tom
Premium Account
Registration Date: 25.04.2007
Posts: 80
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Ok. have the same problem on 3 differente extensions:
284 - in one EU city
701 - in another EU city
711 - In the US on NY server.
We have differerent sip phones on the extensions.
Till yesterday everything was fine 100%
From this morning the teleohione rings but we can't hear any voice, on all the extensions the same problem, I call back the number that I see and it works fine.
what is goning on????
This is the log of a call from 711 to 701 (NY-Rome) with no sound:
Aug 4 15:59:44 VERBOSE[6672] logger.c: Ignoring this INVITE request
Aug 4 15:59:44 VERBOSE[6672] chan_sip.c: Got a SIP re-invite for call ZTA2YjI2NGRmYWYxMDE5YTNiYzAzNGVkZGY4OWMzZDk.
Aug 4 15:59:45 VERBOSE[6672] logger.c: Found RTP audio format 0
Aug 4 15:59:45 VERBOSE[6672] logger.c: Found RTP audio format 98
Aug 4 15:59:45 VERBOSE[6672] logger.c: Found RTP audio format 8
Aug 4 15:59:45 VERBOSE[6672] logger.c: Found RTP audio format 101
Aug 4 15:59:45 VERBOSE[6672] logger.c: Peer audio RTP is at port 69.201.145.71:23700
Aug 4 15:59:45 VERBOSE[6672] logger.c: Peer video RTP is at port 69.201.145.71:65535
Aug 4 15:59:45 VERBOSE[6672] logger.c: Found description format iLBC
Aug 4 15:59:45 VERBOSE[6672] logger.c: Found description format telephone-event
Aug 4 15:59:45 VERBOSE[6672] logger.c: Capabilities: us - 0x18041e (gsm|ulaw|alaw|g726|ilbc|h263|h263p), peer - audio=0x40c (ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40c (ulaw|alaw|ilbc)
Aug 4 15:59:45 VERBOSE[6672] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug 4 15:59:45 VERBOSE[10091] logger.c: We're at 88.198.18.239 port 38824
Aug 4 15:59:45 VERBOSE[10091] logger.c: Video is at 88.198.18.239 port 46136
Aug 4 15:59:45 VERBOSE[10091] logger.c: Adding codec 0x4 (ulaw) to SDP
Aug 4 15:59:45 VERBOSE[10091] logger.c: Adding codec 0x8 (alaw) to SDP
Aug 4 15:59:45 VERBOSE[10091] logger.c: Adding codec 0x10 (g726) to SDP
Aug 4 15:59:45 VERBOSE[10091] logger.c: Adding codec 0x400 (ilbc) to SDP
Aug 4 15:59:45 VERBOSE[10091] logger.c: Adding codec 0x2 (gsm) to SDP
Aug 4 15:59:45 VERBOSE[10091] logger.c: Adding codec 0x80000 (h263) to SDP
Aug 4 15:59:45 VERBOSE[10091] logger.c: Adding codec 0x100000 (h263p) to SDP
Aug 4 15:59:45 VERBOSE[10091] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Aug 4 15:59:45 VERBOSE[10091] logger.c: -- Called tommasodemot-701
Aug 4 15:59:45 VERBOSE[6672] logger.c: Ignoring this INVITE request
Aug 4 15:59:45 VERBOSE[6672] chan_sip.c: Got a SIP re-invite for call ZTA2YjI2NGRmYWYxMDE5YTNiYzAzNGVkZGY4OWMzZDk.
Aug 4 15:59:46 VERBOSE[6672] chan_sip.c: SIP response 100 to standard invite
Aug 4 15:59:46 VERBOSE[6672] chan_sip.c: SIP response 100 to standard invite
Aug 4 15:59:46 VERBOSE[6672] chan_sip.c: SIP response 180 to standard invite
Aug 4 15:59:46 VERBOSE[10091] logger.c: -- SIP/tommasodemot-701-bf6c is ringing
Aug 4 15:59:53 VERBOSE[6672] chan_sip.c: SIP response 200 to standard invite
Aug 4 15:59:53 VERBOSE[6672] logger.c: Found RTP audio format 0
Aug 4 15:59:53 VERBOSE[6672] logger.c: Found RTP audio format 101
Aug 4 15:59:53 VERBOSE[6672] logger.c: Peer audio RTP is at port 93.150.40.224:60000
Aug 4 15:59:53 VERBOSE[6672] logger.c: Peer video RTP is at port 93.150.40.224:65535
Aug 4 15:59:53 VERBOSE[6672] logger.c: Found description format PCMU
Aug 4 15:59:53 VERBOSE[6672] logger.c: Found description format telephone-event
Aug 4 15:59:53 VERBOSE[6672] logger.c: Capabilities: us - 0x18041e (gsm|ulaw|alaw|g726|ilbc|h263|h263p), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Aug 4 15:59:53 VERBOSE[6672] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug 4 15:59:53 VERBOSE[10091] logger.c: -- SIP/tommasodemot-701-bf6c answered SIP/tommasodemot-711-4f73
Aug 4 15:59:53 VERBOSE[10091] logger.c: We're at 88.198.18.239 port 46870
Aug 4 15:59:53 VERBOSE[10091] logger.c: Video is at 88.198.18.239 port 44704
Aug 4 15:59:53 VERBOSE[10091] logger.c: Adding codec 0x4 (ulaw) to SDP
Aug 4 15:59:53 VERBOSE[10091] logger.c: Adding codec 0x8 (alaw) to SDP
Aug 4 15:59:53 VERBOSE[10091] logger.c: Adding codec 0x400 (ilbc) to SDP
Aug 4 15:59:53 VERBOSE[10091] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Aug 4 16:00:15 VERBOSE[10091] chan_sip.c: Hangup call SIP/tommasodemot-701-bf6c, SIP callid 3ecf9dca0289be2d39c246571bb7c421@88.198.18.239
Aug 4 16:00:16 VERBOSE[10091] chan_sip.c: Hangup call SIP/tommasodemot-711-4f73, SIP callid ZTA2YjI2NGRmYWYxMDE5YTNiYzAzNGVkZGY4OWMzZDk
This post has been edited 1 time(s), it was last edited by tom on 04.08.2008 at 15:03.
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04.08.2008 14:29 |
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tom
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04.08.2008 15:08 |
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i-p
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04.08.2008 16:46 |
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tom
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Registration Date: 25.04.2007
Posts: 80
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04.08.2008 16:55 |
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i-p
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04.08.2008 17:13 |
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tom
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04.08.2008 21:14 |
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i-p
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05.08.2008 07:28 |
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tom
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05.08.2008 13:33 |
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i-p
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05.08.2008 13:59 |
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tom
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05.08.2008 14:06 |
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i-p
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05.08.2008 16:48 |
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tom
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05.08.2008 16:55 |
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tom
Premium Account
Registration Date: 25.04.2007
Posts: 80
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Now I can't register with any extension. I try pbxrs.org and all the IP of www0-Frankfurt, www1-Nuernberg-Alt , www2-San-Diego, www3-Nuernberg-Cnt , www4-New-York
NO way: registration error 408 Request timeout
To answer the other questions:
Normally you did: X calls Y - no audio. Y calls back X - works.
CORRECT
Now please try: Y calls *43 echo test. X calls Y. Does this always work?
No, however echo is always workink from all extensions. The propblems are JUST on INcomings calls
the big problem is that now that you moved me I cant get registred with any extension
you write:
The logs of aug 5 17:55-17:56 indicate these because extension 284 does not report ringing nor answer.
284 rings but on 711 (caller) there is just silence.
284 answer and 711 stille hear silence
another crazy thing:
I log out from the web interface and re login:
my borwser is blobked on a white page, this happends in an ethernal loop:
https://www1.pbxes.com/index.php
redirect
https://www1.pbxes.com/index.php
redirect
https://www1.pbxes.com/index.php
so I can't login in my account, I get blanck redirection pages from www1 to www3 and back to 1 to 3 to 1
UPDATE 20.30:
Empty the broweser cache, try with Firefox PC and Mac, Camino, Safari PC and mac and from a computer I Never used before to access PBXes. Always the same problem!
as I try to login:
www3 to www1 and back www3 in an infinite loop.
My extensions are not registred!
UPDATE 20.45:
went on http://www1.pbxes.com and http://www1.pbxes.com
try to login
in this case I was redirected to
http://www1.pbxes.com
but if i enter from
http://www1.pbxes.com
http://www1.pbxes.com
http://www1.pbxes.com
is still get the infinite loop
got 1 extension registred, just one
UPDATE 4: 20.51
The flash panel is not working, shows all extensions registred but they are not. I am not on server www3 but still on www1
UPDATE 5: 20.59
Now from all the www 1,2,3,4,5 I get redirected to www1 and the login is corret.
So is sure, I'm on the www1 server.
last update:
I STILL have the problem
This post has been edited 7 time(s), it was last edited by tom on 05.08.2008 at 20:26.
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05.08.2008 19:30 |
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i-p
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05.08.2008 21:39 |
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tom
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05.08.2008 21:56 |
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i-p
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05.08.2008 22:17 |
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