Re: [asterisk-users] One side voice one side musiconhold
Currently I have no idea. But you wrote, that it does not happen all the
time. Please provide us a log extract from that case where is going
wrong. Perhaps you can do a "diff" for the good and the bad case
yourself before?!
Am 02.10.2012 09:02, schrieb Gianluca Baù:
Hello Thorsten,
i had a trace with core set debug 10 and core set verbose 10 but i
didn't find anything usefull.
The log is very full so it could be that i missed some important information.
This is a the less verbose output of the problem:
-- SIP/22-01b3 answered SIP/64-01b2
-- Started music on hold, class 'default', on SIP/22-01b3
-- Stopped music on hold on SIP/siprouter-01aa
-- Executing [h@to-operators:1] Goto("SIP/64-01b2",
"9991") in new stack
-- Goto (to-operators,h,9991)
-- Executing [h@to-operators:9991] Set("SIP/64-01b2",
"~~parentcxt~~=") in new stack
-- Executing [h@to-operators:9992]
GotoIf("SIP/64-01b2", "1?9996") in new stack
-- Goto (to-operators,h,9996)
-- Executing [h@to-operators:9996] NoOp("SIP/64-01b2",
"") in new stack
Where:
SIP/siprouter-01aa is A
SIP/64 is B
SIP/22 is C
I think this is the moment of the transfer.
-- Started music on hold, class 'default', on SIP/22-01b3
-- Stopped music on hold on SIP/siprouter-01aa
After the transfer of the call from B it seems to start the music to C
and to stop it on A.
I'll try to provide you a better trace. Do you have any ideas about the cause?
Thanks, regards
Gianluca
2012/10/1 Thorsten Göllner :
Did you take a look at the asterisk log? With "core set verbose 3" or more?
Am 01.10.2012 12:46, schrieb Gianluca Baù:
Hello guys,
my name is Gianluca and this is my first post in this ml.
i've a strange problem with my asterisk box. I'll try to explain you.
A (sip from ser) calls --> B (sip asterisk peer)
B put A on hold with musiconhold
B calls C
B transfer the call with A to C
A hears the C voice while C hears musiconhold
C is every peer of the asterisk.
This happens with version 1.6.22 and Asterisk 1.8.14.0 too. I tried to
update but the problem persists.
I've to say that the used phones are the same for both the versions.
They are Snom and Grandstream.
This problem is hard to debug because it doesn't happen everytime.
Did you hear something about this problem? Can you suggest me how to
understand this situation?
Thanks, regards
--
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Re: [asterisk-users] One side voice one side musiconhold
Hello Thorsten,
i had a trace with core set debug 10 and core set verbose 10 but i
didn't find anything usefull.
The log is very full so it could be that i missed some important information.
This is a the less verbose output of the problem:
-- SIP/22-01b3 answered SIP/64-01b2
-- Started music on hold, class 'default', on SIP/22-01b3
-- Stopped music on hold on SIP/siprouter-01aa
-- Executing [h@to-operators:1] Goto("SIP/64-01b2",
"9991") in new stack
-- Goto (to-operators,h,9991)
-- Executing [h@to-operators:9991] Set("SIP/64-01b2",
"~~parentcxt~~=") in new stack
-- Executing [h@to-operators:9992]
GotoIf("SIP/64-01b2", "1?9996") in new stack
-- Goto (to-operators,h,9996)
-- Executing [h@to-operators:9996] NoOp("SIP/64-01b2",
"") in new stack
Where:
SIP/siprouter-01aa is A
SIP/64 is B
SIP/22 is C
I think this is the moment of the transfer.
-- Started music on hold, class 'default', on SIP/22-01b3
-- Stopped music on hold on SIP/siprouter-01aa
After the transfer of the call from B it seems to start the music to C
and to stop it on A.
I'll try to provide you a better trace. Do you have any ideas about the cause?
Thanks, regards
Gianluca
2012/10/1 Thorsten Göllner :
> Did you take a look at the asterisk log? With "core set verbose 3" or more?
>
> Am 01.10.2012 12:46, schrieb Gianluca Baù:
>
>> Hello guys,
>>
>> my name is Gianluca and this is my first post in this ml.
>>
>> i've a strange problem with my asterisk box. I'll try to explain you.
>>
>> A (sip from ser) calls --> B (sip asterisk peer)
>>
>> B put A on hold with musiconhold
>>
>> B calls C
>>
>> B transfer the call with A to C
>>
>> A hears the C voice while C hears musiconhold
>>
>> C is every peer of the asterisk.
>>
>> This happens with version 1.6.22 and Asterisk 1.8.14.0 too. I tried to
>> update but the problem persists.
>> I've to say that the used phones are the same for both the versions.
>> They are Snom and Grandstream.
>>
>> This problem is hard to debug because it doesn't happen everytime.
>>
>> Did you hear something about this problem? Can you suggest me how to
>> understand this situation?
>>
>> Thanks, regards
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
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Re: [asterisk-users] One side voice one side musiconhold
Did you take a look at the asterisk log? With "core set verbose 3" or more? Am 01.10.2012 12:46, schrieb Gianluca Baù: Hello guys, my name is Gianluca and this is my first post in this ml. i've a strange problem with my asterisk box. I'll try to explain you. A (sip from ser) calls --> B (sip asterisk peer) B put A on hold with musiconhold B calls C B transfer the call with A to C A hears the C voice while C hears musiconhold C is every peer of the asterisk. This happens with version 1.6.22 and Asterisk 1.8.14.0 too. I tried to update but the problem persists. I've to say that the used phones are the same for both the versions. They are Snom and Grandstream. This problem is hard to debug because it doesn't happen everytime. Did you hear something about this problem? Can you suggest me how to understand this situation? Thanks, regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One side voice one side musiconhold
Hello guys, my name is Gianluca and this is my first post in this ml. i've a strange problem with my asterisk box. I'll try to explain you. A (sip from ser) calls --> B (sip asterisk peer) B put A on hold with musiconhold B calls C B transfer the call with A to C A hears the C voice while C hears musiconhold C is every peer of the asterisk. This happens with version 1.6.22 and Asterisk 1.8.14.0 too. I tried to update but the problem persists. I've to say that the used phones are the same for both the versions. They are Snom and Grandstream. This problem is hard to debug because it doesn't happen everytime. Did you hear something about this problem? Can you suggest me how to understand this situation? Thanks, regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
