Re: [asterisk-users] help with crash

[Mark Murawski]

Hello Federico,

Can you please review the Bug Report requirements, and submit a new bug 
report for this issue?

https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/

Also Note:
Before filing a bug report... Your issue may not be a bug or could have 
been fixed already. Run through the check list below to verify you have 
done your due diligence.


Also Note:
You need to provide details regarding the crash.  Upload your dialplan 
(hide any secret information/passwords/etc).  Provide the console and 
debug logs that were just prior to the crash.



On 11/9/23 17:24, Federico wrote:


2023-11-08 18:14:13] ERROR[571246][C-17e2] : Got 19 backtrace records

# 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed()

# 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref()

# 2: [0x58e660] asterisk stasis_cache.c:824 update_create()

# 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec()

# 4: [0x586b90] asterisk stasis.c:1380 dispatch_message()

# 5: [inlined] asterisk stasis.c:1490 publish_msg()

# 6: [0x59588e] asterisk stasis_channels.c:796 
ast_channel_publish_snapshot()


# 7: [0x53b54c] asterisk pbx_app.c:488 pbx_exec()

# 8: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

# 9: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension()

#10: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec()

#11: [0x53b599] asterisk pbx_app.c:493 pbx_exec()

#12: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

#13: [0x5302b4] asterisk pbx.c:4377 __ast_pbx_run()

#14: [0x53184b] asterisk pbx.c:4669 decrease_call_count()

#15: [inlined] asterisk pbx.c:4702 pbx_thread()

#16: [0x5b8329] asterisk utils.c:1576 dummy_start()

#17: [0x7f62bec07ea5] libpthread.so.0 :0 __pthread_get_minstack()

#18: [0x7f62bd0fe8dd] libc.so.6 :0 clone()

[2023-11-08 18:14:13] ERROR[571292][C-17e4] stasis_cache.c: 
Excessive refcount 10 reached on ao2 object 0x3616b38


[2023-11-08 18:14:13] ERROR[571292][C-17e4] stasis_cache.c: 
FRACK!, Failed assertion Excessive refcount 10 reached on ao2 
object 0x3616b38 (0)


[2023-11-08 18:14:13] ERROR[571291][C-17e3] stasis_cache.c: 
Excessive refcount 10 reached on ao2 object 0x3616b38


[2023-11-08 18:14:13] ERROR[571291][C-17e3] stasis_cache.c: 
FRACK!, Failed assertion Excessive refcount 10 reached on ao2 
object 0x3616b38 (0)


[2023-11-08 18:14:13] ERROR[571290][C-17e2] stasis_cache.c: 
Excessive refcount 10 reached on ao2 object 0x3616b38


[2023-11-08 18:14:13] ERROR[571290][C-17e2] stasis_cache.c: 
FRACK!, Failed assertion Excessive refcount 10 reached on ao2 
object 0x3616b38 (0)


[2023-11-08 18:14:14] ERROR[571292][C-17e4] : Got 23 backtrace records

# 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed()

# 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref()

# 2: [0x58e660] asterisk stasis_cache.c:824 update_create()

# 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec()

# 4: [0x586b90] asterisk stasis.c:1380 dispatch_message()

# 5: [inlined] asterisk stasis.c:1490 publish_msg()

# 6: [0x59588e] asterisk stasis_channels.c:796 
ast_channel_publish_snapshot()


# 7: [0x53b54c] asterisk pbx_app.c:488 pbx_exec()

# 8: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

# 9: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension()

#10: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec()

#11: [0x53b599] asterisk pbx_app.c:493 pbx_exec()

#12: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

#13: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension()

#14: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec()

#15: [0x53b599] asterisk pbx_app.c:493 pbx_exec()

#16: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

#17: [0x5302b4] asterisk pbx.c:4377 __ast_pbx_run()

#18: [0x53184b] asterisk pbx.c:4669 decrease_call_count()

#19: [inlined] asterisk pbx.c:4702 pbx_thread()

#20: [0x5b8329] asterisk utils.c:1576 dummy_start()

#21: [0x7f62bec07ea5] libpthread.so.0 :0 __pthread_get_minstack()

#22: [0x7f62bd0fe8dd] libc.so.6 :0 clone()

[2023-11-08 18:14:14] ERROR[571291][C-17e3] : Got 19 backtrace records

# 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed()

# 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref()

# 2: [0x58e660] asterisk stasis_cache.c:824 update_create()

# 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec()

# 4: [0x586b90] asterisk stasis.c:1380 dispatch_message()

# 5: [inlined] asterisk stasis.c:1490 publish_msg()

# 6: [0x59588e] asterisk stasis_channels.c:796 
ast_channel_publish_snapshot()


# 7: [0x53b54c] asterisk pbx_app.c:488 pbx_exec()

# 8: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

# 9: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension()

#10: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec()

#11: [0x53b599] asterisk pbx_app.c:493 pbx_exec()

#12: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

#13: [0x5302b4] asterisk pbx.c:4377 __ast_pbx_run()

#14: [0x53184b] asterisk pbx.c:4669 

[asterisk-users] help with crash

[Federico]

2023-11-08 18:14:13] ERROR[571246][C-17e2] : Got 19 backtrace records

# 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed()

# 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref()

# 2: [0x58e660] asterisk stasis_cache.c:824 update_create()

# 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec()

# 4: [0x586b90] asterisk stasis.c:1380 dispatch_message()

# 5: [inlined] asterisk stasis.c:1490 publish_msg()

# 6: [0x59588e] asterisk stasis_channels.c:796
ast_channel_publish_snapshot()

# 7: [0x53b54c] asterisk pbx_app.c:488 pbx_exec()

# 8: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

# 9: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension()

#10: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec()

#11: [0x53b599] asterisk pbx_app.c:493 pbx_exec()

#12: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

#13: [0x5302b4] asterisk pbx.c:4377 __ast_pbx_run()

#14: [0x53184b] asterisk pbx.c:4669 decrease_call_count()

#15: [inlined] asterisk pbx.c:4702 pbx_thread()

#16: [0x5b8329] asterisk utils.c:1576 dummy_start()

#17: [0x7f62bec07ea5] libpthread.so.0 :0 __pthread_get_minstack()

#18: [0x7f62bd0fe8dd] libc.so.6 :0 clone()

 

[2023-11-08 18:14:13] ERROR[571292][C-17e4] stasis_cache.c: Excessive
refcount 10 reached on ao2 object 0x3616b38

[2023-11-08 18:14:13] ERROR[571292][C-17e4] stasis_cache.c: FRACK!,
Failed assertion Excessive refcount 10 reached on ao2 object 0x3616b38
(0)

[2023-11-08 18:14:13] ERROR[571291][C-17e3] stasis_cache.c: Excessive
refcount 10 reached on ao2 object 0x3616b38

[2023-11-08 18:14:13] ERROR[571291][C-17e3] stasis_cache.c: FRACK!,
Failed assertion Excessive refcount 10 reached on ao2 object 0x3616b38
(0)

[2023-11-08 18:14:13] ERROR[571290][C-17e2] stasis_cache.c: Excessive
refcount 10 reached on ao2 object 0x3616b38

[2023-11-08 18:14:13] ERROR[571290][C-17e2] stasis_cache.c: FRACK!,
Failed assertion Excessive refcount 10 reached on ao2 object 0x3616b38
(0)

[2023-11-08 18:14:14] ERROR[571292][C-17e4] : Got 23 backtrace records

# 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed()

# 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref()

# 2: [0x58e660] asterisk stasis_cache.c:824 update_create()

# 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec()

# 4: [0x586b90] asterisk stasis.c:1380 dispatch_message()

# 5: [inlined] asterisk stasis.c:1490 publish_msg()

# 6: [0x59588e] asterisk stasis_channels.c:796
ast_channel_publish_snapshot()

# 7: [0x53b54c] asterisk pbx_app.c:488 pbx_exec()

# 8: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

# 9: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension()

#10: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec()

#11: [0x53b599] asterisk pbx_app.c:493 pbx_exec()

#12: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

#13: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension()

#14: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec()

#15: [0x53b599] asterisk pbx_app.c:493 pbx_exec()

#16: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

#17: [0x5302b4] asterisk pbx.c:4377 __ast_pbx_run()

#18: [0x53184b] asterisk pbx.c:4669 decrease_call_count()

#19: [inlined] asterisk pbx.c:4702 pbx_thread()

#20: [0x5b8329] asterisk utils.c:1576 dummy_start()

#21: [0x7f62bec07ea5] libpthread.so.0 :0 __pthread_get_minstack()

#22: [0x7f62bd0fe8dd] libc.so.6 :0 clone()

 

[2023-11-08 18:14:14] ERROR[571291][C-17e3] : Got 19 backtrace records

# 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed()

# 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref()

# 2: [0x58e660] asterisk stasis_cache.c:824 update_create()

# 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec()

# 4: [0x586b90] asterisk stasis.c:1380 dispatch_message()

# 5: [inlined] asterisk stasis.c:1490 publish_msg()

# 6: [0x59588e] asterisk stasis_channels.c:796
ast_channel_publish_snapshot()

# 7: [0x53b54c] asterisk pbx_app.c:488 pbx_exec()

# 8: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

# 9: [0x52fd83] asterisk pbx.c:4204 ast_spawn_extension()

#10: [0x7f625d802ccf] app_macro.so app_macro.c:463 _macro_exec()

#11: [0x53b599] asterisk pbx_app.c:493 pbx_exec()

#12: [0x52e039] asterisk pbx.c:2989 pbx_extension_helper()

#13: [0x5302b4] asterisk pbx.c:4377 __ast_pbx_run()

#14: [0x53184b] asterisk pbx.c:4669 decrease_call_count()

#15: [inlined] asterisk pbx.c:4702 pbx_thread()

#16: [0x5b8329] asterisk utils.c:1576 dummy_start()

#17: [0x7f62bec07ea5] libpthread.so.0 :0 __pthread_get_minstack()

#18: [0x7f62bd0fe8dd] libc.so.6 :0 clone()

 

[2023-11-08 18:14:14] ERROR[571290][C-17e2] : Got 23 backtrace records

# 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed()

# 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref()

# 2: [0x58e660] asterisk stasis_cache.c:824 update_create()

# 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec()

# 4: [0x586b90] asterisk stasis.c:1380 dispatch_message()

# 5: [inlined] 

Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

[Steve Edwards]

On Wed, 26 May 2021, Jonathan H wrote:


AGI Rx << SET AUTOHANGUP 5
AGI Tx >> 200 result=0
AGI Tx >> HANGUP   <<


This does raise a question in my mind...

The AGI protocol is: your AGI sends a request (the Rx line) and receives 
a response (the Tx line). 1 line out, 1 line in.


If the 'HANGUP' text can arrive asynchronously, how are you supposed to 
know it has arrived? Poll (or select) on the file pointer?


I cannot use other methods like setting the absolute channel timeout 
variable


I don't understand why you can't use the absolute channel timeout. 
Wherever you 'set autohangup x' just set 'TIMEOUT(absolute)=${EPOCH}+x.'


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

--
_
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Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

[Steve Edwards]

On Wed, 26 May 2021, Jonathan H wrote:

It just causes AGI to send "HANGUP" and any audio to stop playing. It 
does NOT hangup the channel, or even send any SIP event. The line just 
goes silent.


I wouldn't expect the AGI() application to send a SIP event. The AGI()
application does not care what technology you use.

Receiving 'HANGUP' as text from Asterisk appears to be a FastAGI thing
which kind of makes sense -- if your FastAGI server is not localhost,
how could Asterisk send it a signal?

Are you supposed to close your TCP connection and exit your AGI when you 
receive the HANGUP text?


When I set autohangup in a 'normal' AGI, it looks like this:

AGI Tx >> agi_request: null-agi.php
AGI Tx >> agi_channel: SIP/poly-77a1-02a2
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1622093977.1168
AGI Tx >> agi_version: 13.14.1~dfsg-2+deb9u4
AGI Tx >> agi_callerid: 55
AGI Tx >> agi_calleridname: Steve Edwards
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: *
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: newline
AGI Tx >> agi_extension: *
AGI Tx >> agi_priority: 6
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode: 
AGI Tx >> agi_threadid: 1945654064
AGI Tx >> 
AGI Rx << set autohangup 5

AGI Tx >> 200 result=0
   > 0x73c3dba0 -- Strict RTP learning complete - Locking on source address 
192.168.0.139:2254
(and then after 5 seconds)
-- AGI Script null-agi.php completed, returning 4

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

--
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

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Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

[Jonathan H]

I think I can confidently say, after most of a day and reading the following

https://stackoverflow.com/questions/66768885/why-doesnt-asterisk-17-catch-hangup-request-from-pjsip-client-solved
https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup
https://community.freepbx.org/t/inbound-calls-dont-hang-up/53612
https://community.freepbx.org/t/pjsip-problem-channel-not-closing/65311/7
https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_ENDPOINT
https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_pjsip

... that Asterisk doesn't like mixing autoHangup with AGI, and there
appears to be no way of the ts-agi library I'm using knowing that it
has autoHungup, so it can't close the AGI connection which seems to
release Asterisk to hangup properly.

I had thought that the AGIEXITONHANGUP variable might help, but it
appears to do nothing, although I'm unsure if I'm setting it correctly
as:

Here it says the flag is "1"
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables

And here it says the flag is "yes"
https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Application_AGI

That said, I wish I could use ARI not AGI but without the current
media offset available on ARI, I need AGI :)

So for now, the workaround is to forget about setting autohangup, and
just hangup the caller manually at the point they don't over-ride the
timeout.

Thanks!

On Wed, 26 May 2021 at 18:01, Joshua C. Colp  wrote:
>
> On Wed, May 26, 2021 at 1:58 PM Jonathan H  wrote:
>>
>> I have also tried configuring pjsip wizard like this.
>>
>> endpoint/rtp_timeout=5
>>
>> And I see this shortly after the "hangup" command has been sent, so
>> that part is working:
>>
>> [May 26 17:36:37] NOTICE[1276]: res_pjsip_sdp_rtp.c:150
>> rtp_check_timeout: Disconnecting channel
>> 'PJSIP/fromvoipfone-206-000b' for lack of audio RTP activity in 5
>> seconds
>>
>> But, again, it doesn't disconnect. The line stays open. And yes, my
>> fallthrough after agi is
>>
>> same => n, Hangup()
>>
>> Also, apparently I now have a load of channels, which won't even hangup with
>>
>> channel request hangup all
>>
>> Requested Hangup on channel 'PJSIP/fromvoipfone-206-000b'
>> Requested Hangup on channel 'PJSIP/fromvoipfone-206-000a'
>> Requested Hangup on channel 'PJSIP/fromvoipfone-206-0009'
>>
>> ...and wait.. and then...
>>
>> Channel  Location State   Application(Data)
>> PJSIP/fromvoipfone-2 s@test:2 Up  AGI(agi://localhost:3456)
>> PJSIP/fromvoipfone-2 s@test:2 Up  AGI(agi://localhost:3456)
>> PJSIP/fromvoipfone-2 s@test:2 Up  AGI(agi://localhost:3456)
>> 3 active channels
>> 3 active calls
>>
>> So they just won't die.
>>
>> Asterisk 18.4.0 - worth filing a bug?
>
>
> Is your AGI closing the connection or are you expecting Asterisk to drop it? 
> (I'm not that familiar with FastAGI or AGI these days, just wondering what 
> happens if you drop the connection)
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
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Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

[Joshua C. Colp]

On Wed, May 26, 2021 at 1:58 PM Jonathan H  wrote:

> I have also tried configuring pjsip wizard like this.
>
> endpoint/rtp_timeout=5
>
> And I see this shortly after the "hangup" command has been sent, so
> that part is working:
>
> [May 26 17:36:37] NOTICE[1276]: res_pjsip_sdp_rtp.c:150
> rtp_check_timeout: Disconnecting channel
> 'PJSIP/fromvoipfone-206-000b' for lack of audio RTP activity in 5
> seconds
>
> But, again, it doesn't disconnect. The line stays open. And yes, my
> fallthrough after agi is
>
> same => n, Hangup()
>
> Also, apparently I now have a load of channels, which won't even hangup
> with
>
> channel request hangup all
>
> Requested Hangup on channel 'PJSIP/fromvoipfone-206-000b'
> Requested Hangup on channel 'PJSIP/fromvoipfone-206-000a'
> Requested Hangup on channel 'PJSIP/fromvoipfone-206-0009'
>
> ...and wait.. and then...
>
> Channel  Location State   Application(Data)
> PJSIP/fromvoipfone-2 s@test:2 Up
> AGI(agi://localhost:3456)
> PJSIP/fromvoipfone-2 s@test:2 Up
> AGI(agi://localhost:3456)
> PJSIP/fromvoipfone-2 s@test:2 Up
> AGI(agi://localhost:3456)
> 3 active channels
> 3 active calls
>
> So they just won't die.
>
> Asterisk 18.4.0 - worth filing a bug?
>

Is your AGI closing the connection or are you expecting Asterisk to drop
it? (I'm not that familiar with FastAGI or AGI these days, just wondering
what happens if you drop the connection)

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-- 
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Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

[Jonathan H]

I have also tried configuring pjsip wizard like this.

endpoint/rtp_timeout=5

And I see this shortly after the "hangup" command has been sent, so
that part is working:

[May 26 17:36:37] NOTICE[1276]: res_pjsip_sdp_rtp.c:150
rtp_check_timeout: Disconnecting channel
'PJSIP/fromvoipfone-206-000b' for lack of audio RTP activity in 5
seconds

But, again, it doesn't disconnect. The line stays open. And yes, my
fallthrough after agi is

same => n, Hangup()

Also, apparently I now have a load of channels, which won't even hangup with

channel request hangup all

Requested Hangup on channel 'PJSIP/fromvoipfone-206-000b'
Requested Hangup on channel 'PJSIP/fromvoipfone-206-000a'
Requested Hangup on channel 'PJSIP/fromvoipfone-206-0009'

...and wait.. and then...

Channel  Location State   Application(Data)
PJSIP/fromvoipfone-2 s@test:2 Up  AGI(agi://localhost:3456)
PJSIP/fromvoipfone-2 s@test:2 Up  AGI(agi://localhost:3456)
PJSIP/fromvoipfone-2 s@test:2 Up  AGI(agi://localhost:3456)
3 active channels
3 active calls

So they just won't die.

Asterisk 18.4.0 - worth filing a bug?

On Wed, 26 May 2021 at 17:22, Jonathan H  wrote:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup
>
> "Cause the channel to automatically hangup at time seconds in the future"
>
> SET AUTOHANGUP TIME
>
> Looks great. Except... it doesn't. It just causes AGI to send "HANGUP"
> and any audio to stop playing.
> It does NOT hangup the channel, or even send any SIP event. The line
> just goes silent.
>
> It's been an entire afternoon of profuse googling; I have tried adding
> and removing hangup handlers, I have even tried setting the
> AGIEXITONHANGUP flag to "1" as per
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
>
> But this is all that happens (5 seconds is ridiculous, it's just to
> test). This is with "pjsip set logger on"
>
> AGI Rx << SET AUTOHANGUP 5
> AGI Tx >> 200 result=0
> AGI Rx << SET VARIABLE AGIEXITONHANGUP "1"
> AGI Tx >> 200 result=1
> AGI Tx >> HANGUP   <<
> AGI Rx << HANGUP
> AGI Tx >> 511 Command Not Permitted on a dead channel or intercept routine
>
> basically, the next log event is whenever the next REGISTER would
> normally happen.
>
> And of course, if I try and respond to the AGI again, it tells me the
> channel is dead. So WHY is the "hangup" event not sent to the phone?
>
> I cannot use other methods like setting the absolute channel timeout
> variable, or using a local "dial" with a timeout message because:
>
> 1: the system tests whether the next file will push the listener over
> the 1 hour limit for included calls in UK packages, so the autohangup
> value has to be dynamic
> 2: It has to be over-ridable so that the listener can continue past the hour
> 3: The message that plays out is dynamically generated
>
> Everything about it works fine... except the fact that it doesn't
> actually hangup.
> Which leaves the possibility that an old person might fall asleep and
> end up with a large bill, which we don't want!

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[asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

[Jonathan H]

https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup

"Cause the channel to automatically hangup at time seconds in the future"

SET AUTOHANGUP TIME

Looks great. Except... it doesn't. It just causes AGI to send "HANGUP"
and any audio to stop playing.
It does NOT hangup the channel, or even send any SIP event. The line
just goes silent.

It's been an entire afternoon of profuse googling; I have tried adding
and removing hangup handlers, I have even tried setting the
AGIEXITONHANGUP flag to "1" as per
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables

But this is all that happens (5 seconds is ridiculous, it's just to
test). This is with "pjsip set logger on"

AGI Rx << SET AUTOHANGUP 5
AGI Tx >> 200 result=0
AGI Rx << SET VARIABLE AGIEXITONHANGUP "1"
AGI Tx >> 200 result=1
AGI Tx >> HANGUP   <<
AGI Rx << HANGUP
AGI Tx >> 511 Command Not Permitted on a dead channel or intercept routine

basically, the next log event is whenever the next REGISTER would
normally happen.

And of course, if I try and respond to the AGI again, it tells me the
channel is dead. So WHY is the "hangup" event not sent to the phone?

I cannot use other methods like setting the absolute channel timeout
variable, or using a local "dial" with a timeout message because:

1: the system tests whether the next file will push the listener over
the 1 hour limit for included calls in UK packages, so the autohangup
value has to be dynamic
2: It has to be over-ridable so that the listener can continue past the hour
3: The message that plays out is dynamically generated

Everything about it works fine... except the fact that it doesn't
actually hangup.
Which leaves the possibility that an old person might fall asleep and
end up with a large bill, which we don't want!

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Re: [asterisk-users] Help needed with ARI RTP externalMedia bridging please

[Jonathan H]

You're a genius, sir! I don't know how I missed the part about ports, but
anyway...

Looks for "channelvars": {
"UNICASTRTP_LOCAL_PORT": "*14880*",

and then

vlc -vvv https://media-ssl.musicradio.com/LBCUK --sout
'#transcode{vcodec=none,acodec=*a*
law,channels=1,samplerate=8000}:rtp{dst=127.0.0.1,port=14880}'

Then the rest as before...

The other change I made was from ulaw to alaw, as ulaw sounded horribly
scratchy, particularly with "s" sounds. alaw was much better.

At last no more dynamic rewriting of moh files and then reloading moh etc
etc!

Thanks again.

On Mon, 4 Jan 2021 at 17:03, Joshua C. Colp  wrote:

> On Sun, Jan 3, 2021 at 4:14 PM Jonathan H  wrote:
>
>> Very simply, I want to pipe some external audio into a channel (bridge)
>> using the externalMedia channel option.
>> Running Asterisk 18 on ubuntu, here's what I did to try and test things
>> out:
>>
>> open a console tab
>> vlc -vvv https://media-ssl.musicradio.com/LBCUK --sout
>> '#transcode{vcodec=none,acodec=ulaw,channels=1,samplerate=8000}:rtp{dst=127.0.0.1,port=5005}'
>>
>> open another console tab
>> wscat -c
>> "ws://localhost:8088/ari/events?api_key=asterisk:asterisk=playback-example"
>>
>> open another console tab
>> curl -v -u asterisk:asterisk -X POST "
>> http://localhost:8088/ari/channels/externalMedia?external_host=127.0.0.1:5005=musicChannel=ulaw=playback-example
>> "
>> curl -v -u asterisk:asterisk -X POST "
>> http://localhost:8088/ari/bridges/musicBridge?type=mixing;
>> then dial in from a phone
>> curl -v -u asterisk:asterisk -X GET "http://localhost:8088/ari/channels
>> "
>> and note the new call channel ID
>> curl -v -u asterisk:asterisk -X POST "
>> http://localhost:8088/ari/bridges/musicBridge/addChannel?channel=> channel from above>,musicChannel"
>>
>
> You're sending media from Asterisk to VLC, not the other way around,
> currently. You need to examine the result from the call to
> channels/externalMedia, this will include the RTP port that Asterisk is
> listening for media on. You then need to pass this to vlc somehow (I'm not
> familar with the VLC options) and have it send RTP to that port.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] Help needed with ARI RTP externalMedia bridging please

[Joshua C. Colp]

On Sun, Jan 3, 2021 at 4:14 PM Jonathan H  wrote:

> Very simply, I want to pipe some external audio into a channel (bridge)
> using the externalMedia channel option.
> Running Asterisk 18 on ubuntu, here's what I did to try and test things
> out:
>
> open a console tab
> vlc -vvv https://media-ssl.musicradio.com/LBCUK --sout
> '#transcode{vcodec=none,acodec=ulaw,channels=1,samplerate=8000}:rtp{dst=127.0.0.1,port=5005}'
>
> open another console tab
> wscat -c
> "ws://localhost:8088/ari/events?api_key=asterisk:asterisk=playback-example"
>
> open another console tab
> curl -v -u asterisk:asterisk -X POST "
> http://localhost:8088/ari/channels/externalMedia?external_host=127.0.0.1:5005=musicChannel=ulaw=playback-example
> "
> curl -v -u asterisk:asterisk -X POST "
> http://localhost:8088/ari/bridges/musicBridge?type=mixing;
> then dial in from a phone
> curl -v -u asterisk:asterisk -X GET "http://localhost:8088/ari/channels;
> and note the new call channel ID
> curl -v -u asterisk:asterisk -X POST "
> http://localhost:8088/ari/bridges/musicBridge/addChannel?channel= channel from above>,musicChannel"
>

You're sending media from Asterisk to VLC, not the other way around,
currently. You need to examine the result from the call to
channels/externalMedia, this will include the RTP port that Asterisk is
listening for media on. You then need to pass this to vlc somehow (I'm not
familar with the VLC options) and have it send RTP to that port.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] Help needed with ARI RTP externalMedia bridging please

[Jonathan H]

Very simply, I want to pipe some external audio into a channel (bridge)
using the externalMedia channel option.
Running Asterisk 18 on ubuntu, here's what I did to try and test things out:

open a console tab
vlc -vvv https://media-ssl.musicradio.com/LBCUK --sout
'#transcode{vcodec=none,acodec=ulaw,channels=1,samplerate=8000}:rtp{dst=127.0.0.1,port=5005}'

open another console tab
wscat -c
"ws://localhost:8088/ari/events?api_key=asterisk:asterisk=playback-example"

open another console tab
curl -v -u asterisk:asterisk -X POST "
http://localhost:8088/ari/channels/externalMedia?external_host=127.0.0.1:5005=musicChannel=ulaw=playback-example
"
curl -v -u asterisk:asterisk -X POST "
http://localhost:8088/ari/bridges/musicBridge?type=mixing;
then dial in from a phone
curl -v -u asterisk:asterisk -X GET "http://localhost:8088/ari/channels;
and note the new call channel ID
curl -v -u asterisk:asterisk -X POST "
http://localhost:8088/ari/bridges/musicBridge/addChannel?channel=,musicChannel"

... and there is silence when I expected either music or at least some kind
of noise.

Is it something I'm doing wrong with the channel mixing, or with the
transcoding with vlc? Been battling this all day and could really use a
second pair of eyes on this.
Thanks.

(PS - of course, I would use the node SDK for live stuff, and not
"asterisk:asterisk" as username/password!)

Here's the VLC output - does "PCMU/8000 on port 5005 RTP/AVP 0" look
reasonable?

o=- 16401160996019665100 16401160996019665100 IN IP4 ip-172-31-37-244
s=Unnamed
i=N/A
c=IN IP4 127.0.0.1
t=0 0
a=tool:vlc 3.0.11
a=recvonly
a=type:broadcast
a=charset:UTF-8
m=audio 5005 RTP/AVP 0
b=AS:64
b=RR:0
a=rtpmap:0 PCMU/8000
a=rtcp:5006
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Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Duncan Turnbull]

> On 25/12/2020, at 3:08 PM, Turritopsis Dohrnii Teo En Ming 
>  wrote:
> 
> Hi Duncan Turnbull,
> 
> My final issue has been resolved.

Very well done

Merry Xmas

Cheers Duncan

> 
> Please refer to the following post.
> 
> Post: Addendum to Teo En Ming's Guide to Configuring Asterisk/FreePBX with 
> Cisco 7960 IP Phones
> Link: 
> http://lists.digium.com/pipermail/asterisk-users/2020-December/295590.html
> 
> Thank you very much.
> 
> Merry Christmas 2020!
> 
> 
> On 2020-12-25 03:12, Duncan Turnbull wrote:
 On 25/12/2020, at 12:40 AM, Turritopsis Dohrnii Teo En Ming 
  wrote:
>>> Hi Duncan Turnbull,
>>> It is a newly created PJSIP extension with default settings. I have never 
>>> configured Do Not Disturb settings before.
>>> Could it be something else?
>>> Do I need to run tcpdump on the Asterisk PBX server again?
>> Running tcpdump shows you the packets. You need to see the packets
>> during call setup and failure to work out what’s happening. So yes you
>> should run tcpdump, and I would recommend sngrep as well as an easier
>> way to understand the sip conversations
>> Merry Xmas
>>> On 2020-12-24 18:55, Duncan Turnbull wrote:
>> On 24/12/2020, at 6:39 PM, Turritopsis Dohrnii Teo En Ming 
>>  wrote:
> Hi Duncan Turnbull,
> I have finally managed to get my Cisco 7960 IP phone to register on my 
> Asterisk PBX appliance on Christmas Eve 2020.
> You can read my guide here:
> Guide: Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 
> 7960 IP Phones
> Link: 
> http://lists.digium.com/pipermail/asterisk-users/2020-December/295581.html
> However, there is still a problem. Please read Section 12. Do you know 
> how to solve it?? Thank you.
 You will need to look at the packets to see what the replies are
 It sounds like it could be a Do not disturb gone wrong
 But if you see the sip request going to the right place you can see
 the error response and that might provide more information
> Merry Christmas 2020!
 You too
> 
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> Check out the new Asterisk community forum at: 
>>> https://community.asterisk.org/
>>> New to Asterisk? Start here:
>>>https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> -- 
> -BEGIN EMAIL SIGNATURE-
> 
> The Gospel for all Targeted Individuals (TIs):
> 
> [The New York Times] Microwave Weapons Are Prime Suspect in Ills of
> U.S. Embassy Workers
> 
> Link: 
> https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html
> 
> 
> 
> Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic
> Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United 
> Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and 
> Australia (25 Dec 2019 to 9 Jan 2020):
> 
> [1] https://tdtemcerts.wordpress.com/
> 
> [2] https://tdtemcerts.blogspot.sg/
> 
> [3] https://www.scribd.com/user/270125049/Teo-En-Ming
> 
> -END EMAIL SIGNATURE-


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Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Turritopsis Dohrnii Teo En Ming]

Hi Duncan Turnbull,

My final issue has been resolved.

Please refer to the following post.

Post: Addendum to Teo En Ming's Guide to Configuring Asterisk/FreePBX 
with Cisco 7960 IP Phones
Link: 
http://lists.digium.com/pipermail/asterisk-users/2020-December/295590.html


Thank you very much.

Merry Christmas 2020!


On 2020-12-25 03:12, Duncan Turnbull wrote:
On 25/12/2020, at 12:40 AM, Turritopsis Dohrnii Teo En Ming 
 wrote:


Hi Duncan Turnbull,

It is a newly created PJSIP extension with default settings. I have 
never configured Do Not Disturb settings before.


Could it be something else?

Do I need to run tcpdump on the Asterisk PBX server again?


Running tcpdump shows you the packets. You need to see the packets
during call setup and failure to work out what’s happening. So yes you
should run tcpdump, and I would recommend sngrep as well as an easier
way to understand the sip conversations

Merry Xmas



On 2020-12-24 18:55, Duncan Turnbull wrote:
On 24/12/2020, at 6:39 PM, Turritopsis Dohrnii Teo En Ming 
 wrote:

Hi Duncan Turnbull,
I have finally managed to get my Cisco 7960 IP phone to register on 
my Asterisk PBX appliance on Christmas Eve 2020.

You can read my guide here:
Guide: Teo En Ming's Guide to Configuring Asterisk/FreePBX with 
Cisco 7960 IP Phones
Link: 
http://lists.digium.com/pipermail/asterisk-users/2020-December/295581.html
However, there is still a problem. Please read Section 12. Do you 
know how to solve it?? Thank you.

You will need to look at the packets to see what the replies are
It sounds like it could be a Do not disturb gone wrong
But if you see the sip request going to the right place you can see
the error response and that might provide more information

Merry Christmas 2020!

You too






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--
-BEGIN EMAIL SIGNATURE-

The Gospel for all Targeted Individuals (TIs):

[The New York Times] Microwave Weapons Are Prime Suspect in Ills of
U.S. Embassy Workers

Link: 
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html




Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's 
Academic
Qualifications as at 14 Feb 2019 and refugee seeking attempts at the 
United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 
2019) and Australia (25 Dec 2019 to 9 Jan 2020):


[1] https://tdtemcerts.wordpress.com/

[2] https://tdtemcerts.blogspot.sg/

[3] https://www.scribd.com/user/270125049/Teo-En-Ming

-END EMAIL SIGNATURE-

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Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Duncan Turnbull]

> On 25/12/2020, at 12:40 AM, Turritopsis Dohrnii Teo En Ming 
>  wrote:
> 
> Hi Duncan Turnbull,
> 
> It is a newly created PJSIP extension with default settings. I have never 
> configured Do Not Disturb settings before.
> 
> Could it be something else?
> 
> Do I need to run tcpdump on the Asterisk PBX server again?
> 
Running tcpdump shows you the packets. You need to see the packets during call 
setup and failure to work out what’s happening. So yes you should run tcpdump, 
and I would recommend sngrep as well as an easier way to understand the sip 
conversations 

Merry Xmas


> On 2020-12-24 18:55, Duncan Turnbull wrote:
 On 24/12/2020, at 6:39 PM, Turritopsis Dohrnii Teo En Ming 
  wrote:
>>> Hi Duncan Turnbull,
>>> I have finally managed to get my Cisco 7960 IP phone to register on my 
>>> Asterisk PBX appliance on Christmas Eve 2020.
>>> You can read my guide here:
>>> Guide: Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 7960 
>>> IP Phones
>>> Link: 
>>> http://lists.digium.com/pipermail/asterisk-users/2020-December/295581.html
>>> However, there is still a problem. Please read Section 12. Do you know how 
>>> to solve it?? Thank you.
>> You will need to look at the packets to see what the replies are
>> It sounds like it could be a Do not disturb gone wrong
>> But if you see the sip request going to the right place you can see
>> the error response and that might provide more information
>>> Merry Christmas 2020!
>> You too
> 
> -- 
> -BEGIN EMAIL SIGNATURE-
> 
> The Gospel for all Targeted Individuals (TIs):
> 
> [The New York Times] Microwave Weapons Are Prime Suspect in Ills of
> U.S. Embassy Workers
> 
> Link: 
> https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html
> 
> 
> 
> Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic
> Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United 
> Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and 
> Australia (25 Dec 2019 to 9 Jan 2020):
> 
> [1] https://tdtemcerts.wordpress.com/
> 
> [2] https://tdtemcerts.blogspot.sg/
> 
> [3] https://www.scribd.com/user/270125049/Teo-En-Ming
> 
> -END EMAIL SIGNATURE-
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> 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] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Turritopsis Dohrnii Teo En Ming]

Hi Duncan Turnbull,

It is a newly created PJSIP extension with default settings. I have 
never configured Do Not Disturb settings before.


Could it be something else?

Do I need to run tcpdump on the Asterisk PBX server again?

On 2020-12-24 18:55, Duncan Turnbull wrote:
On 24/12/2020, at 6:39 PM, Turritopsis Dohrnii Teo En Ming 
 wrote:


Hi Duncan Turnbull,

I have finally managed to get my Cisco 7960 IP phone to register on my 
Asterisk PBX appliance on Christmas Eve 2020.


You can read my guide here:

Guide: Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 
7960 IP Phones
Link: 
http://lists.digium.com/pipermail/asterisk-users/2020-December/295581.html


However, there is still a problem. Please read Section 12. Do you know 
how to solve it?? Thank you.




You will need to look at the packets to see what the replies are

It sounds like it could be a Do not disturb gone wrong

But if you see the sip request going to the right place you can see
the error response and that might provide more information


Merry Christmas 2020!



You too


--
-BEGIN EMAIL SIGNATURE-

The Gospel for all Targeted Individuals (TIs):

[The New York Times] Microwave Weapons Are Prime Suspect in Ills of
U.S. Embassy Workers

Link: 
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html




Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's 
Academic
Qualifications as at 14 Feb 2019 and refugee seeking attempts at the 
United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 
2019) and Australia (25 Dec 2019 to 9 Jan 2020):


[1] https://tdtemcerts.wordpress.com/

[2] https://tdtemcerts.blogspot.sg/

[3] https://www.scribd.com/user/270125049/Teo-En-Ming

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Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Duncan Turnbull]

> On 24/12/2020, at 6:39 PM, Turritopsis Dohrnii Teo En Ming 
>  wrote:
> 
> Hi Duncan Turnbull,
> 
> I have finally managed to get my Cisco 7960 IP phone to register on my 
> Asterisk PBX appliance on Christmas Eve 2020.
> 
> You can read my guide here:
> 
> Guide: Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 7960 IP 
> Phones
> Link: 
> http://lists.digium.com/pipermail/asterisk-users/2020-December/295581.html
> 
> However, there is still a problem. Please read Section 12. Do you know how to 
> solve it?? Thank you.
> 

You will need to look at the packets to see what the replies are

It sounds like it could be a Do not disturb gone wrong

But if you see the sip request going to the right place you can see the error 
response and that might provide more information 

> Merry Christmas 2020!
> 
> 
You too

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Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Turritopsis Dohrnii Teo En Ming]

Hi!

What is sngrep? I have never heard of it before.

Merry Christmas 2020!






On 2020-12-24 13:06, Steve Edwards wrote:

On Thu, 24 Dec 2020, Turritopsis Dohrnii Teo En Ming wrote:


3. secret is 8 char only, must be numeric


My my SIP.cnf file from 2007 contains:

image_version:  P0S3-8-12-00
line1_password: 346cc89a2526255839534c22ad7790c

and my notes say my 9760 only allowed up to 31 character passwords.

You may find it useful to use tcpdump with '-w' to write the packets
to a file and then analyze with sngrep.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 
PST

https://www.linkedin.com/in/steve-edwards-4244281


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The Gospel for all Targeted Individuals (TIs):

[The New York Times] Microwave Weapons Are Prime Suspect in Ills of
U.S. Embassy Workers

Link: 
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html




Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's 
Academic
Qualifications as at 14 Feb 2019 and refugee seeking attempts at the 
United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 
2019) and Australia (25 Dec 2019 to 9 Jan 2020):


[1] https://tdtemcerts.wordpress.com/

[2] https://tdtemcerts.blogspot.sg/

[3] https://www.scribd.com/user/270125049/Teo-En-Ming

-END EMAIL SIGNATURE-

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Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Turritopsis Dohrnii Teo En Ming]

Hi Duncan Turnbull,

I have finally managed to get my Cisco 7960 IP phone to register on my 
Asterisk PBX appliance on Christmas Eve 2020.


You can read my guide here:

Guide: Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 
7960 IP Phones
Link: 
http://lists.digium.com/pipermail/asterisk-users/2020-December/295581.html


However, there is still a problem. Please read Section 12. Do you know 
how to solve it?? Thank you.


Merry Christmas 2020!






On 2020-12-24 12:33, Duncan Turnbull wrote:

Hi Turritopsis

I think the key point maybe making sure the password doesn’t exceed
the capacity of the phone. So an 8 char password is a good idea

I would be surprised if pjsip doesn’t work but I haven’t tried it with
a Cisco phone

Whatever gets you working is what you want

Have a wonderful Xmas

Cheers Duncan

On 24/12/2020, at 1:11 PM, Turritopsis Dohrnii Teo En Ming 
 wrote:


Thank you for your replies, Duncan Turnbull.

I am going to run tcpdump on my Asterisk PBX server.

By the way, I found a Youtube video.

Youtube video: Cisco 7942g IP Phone Configuration on FreePBX 
In-Depth(Without Endpoint Manager)


Link: https://www.youtube.com/watch?v=gk6w8O3fZlc=youtu.be

From the above youtube video, it seems that I cannot use pjsip 
extension for my Cisco 7960 IP phone. I need to delete the pjsip 
extension, and then create a legacy chan_sip extension, it seems.


These are the notes I have taken after watching the above Youtube 
video:


1. Cannot use pjsip extension, need to use legacy chan_sip extension

2. Display name: Your name

3. secret is 8 char only, must be numeric

4. Voicemail: Enabled

5. Require from same extension: yes

6. Go to Advanced, nat mode: never

7. Port 5060

8. Qualify: No

9. Send RPID: Send Remote-Party-ID header

10. Go to Settings > Asterisk SIP Settings > SIP Legacy Settings 
(chan_sip)


11. NAT: No

12. Enable SRV Lookup: No

13. Edit SEP.cnf.xml, sipPort: 5160

14. Line #1, port: 5160




On 2020-12-23 17:55, Duncan Turnbull wrote:

Sent from my iPad
On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming 
 wrote:

Hi Duncan Turnbull,
You can watch my Youtube video of my Cisco 7960 IP phone.
The link is: https://www.youtube.com/watch?v=ip_F08jmmio
My Youtube video shows the Network Configuration settings, SIP 
Configuration settings and Status of my Cisco 7960 IP Phone.

The phone looks like it has picked up the configs however in the
status there are two error messages re parsing SipDefault.cnf and the
specific SIP..MAC.. file - you should try and remedy those errors .
Otherwise most of the settings look to be there
I would suggest cutting out as much of the config as you can
I would also suggest you run tcpdump on the 192.168.1.9 box and
monitor any traffic at all coming from your phone which is now on
192.168.1.130.  You may see the SIP messages there
Cheers Duncan

Did you see anything wrong?

On 2020-12-23 12:38, Duncan Turnbull wrote:
Hi there
On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming 
 wrote:

Good morning Duncan Turnbull,
I have posted my Asterisk PBX server debugging output previously 
in my original post. The link is:

http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html
I saw many REGISTER requests. Are these REGISTER requests from my 
Cisco 7960 IP phone? Could you help me to check? Thank you very 
much.
If they come from the phone they will have the phones ip address. 
The
phone will also try and register with the extension you have given 
it.

None of the registration messages appear to have the up or the
extension so you will need to figure out what’s gone wrong with the
phones config
That’s why checking the phone settings to see fit they have changed
helps understand if your configs were correct. You can do this via 
the
phone screen or telnet. It will take you some time to become 
familiar

with this but it’s worth it
Good luck

I shall reproduce my Asterisk PBX server debugging output below.
SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT
=
# asterisk -vvvr
sip set debug on
freepbx*CLI>
[2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 
sip_reregister:

-- Re-registration for  60...@sip.sg.didlogic.net
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 107.6.123.181:5060:
REGISTER sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
Max-Forwards: 70
From: ;tag=as6df6d977
To: 
Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
CSeq: 165 REGISTER
Supported: replaces, timer
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="60751", 
realm="sip.sg.didlogic.net",

algorithm=MD5, uri="sip:sip.sg.didlogic.net",
nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth,
cnonce="2b1b6d13", nc=0003
Expires: 120
Contact: 
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Steve Edwards]

On Thu, 24 Dec 2020, Turritopsis Dohrnii Teo En Ming wrote:


3. secret is 8 char only, must be numeric


My my SIP.cnf file from 2007 contains:

image_version:  P0S3-8-12-00
line1_password: 346cc89a2526255839534c22ad7790c

and my notes say my 9760 only allowed up to 31 character passwords.

You may find it useful to use tcpdump with '-w' to write the packets to a 
file and then analyze with sngrep.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Duncan Turnbull]

Hi Turritopsis

I think the key point maybe making sure the password doesn’t exceed the 
capacity of the phone. So an 8 char password is a good idea

I would be surprised if pjsip doesn’t work but I haven’t tried it with a Cisco 
phone

Whatever gets you working is what you want

Have a wonderful Xmas 

Cheers Duncan

> On 24/12/2020, at 1:11 PM, Turritopsis Dohrnii Teo En Ming 
>  wrote:
> 
> Thank you for your replies, Duncan Turnbull.
> 
> I am going to run tcpdump on my Asterisk PBX server.
> 
> By the way, I found a Youtube video.
> 
> Youtube video: Cisco 7942g IP Phone Configuration on FreePBX In-Depth(Without 
> Endpoint Manager)
> 
> Link: https://www.youtube.com/watch?v=gk6w8O3fZlc=youtu.be
> 
> From the above youtube video, it seems that I cannot use pjsip extension for 
> my Cisco 7960 IP phone. I need to delete the pjsip extension, and then create 
> a legacy chan_sip extension, it seems.
> 
> These are the notes I have taken after watching the above Youtube video:
> 
> 1. Cannot use pjsip extension, need to use legacy chan_sip extension
> 
> 2. Display name: Your name
> 
> 3. secret is 8 char only, must be numeric
> 
> 4. Voicemail: Enabled
> 
> 5. Require from same extension: yes
> 
> 6. Go to Advanced, nat mode: never
> 
> 7. Port 5060
> 
> 8. Qualify: No
> 
> 9. Send RPID: Send Remote-Party-ID header
> 
> 10. Go to Settings > Asterisk SIP Settings > SIP Legacy Settings (chan_sip)
> 
> 11. NAT: No
> 
> 12. Enable SRV Lookup: No
> 
> 13. Edit SEP.cnf.xml, sipPort: 5160
> 
> 14. Line #1, port: 5160
> 
> 
> 
>> On 2020-12-23 17:55, Duncan Turnbull wrote:
>> 
>> Sent from my iPad
 On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming 
  wrote:
>>> Hi Duncan Turnbull,
>>> You can watch my Youtube video of my Cisco 7960 IP phone.
>>> The link is: https://www.youtube.com/watch?v=ip_F08jmmio
>>> My Youtube video shows the Network Configuration settings, SIP 
>>> Configuration settings and Status of my Cisco 7960 IP Phone.
>> The phone looks like it has picked up the configs however in the
>> status there are two error messages re parsing SipDefault.cnf and the
>> specific SIP..MAC.. file - you should try and remedy those errors .
>> Otherwise most of the settings look to be there
>> I would suggest cutting out as much of the config as you can
>> I would also suggest you run tcpdump on the 192.168.1.9 box and
>> monitor any traffic at all coming from your phone which is now on
>> 192.168.1.130.  You may see the SIP messages there
>> Cheers Duncan
>>> Did you see anything wrong?
 On 2020-12-23 12:38, Duncan Turnbull wrote:
 Hi there
>> On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming 
>>  wrote:
> Good morning Duncan Turnbull,
> I have posted my Asterisk PBX server debugging output previously in my 
> original post. The link is:
> http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html
> I saw many REGISTER requests. Are these REGISTER requests from my Cisco 
> 7960 IP phone? Could you help me to check? Thank you very much.
 If they come from the phone they will have the phones ip address. The
 phone will also try and register with the extension you have given it.
 None of the registration messages appear to have the up or the
 extension so you will need to figure out what’s gone wrong with the
 phones config
 That’s why checking the phone settings to see fit they have changed
 helps understand if your configs were correct. You can do this via the
 phone screen or telnet. It will take you some time to become familiar
 with this but it’s worth it
 Good luck
> I shall reproduce my Asterisk PBX server debugging output below.
> SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT
> =
> # asterisk -vvvr
> sip set debug on
> freepbx*CLI>
> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
> -- Re-registration for  60...@sip.sg.didlogic.net
> REGISTER 12 headers, 0 lines
> Reliably Transmitting (NAT) to 107.6.123.181:5060:
> REGISTER sip:sip.sg.didlogic.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
> Max-Forwards: 70
> From: ;tag=as6df6d977
> To: 
> Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
> CSeq: 165 REGISTER
> Supported: replaces, timer
> User-Agent: FPBX-15.0.16.81(16.13.0)
> Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
> nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
> response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth,
> cnonce="2b1b6d13", nc=0003
> Expires: 120
> Contact: 
> Content-Length: 0
> ---
> <--- SIP read from UDP:107.6.123.181:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received= OFFICE PUBLIC IP>
> 

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Turritopsis Dohrnii Teo En Ming]

Thank you for your replies, Duncan Turnbull.

I am going to run tcpdump on my Asterisk PBX server.

By the way, I found a Youtube video.

Youtube video: Cisco 7942g IP Phone Configuration on FreePBX 
In-Depth(Without Endpoint Manager)


Link: https://www.youtube.com/watch?v=gk6w8O3fZlc=youtu.be

From the above youtube video, it seems that I cannot use pjsip extension 
for my Cisco 7960 IP phone. I need to delete the pjsip extension, and 
then create a legacy chan_sip extension, it seems.


These are the notes I have taken after watching the above Youtube video:

1. Cannot use pjsip extension, need to use legacy chan_sip extension

2. Display name: Your name

3. secret is 8 char only, must be numeric

4. Voicemail: Enabled

5. Require from same extension: yes

6. Go to Advanced, nat mode: never

7. Port 5060

8. Qualify: No

9. Send RPID: Send Remote-Party-ID header

10. Go to Settings > Asterisk SIP Settings > SIP Legacy Settings 
(chan_sip)


11. NAT: No

12. Enable SRV Lookup: No

13. Edit SEP.cnf.xml, sipPort: 5160

14. Line #1, port: 5160



On 2020-12-23 17:55, Duncan Turnbull wrote:



Sent from my iPad

On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming 
 wrote:


Hi Duncan Turnbull,

You can watch my Youtube video of my Cisco 7960 IP phone.

The link is: https://www.youtube.com/watch?v=ip_F08jmmio

My Youtube video shows the Network Configuration settings, SIP 
Configuration settings and Status of my Cisco 7960 IP Phone.

The phone looks like it has picked up the configs however in the
status there are two error messages re parsing SipDefault.cnf and the
specific SIP..MAC.. file - you should try and remedy those errors .
Otherwise most of the settings look to be there

I would suggest cutting out as much of the config as you can

I would also suggest you run tcpdump on the 192.168.1.9 box and
monitor any traffic at all coming from your phone which is now on
192.168.1.130.  You may see the SIP messages there

Cheers Duncan


Did you see anything wrong?





On 2020-12-23 12:38, Duncan Turnbull wrote:
Hi there
On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming 
 wrote:

Good morning Duncan Turnbull,
I have posted my Asterisk PBX server debugging output previously in 
my original post. The link is:

http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html
I saw many REGISTER requests. Are these REGISTER requests from my 
Cisco 7960 IP phone? Could you help me to check? Thank you very 
much.

If they come from the phone they will have the phones ip address. The
phone will also try and register with the extension you have given 
it.

None of the registration messages appear to have the up or the
extension so you will need to figure out what’s gone wrong with the
phones config
That’s why checking the phone settings to see fit they have changed
helps understand if your configs were correct. You can do this via 
the

phone screen or telnet. It will take you some time to become familiar
with this but it’s worth it
Good luck

I shall reproduce my Asterisk PBX server debugging output below.
SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT
=
# asterisk -vvvr
sip set debug on
freepbx*CLI>
[2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
-- Re-registration for  60...@sip.sg.didlogic.net
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 107.6.123.181:5060:
REGISTER sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
Max-Forwards: 70
From: ;tag=as6df6d977
To: 
Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
CSeq: 165 REGISTER
Supported: replaces, timer
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
algorithm=MD5, uri="sip:sip.sg.didlogic.net",
nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth,
cnonce="2b1b6d13", nc=0003
Expires: 120
Contact: 
Content-Length: 0
---
<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received=
From: ;tag=as6df6d977
To:
;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21
Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
CSeq: 165 REGISTER
Contact: IP>:26462>;expires=120;received="sip:IP>:26462"

Content-Length: 0
<->
--- (8 headers 0 lines) ---
[2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961
handle_response_register: Outbound Registration: Expiry for
sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog
'005dbc8238e06ac421ef613a3b55e134@127.0.0.1' Method: REGISTER
<--- SIP read from UDP:107.6.123.181:5060 --->
<->
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK51105854;rport
Max-Forwards: 70
From: "Unknown" ;tag=as41ddf4a6
To: 
Contact: 
Call-ID: 

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Duncan Turnbull]



Sent from my iPad

> On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming 
>  wrote:
> 
> Hi Duncan Turnbull,
> 
> You can watch my Youtube video of my Cisco 7960 IP phone.
> 
> The link is: https://www.youtube.com/watch?v=ip_F08jmmio
> 
> My Youtube video shows the Network Configuration settings, SIP Configuration 
> settings and Status of my Cisco 7960 IP Phone.
The phone looks like it has picked up the configs however in the status there 
are two error messages re parsing SipDefault.cnf and the specific SIP..MAC.. 
file - you should try and remedy those errors . Otherwise most of the settings 
look to be there

I would suggest cutting out as much of the config as you can

I would also suggest you run tcpdump on the 192.168.1.9 box and monitor any 
traffic at all coming from your phone which is now on 192.168.1.130.  You may 
see the SIP messages there

Cheers Duncan

> Did you see anything wrong?
> 
> 
> 
> 
>> On 2020-12-23 12:38, Duncan Turnbull wrote:
>> Hi there
 On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming 
  wrote:
>>> Good morning Duncan Turnbull,
>>> I have posted my Asterisk PBX server debugging output previously in my 
>>> original post. The link is:
>>> http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html
>>> I saw many REGISTER requests. Are these REGISTER requests from my Cisco 
>>> 7960 IP phone? Could you help me to check? Thank you very much.
>> If they come from the phone they will have the phones ip address. The
>> phone will also try and register with the extension you have given it.
>> None of the registration messages appear to have the up or the
>> extension so you will need to figure out what’s gone wrong with the
>> phones config
>> That’s why checking the phone settings to see fit they have changed
>> helps understand if your configs were correct. You can do this via the
>> phone screen or telnet. It will take you some time to become familiar
>> with this but it’s worth it
>> Good luck
>>> I shall reproduce my Asterisk PBX server debugging output below.
>>> SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT
>>> =
>>> # asterisk -vvvr
>>> sip set debug on
>>> freepbx*CLI>
>>> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
>>> -- Re-registration for  60...@sip.sg.didlogic.net
>>> REGISTER 12 headers, 0 lines
>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>> REGISTER sip:sip.sg.didlogic.net SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
>>> Max-Forwards: 70
>>> From: ;tag=as6df6d977
>>> To: 
>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
>>> CSeq: 165 REGISTER
>>> Supported: replaces, timer
>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>> Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
>>> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
>>> nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
>>> response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth,
>>> cnonce="2b1b6d13", nc=0003
>>> Expires: 120
>>> Contact: 
>>> Content-Length: 0
>>> ---
>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP
>>> 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received=>> OFFICE PUBLIC IP>
>>> From: ;tag=as6df6d977
>>> To:
>>> ;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21
>>> Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
>>> CSeq: 165 REGISTER
>>> Contact: >> IP>:26462>;expires=120;received="sip::26462"
>>> Content-Length: 0
>>> <->
>>> --- (8 headers 0 lines) ---
>>> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961
>>> handle_response_register: Outbound Registration: Expiry for
>>> sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s)
>>> Really destroying SIP dialog
>>> '005dbc8238e06ac421ef613a3b55e134@127.0.0.1' Method: REGISTER
>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>> <->
>>> Reliably Transmitting (NAT) to 107.6.123.181:5060:
>>> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK51105854;rport
>>> Max-Forwards: 70
>>> From: "Unknown" ;tag=as41ddf4a6
>>> To: 
>>> Contact: 
>>> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160
>>> CSeq: 102 OPTIONS
>>> User-Agent: FPBX-15.0.16.81(16.13.0)
>>> Date: Sun, 20 Dec 2020 07:07:07 GMT
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>> INFO, PUBLISH, MESSAGE
>>> Supported: replaces, timer
>>> Content-Length: 0
>>> ---
>>> <--- SIP read from UDP:107.6.123.181:5060 --->
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP
>>> 192.168.1.9:5160;branch=z9hG4bK51105854;rport=26462;received=>> OFFICE PUBLIC IP>
>>> From: "Unknown" ;tag=as41ddf4a6
>>> To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924
>>> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160
>>> CSeq: 102 OPTIONS
>>> Content-Length: 0
>>> <->
>>> --- (7 headers 0 lines) ---
>>> Really destroying SIP dialog
>>> 

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Turritopsis Dohrnii Teo En Ming]

Hi Duncan Turnbull,

You can watch my Youtube video of my Cisco 7960 IP phone.

The link is: https://www.youtube.com/watch?v=ip_F08jmmio

My Youtube video shows the Network Configuration settings, SIP 
Configuration settings and Status of my Cisco 7960 IP Phone.


Did you see anything wrong?




On 2020-12-23 12:38, Duncan Turnbull wrote:

Hi there

On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming 
 wrote:


Good morning Duncan Turnbull,

I have posted my Asterisk PBX server debugging output previously in my 
original post. The link is:


http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html

I saw many REGISTER requests. Are these REGISTER requests from my 
Cisco 7960 IP phone? Could you help me to check? Thank you very much.



If they come from the phone they will have the phones ip address. The
phone will also try and register with the extension you have given it.
None of the registration messages appear to have the up or the
extension so you will need to figure out what’s gone wrong with the
phones config

That’s why checking the phone settings to see fit they have changed
helps understand if your configs were correct. You can do this via the
phone screen or telnet. It will take you some time to become familiar
with this but it’s worth it

Good luck


I shall reproduce my Asterisk PBX server debugging output below.

SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT
=

# asterisk -vvvr

sip set debug on

freepbx*CLI>
[2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
-- Re-registration for  60...@sip.sg.didlogic.net
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 107.6.123.181:5060:
REGISTER sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
Max-Forwards: 70
From: ;tag=as6df6d977
To: 
Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
CSeq: 165 REGISTER
Supported: replaces, timer
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
algorithm=MD5, uri="sip:sip.sg.didlogic.net",
nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth,
cnonce="2b1b6d13", nc=0003
Expires: 120
Contact: 
Content-Length: 0


---

<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received=
From: ;tag=as6df6d977
To:
;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21
Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
CSeq: 165 REGISTER
Contact: IP>:26462>;expires=120;received="sip:IP>:26462"

Content-Length: 0

<->
--- (8 headers 0 lines) ---
[2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961
handle_response_register: Outbound Registration: Expiry for
sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog
'005dbc8238e06ac421ef613a3b55e134@127.0.0.1' Method: REGISTER

<--- SIP read from UDP:107.6.123.181:5060 --->

<->
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK51105854;rport
Max-Forwards: 70
From: "Unknown" ;tag=as41ddf4a6
To: 
Contact: 
Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sun, 20 Dec 2020 07:07:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK51105854;rport=26462;received=
From: "Unknown" ;tag=as41ddf4a6
To: 
;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924

Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160
CSeq: 102 OPTIONS
Content-Length: 0

<->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160' Method: OPTIONS

<--- SIP read from UDP:107.6.123.181:5060 --->

<->

<--- SIP read from UDP:107.6.123.181:5060 --->

<->
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK4ff08179;rport
Max-Forwards: 70
From: "Unknown" ;tag=as004073f3
To: 
Contact: 
Call-ID: 500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sun, 20 Dec 2020 07:08:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK4ff08179;rport=26462;received=
From: "Unknown" ;tag=as004073f3
To: 
;tag=b27e1a1d33761e85846fc98f5f3a7e58.385d

Call-ID: 500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160
CSeq: 102 OPTIONS
Content-Length: 0

<->
--- (7 headers 

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Duncan Turnbull]

Hi there

> On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming 
>  wrote:
> 
> Good morning Duncan Turnbull,
> 
> I have posted my Asterisk PBX server debugging output previously in my 
> original post. The link is:
> 
> http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html
> 
> I saw many REGISTER requests. Are these REGISTER requests from my Cisco 7960 
> IP phone? Could you help me to check? Thank you very much.
> 
If they come from the phone they will have the phones ip address. The phone 
will also try and register with the extension you have given it. None of the 
registration messages appear to have the up or the extension so you will need 
to figure out what’s gone wrong with the phones config

That’s why checking the phone settings to see fit they have changed helps 
understand if your configs were correct. You can do this via the phone screen 
or telnet. It will take you some time to become familiar with this but it’s 
worth it

Good luck

> I shall reproduce my Asterisk PBX server debugging output below.
> 
> SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT
> =
> 
> # asterisk -vvvr
> 
> sip set debug on
> 
> freepbx*CLI>
> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
> -- Re-registration for  60...@sip.sg.didlogic.net
> REGISTER 12 headers, 0 lines
> Reliably Transmitting (NAT) to 107.6.123.181:5060:
> REGISTER sip:sip.sg.didlogic.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
> Max-Forwards: 70
> From: ;tag=as6df6d977
> To: 
> Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
> CSeq: 165 REGISTER
> Supported: replaces, timer
> User-Agent: FPBX-15.0.16.81(16.13.0)
> Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
> algorithm=MD5, uri="sip:sip.sg.didlogic.net",
> nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
> response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth,
> cnonce="2b1b6d13", nc=0003
> Expires: 120
> Contact: 
> Content-Length: 0
> 
> 
> ---
> 
> <--- SIP read from UDP:107.6.123.181:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received= OFFICE PUBLIC IP>
> From: ;tag=as6df6d977
> To:
> ;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21
> Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
> CSeq: 165 REGISTER
> Contact:  IP>:26462>;expires=120;received="sip::26462"
> Content-Length: 0
> 
> <->
> --- (8 headers 0 lines) ---
> [2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961
> handle_response_register: Outbound Registration: Expiry for
> sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s)
> Really destroying SIP dialog
> '005dbc8238e06ac421ef613a3b55e134@127.0.0.1' Method: REGISTER
> 
> <--- SIP read from UDP:107.6.123.181:5060 --->
> 
> <->
> Reliably Transmitting (NAT) to 107.6.123.181:5060:
> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK51105854;rport
> Max-Forwards: 70
> From: "Unknown" ;tag=as41ddf4a6
> To: 
> Contact: 
> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160
> CSeq: 102 OPTIONS
> User-Agent: FPBX-15.0.16.81(16.13.0)
> Date: Sun, 20 Dec 2020 07:07:07 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
> 
> 
> ---
> 
> <--- SIP read from UDP:107.6.123.181:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.9:5160;branch=z9hG4bK51105854;rport=26462;received= OFFICE PUBLIC IP>
> From: "Unknown" ;tag=as41ddf4a6
> To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924
> Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160
> CSeq: 102 OPTIONS
> Content-Length: 0
> 
> <->
> --- (7 headers 0 lines) ---
> Really destroying SIP dialog
> '0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160' Method: OPTIONS
> 
> <--- SIP read from UDP:107.6.123.181:5060 --->
> 
> <->
> 
> <--- SIP read from UDP:107.6.123.181:5060 --->
> 
> <->
> Reliably Transmitting (NAT) to 107.6.123.181:5060:
> OPTIONS sip:sip.sg.didlogic.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK4ff08179;rport
> Max-Forwards: 70
> From: "Unknown" ;tag=as004073f3
> To: 
> Contact: 
> Call-ID: 500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160
> CSeq: 102 OPTIONS
> User-Agent: FPBX-15.0.16.81(16.13.0)
> Date: Sun, 20 Dec 2020 07:08:07 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
> 
> 
> ---
> 
> <--- SIP read from UDP:107.6.123.181:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.9:5160;branch=z9hG4bK4ff08179;rport=26462;received= OFFICE PUBLIC IP>
> From: "Unknown" ;tag=as004073f3
> To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.385d
> Call-ID: 500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160
> CSeq: 102 OPTIONS
> Content-Length: 0
> 
> <->
> --- (7 headers 0 lines) 

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Turritopsis Dohrnii Teo En Ming]

Good morning Duncan Turnbull,

I have posted my Asterisk PBX server debugging output previously in my 
original post. The link is:


http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html

I saw many REGISTER requests. Are these REGISTER requests from my Cisco 
7960 IP phone? Could you help me to check? Thank you very much.


I shall reproduce my Asterisk PBX server debugging output below.

SECTION: ASTERISK PBX SERVER DEBUGGING OUTPUT
=

# asterisk -vvvr

sip set debug on

freepbx*CLI>
[2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
-- Re-registration for  60...@sip.sg.didlogic.net
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 107.6.123.181:5060:
REGISTER sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport
Max-Forwards: 70
 From: ;tag=as6df6d977
To: 
Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
CSeq: 165 REGISTER
Supported: replaces, timer
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
algorithm=MD5, uri="sip:sip.sg.didlogic.net",
nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
response="bfadacdd4e745fd4b9e12046e6ce2afc", qop=auth,
cnonce="2b1b6d13", nc=0003
Expires: 120
Contact: 
Content-Length: 0


---

<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK3f11a8b8;rport=26462;received=
 From: ;tag=as6df6d977
To:
;tag=b27e1a1d33761e85846fc98f5f3a7e58.4d21
Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
CSeq: 165 REGISTER
Contact: :26462>;expires=120;received="sip::26462"
Content-Length: 0

<->
--- (8 headers 0 lines) ---
[2020-12-20 07:06:22] NOTICE[2366]: chan_sip.c:24961
handle_response_register: Outbound Registration: Expiry for
sip.sg.didlogic.net is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog
'005dbc8238e06ac421ef613a3b55e134@127.0.0.1' Method: REGISTER

<--- SIP read from UDP:107.6.123.181:5060 --->

<->
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK51105854;rport
Max-Forwards: 70
 From: "Unknown" ;tag=as41ddf4a6
To: 
Contact: 
Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sun, 20 Dec 2020 07:07:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK51105854;rport=26462;received=
 From: "Unknown" ;tag=as41ddf4a6
To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.f924
Call-ID: 0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160
CSeq: 102 OPTIONS
Content-Length: 0

<->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'0b0605df4f4ca7b03402c9fd5a869606@192.168.1.9:5160' Method: OPTIONS

<--- SIP read from UDP:107.6.123.181:5060 --->

<->

<--- SIP read from UDP:107.6.123.181:5060 --->

<->
Reliably Transmitting (NAT) to 107.6.123.181:5060:
OPTIONS sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK4ff08179;rport
Max-Forwards: 70
 From: "Unknown" ;tag=as004073f3
To: 
Contact: 
Call-ID: 500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sun, 20 Dec 2020 07:08:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.9:5160;branch=z9hG4bK4ff08179;rport=26462;received=
 From: "Unknown" ;tag=as004073f3
To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.385d
Call-ID: 500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160
CSeq: 102 OPTIONS
Content-Length: 0

<->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'500c8eb32071e3fc462be80f243d38fc@192.168.1.9:5160' Method: OPTIONS
[2020-12-20 07:08:07] NOTICE[2366]: chan_sip.c:15893 sip_reregister:
-- Re-registration for  60...@sip.sg.didlogic.net
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 107.6.123.181:5060:
REGISTER sip:sip.sg.didlogic.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5160;branch=z9hG4bK6d85e46f;rport
Max-Forwards: 70
 From: ;tag=as6df6d977
To: 
Call-ID: 005dbc8238e06ac421ef613a3b55e134@127.0.0.1
CSeq: 166 REGISTER
Supported: replaces, timer
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="60751", realm="sip.sg.didlogic.net",
algorithm=MD5, uri="sip:sip.sg.didlogic.net",
nonce="X974SF/e9xyVB6XKqpfatDHcb8chw9fPak+Ke4A=",
response="074f1f037639144de751dc9231c191c9", qop=auth,
cnonce="6eb58a86", nc=0004
Expires: 120
Contact: 
Content-Length: 0


---

<--- SIP read from UDP:107.6.123.181:5060 --->
SIP/2.0 401 Unauthorized

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Duncan Turnbull]

Hi there

That answer includes using tcpdump to check for SIP packets and examine the
register packet. At this point you have no SIP packets coming from your
phone so you are not upto that stage yet.

You need to know why there are no SIP packets coming. My guess is your
config files have a typo in them. You can modify the ones I sent to see if
they work for you. You do need to validate your configs. I recommend telnet
to the phone, and checking the display settings to see if it has picked up
the settings. Equally check the phone logs. It will tell you which files
have errors.

Until you get the settings loaded correctly nothing else will matter

Enjoy your break. If you want to use Voip you should definitely spend some
time to learn tcpdump. If you want to use Cisco you need to be able to
understand the configs yourself and get as much info from the phone as
possible. Not hard but it takes a little bit of time.

Cheers Duncan

On Tue, Dec 22, 2020 at 10:43 PM Turritopsis Dohrnii Teo En Ming <
c...@teo-en-ming.com> wrote:

> Good day from Singapore,
>
> I seem to have found the solution at FreePBX community forums. Please
> check out the following discussion thread.
>
> Discussion Thread: Cisco 7940 registration problem RESOLVED
> Link:
>
> https://community.freepbx.org/t/cisco-7940-registration-problem-resolved/30285
>
> But I don't understand very well what users at this discussion thread
> are talking about. Can someone help me understand better after reading
> through the above discussion thread?
>
> For your information, I am using PJSIP extension instead of CHAN_SIP
> extension.
>
> I am planning to work on my Cisco 7960 IP phone registration problem
> this coming Christmas 2020 weekends.
>
> Thank you very much for your kind assistance.
>
>
>
>
> On 2020-12-21 09:58, Duncan Turnbull wrote:
> > Hi there
> >
> > I would normally highlight the part but the email is so long I thought
> > I would just note what I can see
> >
> > It appears the Cisco is downloading files.
> > None of the SIP traces show the IP of the phone of the extension
> >
> > Your proxy is at 192.168.1.9
> > Your phone is at 192.168.1.130
> >
> > These are the details you want the phone to pickup
> > line1_name: "1600"
> > line1_shortname: "TEO EN MING"
> > line1_displayname: "TURRITOPSIS DOHRNII TEO EN MING"
> > line1_authname: "1600"
> > line1_password: "IP Phone Extension Password"
> >
> >  I don't see any registration attempts from your phone.
> >
> > The first thing is to use the phone screen display to check if it
> > actually has picked up the settings.
> > To unlock the Cisco SIP IP phone, press **#
> >
> > You can also telnet to the phone usually cisco as password, and look
> > at logs. Its quite possible some of your config files are not quite
> > right. If they were all wrong the Cisco would keep trying to TFTP the
> > files.
> >
> > This is an old SIPDefault.cnf I used to use in NZ
> > 
> >
> > ; sip default configuration file
> > #Image Version
> > image_version:P0S3-08-6-00 ;
> > #Proxy server address
> > proxy1_address: 10.12.41.1 ;
> > proxy_register: 1;
> > logo_url: "http://10.12.41.1/Logo.bmp;; URL for
> > branding logo to be used on phone display
> > time_format: 0 ;
> > preferred_codec: g711alaw ;
> > sntp_mode: unicast ;
> > dial_template: dialplan
> > sntp_server: 10.12.41.1 ;
> > messages_uri: "*97"
> > time_zone : NZST
> > dst_auto_adjust : 1
> > dst_offset : 01
> > dst_start_month : September
> > dst_start_day : 29
> > dst_start_time : 02:00
> > dst_stop_month : April
> > dst_stop_day : 6
> > dst_stop_time : 02:00
> > =
> >
> > This is a SIP Phone template - you can compare settings and notes on
> > the settings
> > # SIP Configuration Generic File (start)
> >
> > =
> > # Proxy Server
> > proxy1_address: "10.12.41.1"
> >
> > # Line 1 Settings
> > line1_name: "EXTN" ; Line 1 Extension\User ID
> > line1_shortname: "0NXXX"   ; Line 1 Short Name
> > line1_displayname: "0NXXX"   ; Line 1 Display Name
> > line1_authname: "EXTN" ; Line 1 Registration Authentication
> > line1_password: "6gs72ha9" ; Line 1 Registration Password
> > phone_label: "Company Limited"; no effect on SIP messaging
> >
> > # Line 2 Settings
> > line2_name: ""  ; Line 2 Extension\User ID
> > line2_displayname: ""   ; Line 2 Display Name
> > line2_authname: "UNPROVISIONED" ; Line 2 Registration
> > Authentication
> > line2_password: "UNPROVISIONED" ; Line 2 Registration Password
> >
> > # Line 3 Settings
> > line3_name: ""  ; Line 3 Extension\User ID
> > line3_displayname: ""   ; Line 3 Display Name
> > line3_authname: "UNPROVISIONED" ; Line 3 Registration
> > Authentication
> > line3_password: "UNPROVISIONED" 

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Turritopsis Dohrnii Teo En Ming]

Good day from Singapore,

I seem to have found the solution at FreePBX community forums. Please 
check out the following discussion thread.


Discussion Thread: Cisco 7940 registration problem RESOLVED
Link: 
https://community.freepbx.org/t/cisco-7940-registration-problem-resolved/30285


But I don't understand very well what users at this discussion thread 
are talking about. Can someone help me understand better after reading 
through the above discussion thread?


For your information, I am using PJSIP extension instead of CHAN_SIP 
extension.


I am planning to work on my Cisco 7960 IP phone registration problem 
this coming Christmas 2020 weekends.


Thank you very much for your kind assistance.




On 2020-12-21 09:58, Duncan Turnbull wrote:

Hi there

I would normally highlight the part but the email is so long I thought
I would just note what I can see

It appears the Cisco is downloading files.
None of the SIP traces show the IP of the phone of the extension

Your proxy is at 192.168.1.9
Your phone is at 192.168.1.130

These are the details you want the phone to pickup
line1_name: "1600"
line1_shortname: "TEO EN MING"
line1_displayname: "TURRITOPSIS DOHRNII TEO EN MING"
line1_authname: "1600"
line1_password: "IP Phone Extension Password"

 I don't see any registration attempts from your phone.

The first thing is to use the phone screen display to check if it
actually has picked up the settings.
To unlock the Cisco SIP IP phone, press **#

You can also telnet to the phone usually cisco as password, and look
at logs. Its quite possible some of your config files are not quite
right. If they were all wrong the Cisco would keep trying to TFTP the
files.

This is an old SIPDefault.cnf I used to use in NZ


; sip default configuration file
#Image Version
image_version:P0S3-08-6-00 ;
#Proxy server address
proxy1_address: 10.12.41.1 ;
proxy_register: 1;
logo_url: "http://10.12.41.1/Logo.bmp;; URL for
branding logo to be used on phone display
time_format: 0 ;
preferred_codec: g711alaw ;
sntp_mode: unicast ;
dial_template: dialplan
sntp_server: 10.12.41.1 ;
messages_uri: "*97"
time_zone : NZST
dst_auto_adjust : 1
dst_offset : 01
dst_start_month : September
dst_start_day : 29
dst_start_time : 02:00
dst_stop_month : April
dst_stop_day : 6
dst_stop_time : 02:00
=

This is a SIP Phone template - you can compare settings and notes on
the settings
# SIP Configuration Generic File (start)

=
# Proxy Server
proxy1_address: "10.12.41.1"

# Line 1 Settings
line1_name: "EXTN" ; Line 1 Extension\User ID
line1_shortname: "0NXXX"   ; Line 1 Short Name
line1_displayname: "0NXXX"   ; Line 1 Display Name
line1_authname: "EXTN" ; Line 1 Registration Authentication
line1_password: "6gs72ha9" ; Line 1 Registration Password
phone_label: "Company Limited"; no effect on SIP messaging

# Line 2 Settings
line2_name: ""  ; Line 2 Extension\User ID
line2_displayname: ""   ; Line 2 Display Name
line2_authname: "UNPROVISIONED" ; Line 2 Registration 
Authentication

line2_password: "UNPROVISIONED" ; Line 2 Registration Password

# Line 3 Settings
line3_name: ""  ; Line 3 Extension\User ID
line3_displayname: ""   ; Line 3 Display Name
line3_authname: "UNPROVISIONED" ; Line 3 Registration 
Authentication

line3_password: "UNPROVISIONED" ; Line 3 Registration Password

# Line 4 Settings
line4_name: ""  ; Line 4 Extension\User ID
line4_displayname: ""   ; Line 4 Display Name
line4_authname: "UNPROVISIONED" ; Line 4 Registration 
Authentication

line4_password: "UNPROVISIONED" ; Line 4 Registration Password

# Line 5 Settings
line5_name: ""  ; Line 5 Extension\User ID
line5_displayname: ""   ; Line 5 Display Name
line5_authname: "UNPROVISIONED" ; Line 5 Registration 
Authentication

line5_password: "UNPROVISIONED" ; Line 5 Registration Password

# Line 6 Settings
line6_name: ""  ; Line 6 Extension\User ID
line6_displayname: ""   ; Line 6 Display Name
line6_authname: "UNPROVISIONE" ; Line 6 Registration 
Authentication

line6_password: "UNPROVISIONE" ; Line 6 Registration Password

# Emergency Proxy info
proxy_emergency: ""
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: ""
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: "5060"

# NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
start_media_port: "1"
end_media_port:  "2"
nat_received_processing: "0"

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Name's phone" 

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Duncan Turnbull]

Hi there

I would normally highlight the part but the email is so long I thought I 
would just note what I can see


It appears the Cisco is downloading files.
None of the SIP traces show the IP of the phone of the extension

Your proxy is at 192.168.1.9
Your phone is at 192.168.1.130

These are the details you want the phone to pickup
line1_name: "1600"
line1_shortname: "TEO EN MING"
line1_displayname: "TURRITOPSIS DOHRNII TEO EN MING"
line1_authname: "1600"
line1_password: "IP Phone Extension Password"

 I don't see any registration attempts from your phone.

The first thing is to use the phone screen display to check if it 
actually has picked up the settings.

To unlock the Cisco SIP IP phone, press **#

You can also telnet to the phone usually cisco as password, and look at 
logs. Its quite possible some of your config files are not quite right. 
If they were all wrong the Cisco would keep trying to TFTP the files.


This is an old SIPDefault.cnf I used to use in NZ


; sip default configuration file
#Image Version
image_version:P0S3-08-6-00 ;
#Proxy server address
proxy1_address: 10.12.41.1 ;
proxy_register: 1;
logo_url: "http://10.12.41.1/Logo.bmp;; URL for 
branding logo to be used on phone display

time_format: 0 ;
preferred_codec: g711alaw ;
sntp_mode: unicast ;
dial_template: dialplan
sntp_server: 10.12.41.1 ;
messages_uri: "*97"
time_zone : NZST
dst_auto_adjust : 1
dst_offset : 01
dst_start_month : September
dst_start_day : 29
dst_start_time : 02:00
dst_stop_month : April
dst_stop_day : 6
dst_stop_time : 02:00
=

This is a SIP Phone template - you can compare settings and notes on the 
settings

# SIP Configuration Generic File (start)

=
# Proxy Server
proxy1_address: "10.12.41.1"

# Line 1 Settings
line1_name: "EXTN" ; Line 1 Extension\User ID
line1_shortname: "0NXXX"   ; Line 1 Short Name
line1_displayname: "0NXXX"   ; Line 1 Display Name
line1_authname: "EXTN" ; Line 1 Registration Authentication
line1_password: "6gs72ha9" ; Line 1 Registration Password
phone_label: "Company Limited"; no effect on SIP messaging

# Line 2 Settings
line2_name: ""  ; Line 2 Extension\User ID
line2_displayname: ""   ; Line 2 Display Name
line2_authname: "UNPROVISIONED" ; Line 2 Registration 
Authentication

line2_password: "UNPROVISIONED" ; Line 2 Registration Password

# Line 3 Settings
line3_name: ""  ; Line 3 Extension\User ID
line3_displayname: ""   ; Line 3 Display Name
line3_authname: "UNPROVISIONED" ; Line 3 Registration 
Authentication

line3_password: "UNPROVISIONED" ; Line 3 Registration Password

# Line 4 Settings
line4_name: ""  ; Line 4 Extension\User ID
line4_displayname: ""   ; Line 4 Display Name
line4_authname: "UNPROVISIONED" ; Line 4 Registration 
Authentication

line4_password: "UNPROVISIONED" ; Line 4 Registration Password

# Line 5 Settings
line5_name: ""  ; Line 5 Extension\User ID
line5_displayname: ""   ; Line 5 Display Name
line5_authname: "UNPROVISIONED" ; Line 5 Registration 
Authentication

line5_password: "UNPROVISIONED" ; Line 5 Registration Password

# Line 6 Settings
line6_name: ""  ; Line 6 Extension\User ID
line6_displayname: ""   ; Line 6 Display Name
line6_authname: "UNPROVISIONE" ; Line 6 Registration 
Authentication

line6_password: "UNPROVISIONE" ; Line 6 Registration Password

# Emergency Proxy info
proxy_emergency: ""
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: ""
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: "5060"

# NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
start_media_port: "1"
end_media_port:  "2"
nat_received_processing: "0"

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Name's phone"; Has no effect on SIP messaging

# Time Zone phone will reside in
time_zone: NZST
time_format: "D/M/Ya"
# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2"  ; 0-Disabled (default), 1-Enabled, 2-Privileged

# Phone prompt/password for telnet/console session
phone_prompt: ""  ; Telnet/Console Prompt
phone_password: "cisco"  ; Telnet/Console 
Password


# Enable_VAD (1-enabled, 0-disabled)
enable_vad: "0"

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
user_info: none

# URL for external Directory location
directory_url: "http://10.12.41.1/directory.html;

[asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

[Turritopsis Dohrnii Teo En Ming]

Subject: HELP! I can't get my Cisco CP-7960G IP hardphone to register on 
my Asterisk VoIP IP PBX SIP Server with FreePBX GUI


Good day from Singapore,

My Asterisk version: 16.13.0
My FreePBX version: 15.0.16.81

On 7 December 2020, I was able to get Bria softphone to work with my 
Asterisk PBX server successfully.


On 19 December 2020, I bought a refurbished Cisco CP-7960G IP hardphone 
for SGD$30 in Singapore. I have tried all of the following steps, but I 
simply can't get my Cisco CP-7960G IP phone to register on my Asterisk 
PBX server.


TFTP works though. My DHCP server in my pfSense firewall applaince is 
able to assign my Cisco 7960 IP phone with an IP address with DHCP 
option 66 (TFTP server). My Cisco 7960 IP phone is able to connect to my 
TFTP server on my Asterisk PBX appliance and download firmware and 
configuration files successfully. However, it simply cannot register on 
my Asterisk PBX server.


You can watch my Youtube video at

https://www.youtube.com/watch?v=ip_F08jmmio

Appreciate your kind assistance.

Thank you very much.

STEPS I HAVE TRIED
===

Reference Guide: Configure Asterisk with Cisco IP Phones
Link: http://docshare02.docshare.tips/files/6706/67061980.pdf

SECTION: INSTALLING TFTP SERVER ON ASTERISK PBX APPLIANCE
=

Putty/ssh into Teo En Ming's Asterisk VoIP IP PBX SIP Server at 
192.168.1.9.


# yum install tftp-server

Package tftp-server-5.2-23.8.sng7.x86_64 already installed and latest 
version


# chkconfig xinetd on

# chkconfig tftp on

# systemctl start tftp.service

# ps -ef | grep tftp
root  3424 1  0 11:17 ?00:00:00 /usr/sbin/in.tftpd -s 
/tftpboot


SECTION: DOWNLOADING CISCO 7960 IP PHONE SIP FIRMWARE
==

# cd /tftpboot

# wget 
http://www.firewall.cx/downloads/cisco-tools-a-applications/cisco-ip-phone-a-ata-firmware-downloads/107-7940-a-7960-ip-phone-sccp-a-sip/file.html


# mv file.html file.zip

# unzip file.zip

# cd 7940_7960/

# cd SIP/

# tar -xf P0S3-8-12-00.tar

# rm P0S3-8-12-00.tar

# mv * /tftpboot/

# cd /tftpboot/

[root@freepbx tftpboot]# ls
7940_7960  file.zip  OS79XX.TXT  P003-8-12-00.bin  P003-8-12-00.sbn  
P0S3-8-12-00.loads  P0S3-8-12-00.sb2


SECTION: CREATING CISCO 7960 IP PHONE CONFIGURATION FILES
==

# nano OS79XX.TXT (Create configuration file)
=

P003-8-12-00

# nano XMLDefault.cnf.xml (Create configuration file)
=



 
 
 
 
 2000
 
 2427
 2428
 
 
 
 
 
 
 
P0S3-8-12-00
P0S3-8-12-00
SIP45.8-4-2S
SIP45.8-4-2S
SIP70.8-0-3S









# nano SIPDefault.cnf (Create configuration file)
=

image_version: "P0S3-8-12-00"
proxy1_address: "192.168.1.9"
# proxy2_address: "xxx.xxx.xxx.xxx"
# proxy3_address: "xxx.xxx.xxx.xxx"
# proxy4_address: "xxx.xxx.xxx.xxx"

# Proxy Server Port
proxy1_port:"5060"
# proxy2_port:"5060"
# proxy3_port:"5060"
# proxy4_port:"5060"
proxy_emergency: ""
proxy_emergency_port: "5060"
proxy_backup: ""
proxy_backup_port: "5060"
outbound_proxy: ""
outbound_proxy_port: "5060"

nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16348"
end_media_port: "20134"
nat_received_processing: "1"
dyn_dns_addr_1: ""
dyn_dns_addr_2: ""
dyn_tftp_addr: "192.168.1.9"
tftp_cfg_dir: "./"
proxy_register: "1"
timer_register_expires: "120"
preferred_codec: "none"
tos_media: "5"
enable_vad: "0"
dial_template: "dialplan"
network_media_type: "auto"
autocomplete: "1"
telnet_level: "2"
cnf_join_enable: "1"
semi_attended_transfer: "0"
call_waiting: "1"
anonymous_call_block: "0"
callerid_blocking: "0"
dnd_control: "0"
dtmf_inband: "1"
dtmf_outofband: "avt"
dtmf_db_level: "3"
dtmf_avt_payload: "101"
timer_t1: "500"
timer_t2: "4000"
sip_retx: "10"
sip_invite_retx: "6"
timer_invite_expires: "180"

sntp_mode: "directedbroadcast"
sntp_server: "time-a-g.nist.gov"
time_zone: "8"
time_format_24hr: "0"
dst_offset: "0"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "2"
dst_stop_month: "Nov"
dst_stop_day: "1"

dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: ""
dst_stop_time: "2"
dst_auto_adjust: "1"

messages_uri: "*99"
services_url: "http://example.domain.ext/services/menu.xml;
directory_url: "http://example.domain.ext/services/directory.php;
logo_url: "http://example.domain.ext/imagename.bmp;
http_proxy_addr: ""
http_proxy_port: ""
remote_party_id: 0

# nano dialplan.xml (Create configuration file)
===


  


# nano RINGLIST.DAT (Create configuration file)
===

FlintPhone FlintPhone.raw
HarpSynth HarpSynth.raw
Jamaica Jamaica.raw
Klaxons Klaxons.raw
KotoEffect KotoEffect.raw
MusicBox MusicBox.raw
Ohno Ohno.raw
Piano 1 Piano1.raw
Piano 

Re: [asterisk-users] Help missing

[Dovid Bender]

What is the application that you are missing?

On Sun, May 17, 2020 at 01:32 Saint Michael  wrote:

> I want to see the help when I type core show application , and it's
> not available. This is asterisk 16 from sources. I have libxml2-dev
> installed. Ubuntu 19
> What am I missing?
> Philip
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Help missing

[Saint Michael]

I want to see the help when I type core show application , and it's not
available. This is asterisk 16 from sources. I have libxml2-dev  installed.
Ubuntu 19
What am I missing?
Philip
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Help with FUNC_MATH

[Dovid Bender]

I found my mistake. I was running execif on the result. I needed to change:
ExecIf(${MATH(${HOUR_SELECTED}<11)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
TO:
ExecIf($["${MATH(${HOUR_SELECTED}<11)}" ==
"TRUE"]?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))



On Thu, Feb 13, 2020 at 2:12 PM Dovid Bender  wrote:

> John,
>
> That is correct. I am trying to figure out why Asterisk is executing the
> set part of the execif, if it's coming back as false.
>
>
>
> On Thu, Feb 13, 2020 at 2:10 PM John Kiniston 
> wrote:
>
>> My Apologies Dovid, I think I misunderstood your request.
>>
>> You don't have the time you need to convert in the format of date string,
>> Instead you have your users entering via DTMF when they want something to
>> happen?
>>
>> On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender 
>> wrote:
>>
>>> John,
>>>
>>> From looking at the wiki won't STRFIME just give me what I need based on
>>> the unix time that I put in? What I am actually looking to do is convert
>>> over from 12 hour format to 24 (unless strftime does just that and I don't
>>> kow what am I am doing?).
>>>
>>>
>>>
>>> On Thu, Feb 13, 2020 at 12:03 PM John Kiniston 
>>> wrote:
>>>
 Try using the STRFIME function instead of doing this by hand.

 https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME

 *%H*

 The hour as a decimal number using a 24-hour clock (range 00 to 23).

 *%I*

 The hour as a decimal number using a 12-hour clock (range 01 to 12).

 On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender 
 wrote:

> Hi,
>
> I have some dialplan code that is trying to convert 12 hour time with
> AM/PM to 24 hour format. The code has something like this:
> Exten =>
> 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>
> Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press
> option 2 they are selecting PM. If the time is from 1PM to 11PM then I 
> want
> to add 12 to the number (so if it's 1 make it 13 etc.). When I run the
> above the logs show the result as false yet if the user sets HOUR_SELECTED
> to 12 then after this line of dialplan code it gets switched to 24. What 
> am
> I doing wrong here?
>
> The exact DP code is:
> Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
>  same =>n,
> ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>  same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})
>
> And the output of the logs is:
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:1] NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK
> HOUR_SELECTED is 12") in new stack
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '12'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
> MATH(12<12) result is 'FALSE'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '12'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
> MATH(12+12,int) result is '24'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:2] ExecIf("SIP/204.145.219.31-81c6",
> "FALSE?Set(HOUR_SELECTED=24)") in new stack
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '24'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:3] NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK
> HOUR_SELECTED IS 24") in new stack
>
>
> TIA.
>
> Dovid
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



 --
 A human being should be able to change a diaper, plan an invasion,
 butcher a hog, conn a ship, design a building, write a sonnet, balance
 accounts, build a wall, set a bone, comfort the dying, take orders, give
 orders, cooperate, act alone, solve equations, analyze a new problem, pitch
 manure, program a computer, cook a tasty meal, fight efficiently, die
 gallantly. Specialization is for insects.
 ---Heinlein
 --
 

Re: [asterisk-users] Help with FUNC_MATH

[Don Kelly]

Do you know that it is coming back as FALSE, or are you assuming that from 
examining the expression?

 

  --Don

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Thursday, February 13, 2020 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with FUNC_MATH

 

John,

 

That is correct. I am trying to figure out why Asterisk is executing the set 
part of the execif, if it's coming back as false.

 

 

 

On Thu, Feb 13, 2020 at 2:10 PM John Kiniston  wrote:

My Apologies Dovid, I think I misunderstood your request.

You don't have the time you need to convert in the format of date string, 
Instead you have your users entering via DTMF when they want something to 
happen?

 

On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender  wrote:

John,

 

>From looking at the wiki won't STRFIME just give me what I need based on the 
>unix time that I put in? What I am actually looking to do is convert over from 
>12 hour format to 24 (unless strftime does just that and I don't kow what am I 
>am doing?).

 

 

 

On Thu, Feb 13, 2020 at 12:03 PM John Kiniston  wrote:

Try using the STRFIME function instead of doing this by hand.

https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME

%H 

The hour as a decimal number using a 24-hour clock (range 00 to 23). 

%I 

The hour as a decimal number using a 12-hour clock (range 01 to 12). 

 

On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender  wrote:

Hi,

 

I have some dialplan code that is trying to convert 12 hour time with AM/PM to 
24 hour format. The code has something like this:
Exten => 
2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))

 

Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press option 2 
they are selecting PM. If the time is from 1PM to 11PM then I want to add 12 to 
the number (so if it's 1 make it 13 etc.). When I run the above the logs show 
the result as false yet if the user sets HOUR_SELECTED to 12 then after this 
line of dialplan code it gets switched to 24. What am I doing wrong here?

 

The exact DP code is:

Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
 same =>n, 
ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
 same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})

 

And the output of the logs is:

[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:1] 
NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK HOUR_SELECTED is 12") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function MATH(12<12) 
result is 'FALSE'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function 
MATH(12+12,int) result is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:2] 
ExecIf("SIP/204.145.219.31-81c6", "FALSE?Set(HOUR_SELECTED=24)") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:3] 
NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK HOUR_SELECTED IS 24") in new 
stack

 

 

TIA.

 

Dovid

 

-- 
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-- 

A human being should be able to change a diaper, plan an invasion, butcher a 
hog, conn a ship, design a building, write a sonnet, balance accounts, build a 
wall, set a bone, comfort the dying, take orders, give orders, cooperate, act 
alone, solve equations, analyze a new problem, pitch manure, program a 
computer, cook a tasty meal, fight efficiently, die gallantly. Specialization 
is for insects.
---Heinlein

-- 
_
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asterisk-users mailing list
To UN

Re: [asterisk-users] Help with FUNC_MATH

[Dovid Bender]

John,

That is correct. I am trying to figure out why Asterisk is executing the
set part of the execif, if it's coming back as false.



On Thu, Feb 13, 2020 at 2:10 PM John Kiniston 
wrote:

> My Apologies Dovid, I think I misunderstood your request.
>
> You don't have the time you need to convert in the format of date string,
> Instead you have your users entering via DTMF when they want something to
> happen?
>
> On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender  wrote:
>
>> John,
>>
>> From looking at the wiki won't STRFIME just give me what I need based on
>> the unix time that I put in? What I am actually looking to do is convert
>> over from 12 hour format to 24 (unless strftime does just that and I don't
>> kow what am I am doing?).
>>
>>
>>
>> On Thu, Feb 13, 2020 at 12:03 PM John Kiniston 
>> wrote:
>>
>>> Try using the STRFIME function instead of doing this by hand.
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME
>>>
>>> *%H*
>>>
>>> The hour as a decimal number using a 24-hour clock (range 00 to 23).
>>>
>>> *%I*
>>>
>>> The hour as a decimal number using a 12-hour clock (range 01 to 12).
>>>
>>> On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender 
>>> wrote:
>>>
 Hi,

 I have some dialplan code that is trying to convert 12 hour time with
 AM/PM to 24 hour format. The code has something like this:
 Exten =>
 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))

 Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press
 option 2 they are selecting PM. If the time is from 1PM to 11PM then I want
 to add 12 to the number (so if it's 1 make it 13 etc.). When I run the
 above the logs show the result as false yet if the user sets HOUR_SELECTED
 to 12 then after this line of dialplan code it gets switched to 24. What am
 I doing wrong here?

 The exact DP code is:
 Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
  same =>n,
 ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
  same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})

 And the output of the logs is:
 [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
 [2@am_pm_select:1] NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK
 HOUR_SELECTED is 12") in new stack
 [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
 'HOUR_SELECTED' is '12'
 [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
 MATH(12<12) result is 'FALSE'
 [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
 'HOUR_SELECTED' is '12'
 [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
 MATH(12+12,int) result is '24'
 [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
 [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
 [2@am_pm_select:2] ExecIf("SIP/204.145.219.31-81c6",
 "FALSE?Set(HOUR_SELECTED=24)") in new stack
 [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
 'HOUR_SELECTED' is '24'
 [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
 [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
 [2@am_pm_select:3] NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK
 HOUR_SELECTED IS 24") in new stack


 TIA.

 Dovid

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 Check out the new Asterisk community forum at:
 https://community.asterisk.org/

 New to Asterisk? Start here:
   https://wiki.asterisk.org/wiki/display/AST/Getting+Started

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>> --
>>> A human being should be able to change a diaper, plan an invasion,
>>> butcher a hog, conn a ship, design a building, write a sonnet, balance
>>> accounts, build a wall, set a bone, comfort the dying, take orders, give
>>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch
>>> manure, program a computer, cook a tasty meal, fight efficiently, die
>>> gallantly. Specialization is for insects.
>>> ---Heinlein
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> 

Re: [asterisk-users] Help with FUNC_MATH

[Don Kelly]

Is it stored as a floating-point number or an integer? Floating-point decimal 
numbers are often not stored precisely. As in my example, below, “12” may be 
stored as 11.999 (simplified) and, although it will be treated  as “12” 
in most cases, it will appear to be less than 12 in a comparison.

 

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Thursday, February 13, 2020 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with FUNC_MATH

 

HOUR_SELECTED is going to be 1-12

 

 

On Thu, Feb 13, 2020 at 2:05 PM Don Kelly  wrote:

Is HOUR_SELECTED a floating-point number (e.g. 11.999)? If so, you need 
to account for that in your comparison.

 

  --Don

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Thursday, February 13, 2020 4:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help with FUNC_MATH

 

Hi,

 

I have some dialplan code that is trying to convert 12 hour time with AM/PM to 
24 hour format. The code has something like this:
Exten => 
2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))

 

Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press option 2 
they are selecting PM. If the time is from 1PM to 11PM then I want to add 12 to 
the number (so if it's 1 make it 13 etc.). When I run the above the logs show 
the result as false yet if the user sets HOUR_SELECTED to 12 then after this 
line of dialplan code it gets switched to 24. What am I doing wrong here?

 

The exact DP code is:

Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
 same =>n, 
ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
 same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})

 

And the output of the logs is:

[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:1] 
NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK HOUR_SELECTED is 12") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function MATH(12<12) 
result is 'FALSE'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function 
MATH(12+12,int) result is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:2] 
ExecIf("SIP/204.145.219.31-81c6", "FALSE?Set(HOUR_SELECTED=24)") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:3] 
NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK HOUR_SELECTED IS 24") in new 
stack

 

 

TIA.

 

Dovid

 

-- 
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Re: [asterisk-users] Help with FUNC_MATH

[John Kiniston]

My Apologies Dovid, I think I misunderstood your request.

You don't have the time you need to convert in the format of date string,
Instead you have your users entering via DTMF when they want something to
happen?

On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender  wrote:

> John,
>
> From looking at the wiki won't STRFIME just give me what I need based on
> the unix time that I put in? What I am actually looking to do is convert
> over from 12 hour format to 24 (unless strftime does just that and I don't
> kow what am I am doing?).
>
>
>
> On Thu, Feb 13, 2020 at 12:03 PM John Kiniston 
> wrote:
>
>> Try using the STRFIME function instead of doing this by hand.
>>
>> https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME
>>
>> *%H*
>>
>> The hour as a decimal number using a 24-hour clock (range 00 to 23).
>>
>> *%I*
>>
>> The hour as a decimal number using a 12-hour clock (range 01 to 12).
>>
>> On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender  wrote:
>>
>>> Hi,
>>>
>>> I have some dialplan code that is trying to convert 12 hour time with
>>> AM/PM to 24 hour format. The code has something like this:
>>> Exten =>
>>> 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>>>
>>> Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press
>>> option 2 they are selecting PM. If the time is from 1PM to 11PM then I want
>>> to add 12 to the number (so if it's 1 make it 13 etc.). When I run the
>>> above the logs show the result as false yet if the user sets HOUR_SELECTED
>>> to 12 then after this line of dialplan code it gets switched to 24. What am
>>> I doing wrong here?
>>>
>>> The exact DP code is:
>>> Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
>>>  same =>n,
>>> ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>>>  same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})
>>>
>>> And the output of the logs is:
>>> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
>>> [2@am_pm_select:1] NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK
>>> HOUR_SELECTED is 12") in new stack
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
>>> 'HOUR_SELECTED' is '12'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
>>> MATH(12<12) result is 'FALSE'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
>>> 'HOUR_SELECTED' is '12'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
>>> MATH(12+12,int) result is '24'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
>>> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
>>> [2@am_pm_select:2] ExecIf("SIP/204.145.219.31-81c6",
>>> "FALSE?Set(HOUR_SELECTED=24)") in new stack
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
>>> 'HOUR_SELECTED' is '24'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
>>> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
>>> [2@am_pm_select:3] NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK
>>> HOUR_SELECTED IS 24") in new stack
>>>
>>>
>>> TIA.
>>>
>>> Dovid
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>> A human being should be able to change a diaper, plan an invasion,
>> butcher a hog, conn a ship, design a building, write a sonnet, balance
>> accounts, build a wall, set a bone, comfort the dying, take orders, give
>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch
>> manure, program a computer, cook a tasty meal, fight efficiently, die
>> gallantly. Specialization is for insects.
>> ---Heinlein
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To 

Re: [asterisk-users] Help with FUNC_MATH

[Dovid Bender]

HOUR_SELECTED is going to be 1-12


On Thu, Feb 13, 2020 at 2:05 PM Don Kelly  wrote:

> Is HOUR_SELECTED a floating-point number (e.g. 11.999)? If so, you
> need to account for that in your comparison.
>
>
>
>   --Don
>
>
>
>
>
> *From:* asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On
> Behalf Of *Dovid Bender
> *Sent:* Thursday, February 13, 2020 4:47 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Help with FUNC_MATH
>
>
>
> Hi,
>
>
>
> I have some dialplan code that is trying to convert 12 hour time with
> AM/PM to 24 hour format. The code has something like this:
> Exten =>
> 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>
>
>
> Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press
> option 2 they are selecting PM. If the time is from 1PM to 11PM then I want
> to add 12 to the number (so if it's 1 make it 13 etc.). When I run the
> above the logs show the result as false yet if the user sets HOUR_SELECTED
> to 12 then after this line of dialplan code it gets switched to 24. What am
> I doing wrong here?
>
>
>
> The exact DP code is:
>
> Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
>  same =>n,
> ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>  same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})
>
>
>
> And the output of the logs is:
>
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:1] NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK
> HOUR_SELECTED is 12") in new stack
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '12'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
> MATH(12<12) result is 'FALSE'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '12'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
> MATH(12+12,int) result is '24'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:2] ExecIf("SIP/204.145.219.31-81c6",
> "FALSE?Set(HOUR_SELECTED=24)") in new stack
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '24'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:3] NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK
> HOUR_SELECTED IS 24") in new stack
>
>
>
>
>
> TIA.
>
>
>
> Dovid
>
>
> --
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Re: [asterisk-users] Help with FUNC_MATH

[Don Kelly]

Is HOUR_SELECTED a floating-point number (e.g. 11.999)? If so, you need 
to account for that in your comparison.

 

  --Don

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Thursday, February 13, 2020 4:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help with FUNC_MATH

 

Hi,

 

I have some dialplan code that is trying to convert 12 hour time with AM/PM to 
24 hour format. The code has something like this:
Exten => 
2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))

 

Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press option 2 
they are selecting PM. If the time is from 1PM to 11PM then I want to add 12 to 
the number (so if it's 1 make it 13 etc.). When I run the above the logs show 
the result as false yet if the user sets HOUR_SELECTED to 12 then after this 
line of dialplan code it gets switched to 24. What am I doing wrong here?

 

The exact DP code is:

Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
 same =>n, 
ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
 same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})

 

And the output of the logs is:

[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:1] 
NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK HOUR_SELECTED is 12") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function MATH(12<12) 
result is 'FALSE'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function 
MATH(12+12,int) result is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:2] 
ExecIf("SIP/204.145.219.31-81c6", "FALSE?Set(HOUR_SELECTED=24)") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:3] 
NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK HOUR_SELECTED IS 24") in new 
stack

 

 

TIA.

 

Dovid

 

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Re: [asterisk-users] Help with FUNC_MATH

[Dovid Bender]

John,

>From looking at the wiki won't STRFIME just give me what I need based on
the unix time that I put in? What I am actually looking to do is convert
over from 12 hour format to 24 (unless strftime does just that and I don't
kow what am I am doing?).



On Thu, Feb 13, 2020 at 12:03 PM John Kiniston 
wrote:

> Try using the STRFIME function instead of doing this by hand.
>
> https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME
>
> *%H*
>
> The hour as a decimal number using a 24-hour clock (range 00 to 23).
>
> *%I*
>
> The hour as a decimal number using a 12-hour clock (range 01 to 12).
>
> On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender  wrote:
>
>> Hi,
>>
>> I have some dialplan code that is trying to convert 12 hour time with
>> AM/PM to 24 hour format. The code has something like this:
>> Exten =>
>> 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>>
>> Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press
>> option 2 they are selecting PM. If the time is from 1PM to 11PM then I want
>> to add 12 to the number (so if it's 1 make it 13 etc.). When I run the
>> above the logs show the result as false yet if the user sets HOUR_SELECTED
>> to 12 then after this line of dialplan code it gets switched to 24. What am
>> I doing wrong here?
>>
>> The exact DP code is:
>> Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
>>  same =>n,
>> ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>>  same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})
>>
>> And the output of the logs is:
>> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
>> [2@am_pm_select:1] NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK
>> HOUR_SELECTED is 12") in new stack
>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
>> 'HOUR_SELECTED' is '12'
>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
>> MATH(12<12) result is 'FALSE'
>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
>> 'HOUR_SELECTED' is '12'
>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
>> MATH(12+12,int) result is '24'
>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
>> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
>> [2@am_pm_select:2] ExecIf("SIP/204.145.219.31-81c6",
>> "FALSE?Set(HOUR_SELECTED=24)") in new stack
>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
>> 'HOUR_SELECTED' is '24'
>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
>> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
>> [2@am_pm_select:3] NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK
>> HOUR_SELECTED IS 24") in new stack
>>
>>
>> TIA.
>>
>> Dovid
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
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Re: [asterisk-users] Help with FUNC_MATH

[John Kiniston]

Try using the STRFIME function instead of doing this by hand.

https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME

*%H*

The hour as a decimal number using a 24-hour clock (range 00 to 23).

*%I*

The hour as a decimal number using a 12-hour clock (range 01 to 12).

On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender  wrote:

> Hi,
>
> I have some dialplan code that is trying to convert 12 hour time with
> AM/PM to 24 hour format. The code has something like this:
> Exten =>
> 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>
> Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press
> option 2 they are selecting PM. If the time is from 1PM to 11PM then I want
> to add 12 to the number (so if it's 1 make it 13 etc.). When I run the
> above the logs show the result as false yet if the user sets HOUR_SELECTED
> to 12 then after this line of dialplan code it gets switched to 24. What am
> I doing wrong here?
>
> The exact DP code is:
> Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
>  same =>n,
> ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>  same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})
>
> And the output of the logs is:
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:1] NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK
> HOUR_SELECTED is 12") in new stack
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '12'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
> MATH(12<12) result is 'FALSE'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '12'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
> MATH(12+12,int) result is '24'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:2] ExecIf("SIP/204.145.219.31-81c6",
> "FALSE?Set(HOUR_SELECTED=24)") in new stack
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '24'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:3] NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK
> HOUR_SELECTED IS 24") in new stack
>
>
> TIA.
>
> Dovid
>
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>
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a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
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Specialization is for insects.
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[asterisk-users] Help with FUNC_MATH

[Dovid Bender]

Hi,

I have some dialplan code that is trying to convert 12 hour time with AM/PM
to 24 hour format. The code has something like this:
Exten =>
2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))

Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press
option 2 they are selecting PM. If the time is from 1PM to 11PM then I want
to add 12 to the number (so if it's 1 make it 13 etc.). When I run the
above the logs show the result as false yet if the user sets HOUR_SELECTED
to 12 then after this line of dialplan code it gets switched to 24. What am
I doing wrong here?

The exact DP code is:
Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
 same =>n,
ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
 same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})

And the output of the logs is:
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:1]
NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK HOUR_SELECTED is 12") in
new stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
MATH(12<12) result is 'FALSE'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
MATH(12+12,int) result is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:2]
ExecIf("SIP/204.145.219.31-81c6", "FALSE?Set(HOUR_SELECTED=24)") in new
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
'HOUR_SELECTED' is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:3]
NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK HOUR_SELECTED IS 24") in
new stack


TIA.

Dovid
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[asterisk-users] Help migrating voicemail to database

[mdiehl]

I am in the process of configuring my systems to store voicemail in a mysql 
databse as opposed to on the filesystem, as it is now.

My backup server is currently configured for db storage, while my production 
server is still using the filesystem during testing.

When I record a vm message on my backup server, I am able to retrieve it, hear 
it, and delete it as expected.  So, I believe my backup server is properly 
configured.

However, when I move to my production server and run a script, that I wrote, to 
import the vm from the file system to the database, I have problems.  I'm able 
to access the message from the voicemail system.  However, when I go to listen 
to it, I get silence and the Asterisk server stops responding to new calls; I 
have to restart asterisk.

The console log looks like:
[Jun 14 07:30:58] --  Playing 
'vm-youhave.ulaw' (language 'en')
[Jun 14 07:30:59] --  Playing 'digits/1.ulaw' 
(language 'en')
[Jun 14 07:31:00] --  Playing 'vm-INBOX.ulaw' 
(language 'en')
[Jun 14 07:31:01] --  Playing 'vm-and.ulaw' 
(language 'en')
[Jun 14 07:31:01] --  Playing 'vm-first.ulaw' 
(language 'en')
voip11*CLI>
voip11*CLI>

The console remains responsive, but I'm unable to make any for test calls w/o a 
restart.

The database record that results from my import script looks just like a 
comparable one created by recording a message from my phone, so I don't THINK 
it's a data issue.

Also, I've been able to verify that the database does contain an actual .wav 
file with the correct format.

Has anyone seen this before?

Any suggestions would be welcome.

Mike Diehl.



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Re: [asterisk-users] Help me please i am facing much trouble

[Steve Edwards]

On Sat, 16 Jan 2016, waqas.mehmood90 wrote:

How to get user extension number in agi php scrip from which he's 
calling on ivr i am using cid and able to get his name but not his 
extension no please help me thanx in advance


You can use the 'agi set debug on' CLI command to enable AGI debugging. 
This will show you the AGI variables passed to your script.


You can use the 'dumpchan()' application to display the available channel 
variables.


There are also CLI commands to interrogate the Asterisk database.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST

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[asterisk-users] Help me please i am facing much trouble

[waqas.mehmood90]

How to get user extension number in agi php scrip from which he's calling on 
ivr i am using cid and able to get his name but not his extension no please 
help me thanx in advance



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Re: [asterisk-users] Help with CDR-Stats

[Vitor Mazuco]

Right, thanks for your reply!



2015-12-16 14:45 GMT-02:00, Bruce Ferrell :
> billing is sending invoices for calls to customers.
>
> reporting is overall statistics on the aggregate of your calls...
> Average call hold time, common (or uncommon destinations) etc.  If you
> see a destination that suddenly has a lot of calls with hold time below
> normal, there may be a call quality problem.
>
> TPC (the phone company) has used statistical troubleshooting techniques
> for decades to keep quality up so customers don't have to complain, not
> to mention for sizing.
>
>
>
> On 12/16/15 8:23 AM, Vitor Mazuco wrote:
>>
>> Humm whats is the diferent?
>>
>> Em 16/12/2015 14:19, "Annus Fictus" > > escreveu:
>>
>> CDR-STATS is for reporting.
>>
>> A2Billing is for billing...
>>
>> Regards
>>
>> El 16/12/2015 a las 11:15, Vitor Mazuco escribió:
>>
>> Hi everyone!
>>
>> I'm trying to install CDR-Stats (cdr-stats.org
>> ), but it very difficult.
>>
>> Is there others optins for billing?
>>
>> Thanks
>>
>>
>>
>> --
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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>> http://www.asterisk.org/hello
>>
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>
>

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[asterisk-users] Help with CDR-Stats

[Vitor Mazuco]

Hi everyone!

I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult.

Is there others optins for billing?

Thanks

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Re: [asterisk-users] Help with CDR-Stats

[Annus Fictus]

CDR-STATS is for reporting.

A2Billing is for billing...

Regards

El 16/12/2015 a las 11:15, Vitor Mazuco escribió:

Hi everyone!

I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult.

Is there others optins for billing?

Thanks




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Re: [asterisk-users] Help with CDR-Stats

[Vitor Mazuco]

Humm whats is the diferent?
Em 16/12/2015 14:19, "Annus Fictus"  escreveu:

> CDR-STATS is for reporting.
>
> A2Billing is for billing...
>
> Regards
>
> El 16/12/2015 a las 11:15, Vitor Mazuco escribió:
>
>> Hi everyone!
>>
>> I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult.
>>
>> Is there others optins for billing?
>>
>> Thanks
>>
>>
>
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Re: [asterisk-users] Help with CDR-Stats

[Bruce Ferrell]

billing is sending invoices for calls to customers.

reporting is overall statistics on the aggregate of your calls... 
Average call hold time, common (or uncommon destinations) etc.  If you 
see a destination that suddenly has a lot of calls with hold time below 
normal, there may be a call quality problem.


TPC (the phone company) has used statistical troubleshooting techniques 
for decades to keep quality up so customers don't have to complain, not 
to mention for sizing.




On 12/16/15 8:23 AM, Vitor Mazuco wrote:


Humm whats is the diferent?

Em 16/12/2015 14:19, "Annus Fictus" > escreveu:


CDR-STATS is for reporting.

A2Billing is for billing...

Regards

El 16/12/2015 a las 11:15, Vitor Mazuco escribió:

Hi everyone!

I'm trying to install CDR-Stats (cdr-stats.org
), but it very difficult.

Is there others optins for billing?

Thanks



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[asterisk-users] Help - Asterisk SIP Messages Parameter Modification

[WALEED AHMED KHAN]

Dear All,

I have a query.

I want to know if there is any possiblity to modify SIP Messages Parameters
using the asterisk CLI mode.

I want to change the parameters for e.g in INVITE  message. How it can be
done in asterisk.

Kindly assist me.

Regards,

*Waleed A. Khan*
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[asterisk-users] Help with voicemail

[Luca Bertoncello]

Hi list!

My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a
voicemail.
On two of these numbers the voicemail works without any problem, on the other
it doesn't...
I get this error:

[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a 
codec translation path from 0x100 (g729) to 0x2 (gsm)
[Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open 
/var/spool/asterisk/voicemail/default/0039015111/unavail (format 0x100 
(g729)): No such file or directory
[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a 
codec translation path from 0x100 (g729) to 0x2 (gsm)
[Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open 
beep (format 0x100 (g729)): No such file or directory
-- Recording the message
-- x=0, open writing:  
/var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format: wav, 
0x6edbd8
-- x=1, open writing:  
/var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format: gsm, 
0x7c6978
[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a 
codec translation path from 0x100 (g729) to 0x40 (slin)

Of course, I have a
file /var/spool/asterisk/voicemail/default/0039015111/unavail.gsm...

Can someone help me to solve my problem?

Thanks a lot!
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Help with voicemail

[Luca Bertoncello]

Matthew Jordan  schrieb:

> Do you have a g729 codec module loaded? If so, does it show a

Bingo!

> translation path between g729 and gsm? If so, do you have sufficient
> encoder/decoder licenses?

I don't have a translation path between g729 and gsm...
Since I don't have a g729 codec, I changed the properties of this peer
enabling other codecs.
Now the voicemail works as expected...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Help with voicemail

[Matthew Jordan]

On Sat, Oct 17, 2015 at 10:12 AM, Luca Bertoncello  wrote:
> Hi list!
>
> My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a
> voicemail.
> On two of these numbers the voicemail works without any problem, on the other
> it doesn't...
> I get this error:
>
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a 
> codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open 
> /var/spool/asterisk/voicemail/default/0039015111/unavail (format 0x100 
> (g729)): No such file or directory
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a 
> codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open 
> beep (format 0x100 (g729)): No such file or directory
> -- Recording the message
> -- x=0, open writing:  
> /var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format: wav, 
> 0x6edbd8
> -- x=1, open writing:  
> /var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format: gsm, 
> 0x7c6978
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a 
> codec translation path from 0x100 (g729) to 0x40 (slin)
>
> Of course, I have a
> file /var/spool/asterisk/voicemail/default/0039015111/unavail.gsm...
>
> Can someone help me to solve my problem?
>

Do you have a g729 codec module loaded? If so, does it show a
translation path between g729 and gsm? If so, do you have sufficient
encoder/decoder licenses?

Matt

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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Help with voicemail

[Sam Basan]

Check your phone codecs.
It set to g729 while you don't have this codec in your asterisk nor files
in this codec.
בתאריך 17 באוק' 2015 18:34,‏ "Luca Bertoncello"  כתב:

> Hi list!
>
> My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a
> voicemail.
> On two of these numbers the voicemail works without any problem, on the
> other
> it doesn't...
> I get this error:
>
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to
> find a codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to
> open /var/spool/asterisk/voicemail/default/0039015111/unavail (format
> 0x100 (g729)): No such file or directory
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to
> find a codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to
> open beep (format 0x100 (g729)): No such file or directory
> -- Recording the message
> -- x=0, open writing:
> /var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format:
> wav, 0x6edbd8
> -- x=1, open writing:
> /var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format:
> gsm, 0x7c6978
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to
> find a codec translation path from 0x100 (g729) to 0x40 (slin)
>
> Of course, I have a
> file /var/spool/asterisk/voicemail/default/0039015111/unavail.gsm...
>
> Can someone help me to solve my problem?
>
> Thanks a lot!
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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> To UNSUBSCRIBE or update options visit:
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>
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Re: [asterisk-users] Help With Physical Layer

[ludwingn]

  Try to check CRC4, both hace to be the sameEnviadodesdemismartphoneBlackBerry10.De: Tony KasuleEnviado: miércoles 1 de julio de 2015 01:11 a.m.Para: Duncan Turnbull; Asterisk Users Mailing List - Non-Commercial DiscussionResponder a: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [asterisk-users] Help With Physical LayerOn Tue, Jun 30, 2015 at 2:15 PM, Duncan Turnbull dun...@e-simple.co.nz wrote:






Hi Tony

I'm not familiar with the card you but 120 ohm is usually twisted pair, and 75 ohm is coax (usually). If it is changeable its usually done with jumpers on the card. The new Digium cards have no jumpers anymore and I don't think they support coaxial cables. The only changes we can make on the digium cards is to switch between E1/T1 modes and that's done when loading it in the kernel using modprobe.I have used RAD modems before and their cable will have a specific pinoutthat you need to match. I doubt your digium card by default matches the pins, but maybe, and perhaps maybe your alcatel did. There will be a tx pair and an rx pair and you need to make sure they connect to the rx and tx pairs on the digium preferably with the same polarity. If you are getting no signal I would think its that. You can use a multimeter to check for voltage on the pins. 
I would doubt it's the pinout but it's possible. I have tried several cables (cross-over, straight-through, E1) and both didn't work but surprisingly, they do on the other RAD modem that comes to the same PBX.
Once you get that sortedanother catch maybe timing. You will need to take E1 timing from one or other of the Telco's, with one as the primary source.If they for some reason aren't synced you will get errors every so oftenThanks for the pointers.I am going to keep trying several things and if I get a break through, I'll definitely share the results.For no, any more help is welcome.thanks!


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Re: [asterisk-users] Help With Physical Layer

[Tony Kasule]

Dear David,

I am sorry, I can give the answer right now as the box is at a remote
location where I don't have access right now. However, I think the card is
ok and al the spans are working. Yesterday I had asked the telco to bring a
new RAD modem and I also took there another dell optiplex 3010 desktop with
another TE235 card in it but same results! I can confirm that the cards are
ok!


On Tue, Jun 30, 2015 at 12:36 PM, David Duffett dduff...@digium.com wrote:

 What response do you get to *CLI pri show spans ?

 On 30 June 2015 at 09:34, Tony Kasule timotsm...@gmail.com wrote:

 Hello,

 Anyone to help me with this issue? It has never worked :(

 On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com
 wrote:

 Hello users,

 I have a Digium Te235 and asterisk 13  which have worked well with 1
 carrier but we have failed to add a 2nd carrier. The second telco brings
 their E1 line over finer, terminated in a RAD modem and they give me
 ethernet to the E1 card. It's the first time i am having install such a
 solution, which ideally would be not a big problem.

 However, The  physical layer has failed to come up! I have tried the
 same setup in an Alcatel OmniOCX and it works well. I can confirm that the
 port is also well configured because when I interchange the cables (with
 the exiting cable from the other telco), the alarm clears emmediately for
 the 1st telco and becomes RED for the 2nd telco. A loop also clears the
 alarm on both ports.

 The telco has told me to make sure that Line Impedance is 120 ohms but
 There's no where to set that and when I was reading, I was told that E1 is
 already 120 ohms so no need to change anything.

 Has anyone here experienced this?  What other things can I try?

 Thank you!

 Regards,
 Tim



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 *David Duffett*
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 6 Landscape Close · Weston on the Green · Bicester · Oxfordshire OX25 3SX
  · UK
 direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
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 http://www.asterisk.org/

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Re: [asterisk-users] Help With Physical Layer

[Tony Kasule]

Hi Dale,

Yes, I tried a cross-over cable, I also tried terminated a new E1- cable
with only PINS 1-2 and 4-5 but still no luck. Everything I tried, I would
replicate with the other telco's setup  and results would be positive.

I have a feeling this new telco brought a new model of a RAD modem that
somehow doesn't work with our cards but am struggling to find the technical
reasons (and possible fixes) to support that hypothesis.

On Tue, Jun 30, 2015 at 1:43 PM, Dale Noll dn...@wi.rr.com wrote:


 On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule timotsm...@gmail.com wrote:

 Hello,

 Anyone to help me with this issue? It has never worked :(

 On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com
 wrote:

 Hello users,

 I have a Digium Te235 and asterisk 13  which have worked well with 1
 carrier but we have failed to add a 2nd carrier. The second telco brings
 their E1 line over finer, terminated in a RAD modem and they give me
 ethernet to the E1 card. It's the first time i am having install such a
 solution, which ideally would be not a big problem.

 However, The  physical layer has failed to come up! I have tried the
 same setup in an Alcatel OmniOCX and it works well. I can confirm that the
 port is also well configured because when I interchange the cables (with
 the exiting cable from the other telco), the alarm clears emmediately for
 the 1st telco and becomes RED for the 2nd telco. A loop also clears the
 alarm on both ports.

 The telco has told me to make sure that Line Impedance is 120 ohms but
 There's no where to set that and when I was reading, I was told that E1 is
 already 120 ohms so no need to change anything.

 Has anyone here experienced this?  What other things can I try?

 Thank you!

 Regards,
 Tim



 Have you tried a E-1 Cross Over cable from your RAD modem to the Digium
 card?

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Re: [asterisk-users] Help With Physical Layer

[Tony Kasule]

On Tue, Jun 30, 2015 at 2:15 PM, Duncan Turnbull dun...@e-simple.co.nz
wrote:


 Hi Tony

 I'm not familiar with the card you but 120 ohm is usually twisted pair,
 and 75 ohm is coax (usually). If it is changeable its usually done with
 jumpers on the card.


The new Digium cards have no jumpers anymore and I don't think they support
coaxial cables. The only changes we can make on the digium cards is to
switch between E1/T1 modes and that's done when loading it in the kernel
using modprobe.


 I have used RAD modems before and their cable will have a specific
 pinout that you need to match. I doubt your digium card by default matches
 the pins, but maybe, and perhaps maybe your alcatel did. There will be a tx
 pair and an rx pair and you need to make sure they connect to the rx and tx
 pairs on the digium preferably with the same polarity. If you are getting
 no signal I would think its that. You can use a multimeter to check for
 voltage on the pins.


I would doubt it's the pinout but it's possible. I have tried several
cables (cross-over, straight-through, E1) and both didn't work but
surprisingly, they do on the other RAD modem that comes to the same PBX.



 Once you get that sorted another catch maybe timing. You will need to take
 E1 timing from one or other of the Telco's, with one as the primary
 source. If they for some reason aren't synced you will get errors every so
 often


Thanks for the pointers.

I am going to keep trying several things and if I get a break through, I'll
definitely share the results.

For no, any more help is welcome.

thanks!
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Re: [asterisk-users] Help With Physical Layer

[Tony Kasule]

Hello,

Anyone to help me with this issue? It has never worked :(

On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com wrote:

 Hello users,

 I have a Digium Te235 and asterisk 13  which have worked well with 1
 carrier but we have failed to add a 2nd carrier. The second telco brings
 their E1 line over finer, terminated in a RAD modem and they give me
 ethernet to the E1 card. It's the first time i am having install such a
 solution, which ideally would be not a big problem.

 However, The  physical layer has failed to come up! I have tried the same
 setup in an Alcatel OmniOCX and it works well. I can confirm that the port
 is also well configured because when I interchange the cables (with the
 exiting cable from the other telco), the alarm clears emmediately for the
 1st telco and becomes RED for the 2nd telco. A loop also clears the alarm
 on both ports.

 The telco has told me to make sure that Line Impedance is 120 ohms but
 There's no where to set that and when I was reading, I was told that E1 is
 already 120 ohms so no need to change anything.

 Has anyone here experienced this?  What other things can I try?

 Thank you!

 Regards,
 Tim

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Re: [asterisk-users] Help With Physical Layer (Tony Kasule)

[Mc GRATH Ricardo]

Tim 

At first should take a look to cable pinout (RAD documents) as  pin 1,2, 
Transmit (output) and 4, 5  Receive (input) for Digium card  you should use a 
straight cable (try to test with new cable one too).
Second check Dahdi configuration parameters, use dahdi commands as; dahdi show 
status, service dahdi restart and check  result, (could a mistake on parameter 
value on system.conf).
 
Mc GRATH Ricardo
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Re: [asterisk-users] Help With Physical Layer

[Dale Noll]

On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule timotsm...@gmail.com wrote:

 Hello,

 Anyone to help me with this issue? It has never worked :(

 On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com
 wrote:

 Hello users,

 I have a Digium Te235 and asterisk 13  which have worked well with 1
 carrier but we have failed to add a 2nd carrier. The second telco brings
 their E1 line over finer, terminated in a RAD modem and they give me
 ethernet to the E1 card. It's the first time i am having install such a
 solution, which ideally would be not a big problem.

 However, The  physical layer has failed to come up! I have tried the same
 setup in an Alcatel OmniOCX and it works well. I can confirm that the port
 is also well configured because when I interchange the cables (with the
 exiting cable from the other telco), the alarm clears emmediately for the
 1st telco and becomes RED for the 2nd telco. A loop also clears the alarm
 on both ports.

 The telco has told me to make sure that Line Impedance is 120 ohms but
 There's no where to set that and when I was reading, I was told that E1 is
 already 120 ohms so no need to change anything.

 Has anyone here experienced this?  What other things can I try?

 Thank you!

 Regards,
 Tim



Have you tried a E-1 Cross Over cable from your RAD modem to the Digium
card?
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Re: [asterisk-users] Help With Physical Layer

[David Duffett]

What response do you get to *CLI pri show spans ?

On 30 June 2015 at 09:34, Tony Kasule timotsm...@gmail.com wrote:

 Hello,

 Anyone to help me with this issue? It has never worked :(

 On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com
 wrote:

 Hello users,

 I have a Digium Te235 and asterisk 13  which have worked well with 1
 carrier but we have failed to add a 2nd carrier. The second telco brings
 their E1 line over finer, terminated in a RAD modem and they give me
 ethernet to the E1 card. It's the first time i am having install such a
 solution, which ideally would be not a big problem.

 However, The  physical layer has failed to come up! I have tried the same
 setup in an Alcatel OmniOCX and it works well. I can confirm that the port
 is also well configured because when I interchange the cables (with the
 exiting cable from the other telco), the alarm clears emmediately for the
 1st telco and becomes RED for the 2nd telco. A loop also clears the alarm
 on both ports.

 The telco has told me to make sure that Line Impedance is 120 ohms but
 There's no where to set that and when I was reading, I was told that E1 is
 already 120 ohms so no need to change anything.

 Has anyone here experienced this?  What other things can I try?

 Thank you!

 Regards,
 Tim



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Re: [asterisk-users] Help With Physical Layer

[Duncan Turnbull]

-- Original Message --
From: Tony Kasule timotsm...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: 30/06/2015 8:34:47 p.m.
Subject: Re: [asterisk-users] Help With Physical Layer


Hello,

Anyone to help me with this issue? It has never worked :(


Hi Tony

I'm not familiar with the card you but 120 ohm is usually twisted pair, 
and 75 ohm is coax (usually). If it is changeable its usually done with 
jumpers on the card. I have used RAD modems before and their cable will 
have a specific pinout that you need to match. I doubt your digium card 
by default matches the pins, but maybe, and perhaps maybe your alcatel 
did. There will be a tx pair and an rx pair and you need to make sure 
they connect to the rx and tx pairs on the digium preferably with the 
same polarity. If you are getting no signal I would think its that. You 
can use a multimeter to check for voltage on the pins.


Once you get that sorted another catch maybe timing. You will need to 
take E1 timing from one or other of the Telco's, with one as the primary 
source. If they for some reason aren't synced you will get errors every 
so often


Good luck


Cheers Duncan


On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com 
wrote:

Hello users,

I have a Digium Te235 and asterisk 13  which have worked well with 1 
carrier but we have failed to add a 2nd carrier. The second telco 
brings their E1 line over finer, terminated in a RAD modem and they 
give me ethernet to the E1 card. It's the first time i am having 
install such a solution, which ideally would be not a big problem.


However, The  physical layer has failed to come up! I have tried the 
same setup in an Alcatel OmniOCX and it works well. I can confirm that 
the port is also well configured because when I interchange the cables 
(with the exiting cable from the other telco), the alarm clears 
emmediately for the 1st telco and becomes RED for the 2nd telco. A 
loop also clears the alarm on both ports.


The telco has told me to make sure that Line Impedance is 120 ohms but 
There's no where to set that and when I was reading, I was told that 
E1 is already 120 ohms so no need to change anything.


Has anyone here experienced this?  What other things can I try?

Thank you!

Regards,
Tim
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[asterisk-users] Help With Physical Layer

[Tony Kasule]

Hello users,

I have a Digium Te235 and asterisk 13  which have worked well with 1
carrier but we have failed to add a 2nd carrier. The second telco brings
their E1 line over finer, terminated in a RAD modem and they give me
ethernet to the E1 card. It's the first time i am having install such a
solution, which ideally would be not a big problem.

However, The  physical layer has failed to come up! I have tried the same
setup in an Alcatel OmniOCX and it works well. I can confirm that the port
is also well configured because when I interchange the cables (with the
exiting cable from the other telco), the alarm clears emmediately for the
1st telco and becomes RED for the 2nd telco. A loop also clears the alarm
on both ports.

The telco has told me to make sure that Line Impedance is 120 ohms but
There's no where to set that and when I was reading, I was told that E1 is
already 120 ohms so no need to change anything.

Has anyone here experienced this?  What other things can I try?

Thank you!

Regards,
Tim
-- 
_
-- 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

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Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

[Patrick Beaumont]

I have seen a similar problem occasionally. We will be doing maintenance on a 
customer's server and they will have one or two ghost channels on their 
machine hundreds of hours old but with no call associated with them. So far we 
have just been rebooting their server or issuing a hangup command to the 
channels.

From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Alex Villací­s Lasso 
a_villa...@palosanto.com
Sent: 08 April 2015 00:33
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Help debugging a possible SIP channel leak in 
11.17.0, possible race condition

El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió:
 I am trying to collect enough information about an problem a client is having 
 with its asterisk 11.17.0  x86_64. This issue was observed before in 1.8.20, 
 and we upgraded to 11.15.0 and then to 11.17.0 with no solution.

 Background: this client is a telemarketing call-center that generates 
 outgoing calls with close to a hundred agents operating simultaneously in 
 peak hours. The system uses asterisk with FreePBX 2.8. In order to generate 
 the calls, I wrote a program that
 connects to Asterisk using the AMI protocol. This program expects the SIP 
 agent extensions to be assigned as members of queues, of which there are 
 about 20, as shown below:

 9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s 
 talktime), W:0, C:581, A:260, SL:82.6% within 60s
Members:
   SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls 
 (last was 800 secs ago)
   SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls 
 (last was 708 secs ago)
   SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls 
 (last was 656 secs ago)
   SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls 
 (last was 789 secs ago)
No Callers

 The program runs queue show through AMI every few seconds. For each queue 
 to be used in telemarketing, the program counts the number of members that 
 are Not In Use. If at least one is found, it reads that many phone numbers 
 from the database and uses
 the AMI Originate command on each one, as follows:

 Action: Originate
 Channel: Local/NN@from-internal
 Exten: 
 Context: from-internal
 Priority: 1
 Async: true
 ActionID: xxx

 Here, NN is the number read from the database and  is the queue 
 extension in the FreePBX-created context that eventually runs the Queue() 
 dialplan application for the corresponding queue. This causes the call to be 
 connected between the
 outgoing number and the queue, and is then assigned to a queue member by 
 Asterisk. The dialplan is configured to route NN through one of a 
 series of SIP trunks using the outbound routes as configured by FreePBX.

 The issue is that although this strategy works correctly on the user's 
 machine for a few days, we have been observing that eventually the 
 application stops placing calls. The agents are all idle (all 90 to 100 of 
 them), but the queue show command shows
 them to be In Use on all queues. Furthermore, in normal operation, the 
 core show channels command shows at most one channel for each configured 
 SIP client in the Up state, but when calls stop being placed, the same 
 command reports multiple channels
 in the Up state, as follows (after sorting):

 Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192
 Local/9759315789@from-internal-a456;1ZOMBIE!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing
  Line)!9759315789!!!3!500!(None)!1428426084.169326
 Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323
 SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224
 SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
 Line)!110!!!3!590!(None)!1428425994.169124
 SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760
 SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273
 SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384
 SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850
 SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113
 SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
 SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47740413!!!3!327

Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

[Vinicius Fontes]

Have you tried Asterisk 13? The bridging mechanism has been completely
rewritten on Asterisk 12, so there's no longer channel masquerading and
zombie channels. Might be worth a try.

2015-04-07 20:33 GMT-03:00 Alex Villací­s Lasso a_villa...@palosanto.com:

 El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió:

  I am trying to collect enough information about an problem a client is
 having with its asterisk 11.17.0  x86_64. This issue was observed before in
 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution.

 Background: this client is a telemarketing call-center that generates
 outgoing calls with close to a hundred agents operating simultaneously in
 peak hours. The system uses asterisk with FreePBX 2.8. In order to generate
 the calls, I wrote a program that connects to Asterisk using the AMI
 protocol. This program expects the SIP agent extensions to be assigned as
 members of queues, of which there are about 20, as shown below:

 9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s
 talktime), W:0, C:581, A:260, SL:82.6% within 60s
Members:
   SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls
 (last was 800 secs ago)
   SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls
 (last was 708 secs ago)
   SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls
 (last was 656 secs ago)
   SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls
 (last was 789 secs ago)
No Callers

 The program runs queue show through AMI every few seconds. For each
 queue to be used in telemarketing, the program counts the number of members
 that are Not In Use. If at least one is found, it reads that many phone
 numbers from the database and uses the AMI Originate command on each one,
 as follows:

 Action: Originate
 Channel: Local/NN@from-internal
 Exten: 
 Context: from-internal
 Priority: 1
 Async: true
 ActionID: xxx

 Here, NN is the number read from the database and  is the
 queue extension in the FreePBX-created context that eventually runs the
 Queue() dialplan application for the corresponding queue. This causes the
 call to be connected between the outgoing number and the queue, and is then
 assigned to a queue member by Asterisk. The dialplan is configured to route
 NN through one of a series of SIP trunks using the outbound routes
 as configured by FreePBX.

 The issue is that although this strategy works correctly on the user's
 machine for a few days, we have been observing that eventually the
 application stops placing calls. The agents are all idle (all 90 to 100 of
 them), but the queue show command shows them to be In Use on all
 queues. Furthermore, in normal operation, the core show channels command
 shows at most one channel for each configured SIP client in the Up state,
 but when calls stop being placed, the same command reports multiple
 channels in the Up state, as follows (after sorting):

 Local/9757007441@from-internal-a447;2!macro-
 dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,
 !47740412!!!3!572!(None)!1428426012.169192
 Local/9759315789@from-internal-a456;1ZOMBIE!
 from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing
 Line)!9759315789!!!3!500!(None)!1428426084.169326
 Local/9759315789@from-internal-a456;2!macro-
 dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,
 !47740412!!!3!500!(None)!1428426084.169323
 SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/
 5547740413-00014c62!1428426022.169224
 SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing
 Line)!110!!!3!590!(None)!1428425994.169124
 SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760
 SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!
 1428426052.169273
 SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384
 SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!
 SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!
 1428426574.169850
 SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!
 1428425991.169113
 SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
 SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!
 SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493
 SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,,
 trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510
 SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing
 Line)!121!!!3!596!(None)!1428425988.169111
 SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing
 Line)!122!!!3!677!(None)!1428425906.168693
 SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!
 

[asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

[Alex Villací­s Lasso]

I am trying to collect enough information about an problem a client is having 
with its asterisk 11.17.0  x86_64. This issue was observed before in 1.8.20, 
and we upgraded to 11.15.0 and then to 11.17.0 with no solution.

Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that 
connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which there are about 20, as shown below:


9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s 
talktime), W:0, C:581, A:260, SL:82.6% within 60s
   Members:
  SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls (last 
was 800 secs ago)
  SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls (last 
was 708 secs ago)
  SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls (last 
was 656 secs ago)
  SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls (last 
was 789 secs ago)
   No Callers

The program runs queue show through AMI every few seconds. For each queue to be used in telemarketing, the program counts the number of members that are Not In Use. If at least one is found, it reads that many phone numbers from the database and uses 
the AMI Originate command on each one, as follows:


Action: Originate
Channel: Local/NN@from-internal
Exten: 
Context: from-internal
Priority: 1
Async: true
ActionID: xxx

Here, NN is the number read from the database and  is the queue extension in the FreePBX-created context that eventually runs the Queue() dialplan application for the corresponding queue. This causes the call to be connected between the 
outgoing number and the queue, and is then assigned to a queue member by Asterisk. The dialplan is configured to route NN through one of a series of SIP trunks using the outbound routes as configured by FreePBX.


The issue is that although this strategy works correctly on the user's machine for a few days, we have been observing that eventually the application stops placing calls. The agents are all idle (all 90 to 100 of them), but the queue show command shows 
them to be In Use on all queues. Furthermore, in normal operation, the core show channels command shows at most one channel for each configured SIP client in the Up state, but when calls stop being placed, the same command reports multiple channels 
in the Up state, as follows (after sorting):


Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192
Local/9759315789@from-internal-a456;1ZOMBIE!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing
 Line)!9759315789!!!3!500!(None)!1428426084.169326
Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323
SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224
SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
Line)!110!!!3!590!(None)!1428425994.169124
SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760
SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273
SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384
SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850
SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113
SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493
SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,,trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510
SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
Line)!121!!!3!596!(None)!1428425988.169111
SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing 
Line)!122!!!3!677!(None)!1428425906.168693
SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!320!(None)!1428426264.169499
SIP/123-00014e35!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!121!(None)!1428426463.169735
SIP/123-00014e9e!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740410/5556350254,300,!47740410!!!3!13!(None)!1428426570.169844

Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

[Alex Villací­s Lasso]

El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió:

I am trying to collect enough information about an problem a client is having 
with its asterisk 11.17.0  x86_64. This issue was observed before in 1.8.20, 
and we upgraded to 11.15.0 and then to 11.17.0 with no solution.

Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that 
connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which there are about 20, as shown below:


9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s 
talktime), W:0, C:581, A:260, SL:82.6% within 60s
   Members:
  SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls (last 
was 800 secs ago)
  SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls (last 
was 708 secs ago)
  SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls (last 
was 656 secs ago)
  SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls (last 
was 789 secs ago)
   No Callers

The program runs queue show through AMI every few seconds. For each queue to be used in telemarketing, the program counts the number of members that are Not In Use. If at least one is found, it reads that many phone numbers from the database and uses 
the AMI Originate command on each one, as follows:


Action: Originate
Channel: Local/NN@from-internal
Exten: 
Context: from-internal
Priority: 1
Async: true
ActionID: xxx

Here, NN is the number read from the database and  is the queue extension in the FreePBX-created context that eventually runs the Queue() dialplan application for the corresponding queue. This causes the call to be connected between the 
outgoing number and the queue, and is then assigned to a queue member by Asterisk. The dialplan is configured to route NN through one of a series of SIP trunks using the outbound routes as configured by FreePBX.


The issue is that although this strategy works correctly on the user's machine for a few days, we have been observing that eventually the application stops placing calls. The agents are all idle (all 90 to 100 of them), but the queue show command shows 
them to be In Use on all queues. Furthermore, in normal operation, the core show channels command shows at most one channel for each configured SIP client in the Up state, but when calls stop being placed, the same command reports multiple channels 
in the Up state, as follows (after sorting):


Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192
Local/9759315789@from-internal-a456;1ZOMBIE!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing
 Line)!9759315789!!!3!500!(None)!1428426084.169326
Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323
SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224
SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
Line)!110!!!3!590!(None)!1428425994.169124
SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760
SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273
SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384
SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850
SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113
SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493
SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,,trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510
SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing 
Line)!121!!!3!596!(None)!1428425988.169111
SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing 
Line)!122!!!3!677!(None)!1428425906.168693
SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!320!(None)!1428426264.169499
SIP/123-00014e35!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!121!(None)!1428426463.169735
SIP/123-00014e9e!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740410/5556350254,300,!47740410!!!3!13!(None)!1428426570.169844

[asterisk-users] help : annoucement queue

[Anicet LANJANIAINA]

Hi everybody,

I've a matter with the queue annoucement with the thereare, because if 
I put just one member in my configuration (member = SIP/2098), the ivr 
gave me that I was the firt or second in the next at the queue. But the 
problem is, if I add one member (eg: member = SIP/2098 and member = 
SIP/2099), the ivr don't gave me the range but It play the background 
sound that I declare in my musiconhold.


Very thanks for your helps.

Have a nice day.

--
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Gulfsat Madagascar
(+261) 345 600 259
Service Technique -Blueline Madagascar www.blueline.mg -
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Today's Topics:

1. PJSIP Video on WebRTC Ast 13 (Gosmac)
2. Re: res_xmpp.c:3468 xmpp_client_reconnect: (ricky gutierrez)
3. Re: Asterisk 13 : SILK codec ? (Steve Murphy)
4. Re: Asterisk switching bridge to native_rtp even with
   direct_media=no (Matthew Jordan)
5. Re: Asterisk 13 : SILK codec ? (Matthew Jordan)
6. Problems playing an audio file over an   intercom/paging system
   (Tech Support)
7. Asterisk on OpenWrt (first time user) (Sebastian Kemper)
8. Dahdi ISDN logging (Grant Bagdasarian)
9. Re: Dahdi ISDN logging (Tony Mountifield)
   10. UNREACHABLE peer (thufir)
   11. Re: UNREACHABLE peer (dotnetdub)
   12. Re: UNREACHABLE peer (thufir)
   13. Re: UNREACHABLE peer (thufir)
   14. Re: UNREACHABLE peer (thufir)
   15. Re: Caller ID Names (Jordan Cook - Gyron Networks)
   16. Re: Caller ID Names (Jordan Cook - Gyron Networks)


--

Message: 1
Date: Thu, 19 Mar 2015 12:36:54 -0430
From: Gosmac gosee...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PJSIP Video on WebRTC Ast 13
Message-ID: 9ce929c6-8e20-4794-a44f-e55ac877d...@gmail.com
Content-Type: text/plain; charset=utf-8

Hey i have an interesting topic to discuss here.

The main goal here is to be able to make a video call between two WebRTC 
endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 
should support .

the problems that i faced with this is the following and i hope i could get an 
advise here.

asterisk 13 vanilla version has some issues marking the video packets this 
complain web browser specially VP8 codecs so a friend of mine help me to patch 
res_rtp_asterisk and now asterisk is marking video streams :) it just mark 
video packets not touch anything else and web browser show video on web page 
now I?m using online demo http://tryit.jssip.net/ is stable and get more 
updates than sipml5. so i try echo() dialplan test and everything work perfect 
on echo test :).

i have two questions and i hope you could give me some advise.

1) after marking video packet I?m able to make Dial() between two webrtc peers 
but i get one way audio and video on callee party, ?after 3 minutes on call? i 
get two way audio and video on all parties seems to be not just a problem on a 
missing keyframe.

  1.1) the 3 minutes delay only happen using chrome stable , could be a dtls 
problem when asterisk make an offer to other endpoint?
  1.2) when i use chrome-dev and i disable dlts encryption everything work 
perfect on video call.

2) after marking video packets i realize that when you make a call with video 
and you involve on dialplan an application like playback or music on hold any 
application that  played audio files (audio and video never work).
  
2.1) asterisk is muggling the audio and video streams ?


This is good information for all guys out there that wants to support video on 
webrtc in asterisk 13

Javier Riveros


--

Message: 2
Date: Thu, 19 Mar 2015 11:42:36 -0600
From: ricky gutierrez xserverli...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_xmpp.c:3468 xmpp_client_reconnect:
Message-ID:
CAL_GE3To07V8gZ6SaCFhO1=x1jakto595kctmnlnkaaa-bq...@mail.gmail.com
Content-Type: text/plain; charset=UTF-8

2015-03-18 12:54 GMT-06:00 ricky gutierrez xserverli...@gmail.com:


I'm confused this is not a patch, it's just garbage ;), I'm making a
connection xmpp with asterisk and not connected, at the cli shows me
the message every second:

RROR[2545]: res_xmpp.c:3468 

[asterisk-users] help : annoucement queue

[Anicet LANJANIAINA]

Hi everybody,

I've a matter with the queue annoucement with the thereare, because if 
I put just one member in my configuration (member = SIP/2098), the ivr 
gave me that I was the firt or second in the next at the queue. But the 
problem is, if I add one member (eg: member = SIP/2098 and member = 
SIP/2099), the ivr don't gave me the range but It play the background 
sound that I declare in my musiconhold.


ipbx-digue*CLI core show version
Asterisk 1.8.13.1~dfsg1-3+deb7u3 built by pbuilder @ pungenday on a 
x86_64 running Linux on 2014-01-04 01:03:48 UTC


Very thanks for your helps.

Have a nice day.

--
--
Anicet LANJANIAINA
Gulfsat Madagascar
(+261) 345 600 259
Service Technique -Blueline Madagascar www.blueline.mg -
Facebook : blueline Madagascar – Twitter : blueline_MG

Please think about the environment before printing this e-mail.


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[asterisk-users] Help! How to make Asterisk support ICE in public network

[曹贵林]

Hi friends,
I am just starting use asterisk for our VoIP server. It works fine in LAN. But 
when it is deployed in public network(with a public IP), the SIP clients in 
different NAT fails to communicate with each other. I have set 'icesupport' to 
'yes' in sip.conf and set STURN and TURN server in rtp.conf. It still fails! 


Hope someone to help me out! Thanks in advance:) 


This is the output of CLI:
~
  == Using SIP RTP CoS mark 5
-- Called SIP/6003
-- SIP/6003-0001 is ringing
-- SIP/6003-0001 is ringing
-- SIP/6003-0001 is ringing
-- SIP/6003-0001 is ringing
-- SIP/6003-0001 is ringing
-- SIP/6003-0001 answered SIP/6004-
-- Channel SIP/6004- joined 'simple_bridge' basic-bridge 
2a01fb30-96e2-48b7-baaa-c2f172127c07
-- Channel SIP/6003-0001 joined 'simple_bridge' basic-bridge 
2a01fb30-96e2-48b7-baaa-c2f172127c07
Bridge 2a01fb30-96e2-48b7-baaa-c2f172127c07: switching from 
simple_bridge technology to native_rtp
Remotely bridged 'SIP/6003-0001' and 'SIP/6004-' - media 
will flow directly between them
Remotely bridged 'SIP/6003-0001' and 'SIP/6004-' - media 
will flow directly between them
0x7f5968006760 -- Probation passed - setting RTP source address to 
114.81.254.172:4145
0x1fefbb0 -- Probation passed - setting RTP source address to 
114.92.58.65:7076
st-srv-cs2*CLI 
st-srv-cs2*CLI 
st-srv-cs2*CLI 
-- Channel SIP/6004- left 'native_rtp' basic-bridge 
2a01fb30-96e2-48b7-baaa-c2f172127c07
  == Spawn extension (my-phone, 6003, 1) exited non-zero on 'SIP/6004-'
-- Channel SIP/6003-0001 left 'native_rtp' basic-bridge 
2a01fb30-96e2-48b7-baaa-c2f172127c07
[Mar 18 12:04:22] NOTICE[30157]: chan_sip.c:23890 handle_response_peerpoke: 
Peer '6003' is now Lagged. (3285ms / 2000ms)
[Mar 18 12:04:33] NOTICE[30157]: chan_sip.c:23890 handle_response_peerpoke: 
Peer '6003' is now Reachable. (1244ms / 2000ms)
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5




--
Dennis Cao (曹贵林 )-- 
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[asterisk-users] help

[chandapure shiva]

Dear all,
   I was going through sip.conf file and i am not able to
understand the working and how to test the functionality of below fields.


1.tcpauthlimit
2.tcpauthtimeout

any inputs regarding this will appreciated, thanks in advance


Thanks
SHIVAKUMAR
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Re: [asterisk-users] help

[Rusty Newton]

On Fri, Aug 22, 2014 at 6:48 AM, chandapure shiva
chandapure.shiv...@gmail.com wrote:
 Dear all,
I was going through sip.conf file and i am not able to
 understand the working and how to test the functionality of below fields.


 1.tcpauthlimit
 2.tcpauthtimeout

 any inputs regarding this will appreciated, thanks in advance

Do you have a specific question?

What do you mean How to test the functionality of below fields?

Here is the documentation on those options from the sip.conf sample file:


;tcpauthtimeout = 30; tcpauthtimeout specifies the maximum number
; of seconds a client has to authenticate.  If
; the client does not authenticate beofre this
; timeout expires, the client will be
; disconnected. (default: 30 seconds)

;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
; unauthenticated sessions that will be allowed
; to connect at any given time. (default: 100)



-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Help with a bug

[A J Stiles]

On Wednesday 23 Apr 2014, CDR wrote:
 Dear friends
 I filed a bug
 https://issues.asterisk.org/jira/browse/ASTERISK-23656
 but I am wondering if somebody can figure a workaround. I am stuck
 trying to deliver an application.
 The case is this: A Record is executed and an immediate Playback
 follows. Asterisk returns an error, saying that the file does not
 exist, but a few seconds later, it does.
 It does not help if after the Record application I do SHELL(sync).
 Asterisk has not flushed the file out to the OS and it already
 returned. Maybe the application record should have a parameter about
 this behavior. For some application is fine, for some others is not.

You have run up against a race condition.  Not unusual in asynchronous 
environments.

As a workaround, I would advise handing over starting the recording to an AGI 
script.  But don't do the usual
fork  exit
Instead, start the recording; and then just loop, until *either* a -e test 
shows that the recording file definitely exists within the file system, or you 
have waited far too long.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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[asterisk-users] Help with a bug (CDR)

[CDR]

I fund the issue and it was in my own code. I apologize.

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[asterisk-users] Help with a bug

[CDR]

Dear friends
I filed a bug
https://issues.asterisk.org/jira/browse/ASTERISK-23656
but I am wondering if somebody can figure a workaround. I am stuck
trying to deliver an application.
The case is this: A Record is executed and an immediate Playback
follows. Asterisk returns an error, saying that the file does not
exist, but a few seconds later, it does.
It does not help if after the Record application I do SHELL(sync).
Asterisk has not flushed the file out to the OS and it already
returned. Maybe the application record should have a parameter about
this behavior. For some application is fine, for some others is not.

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Re: [asterisk-users] Help with a bug

[Josh Metzger]

How many seconds later does the file show up?  Can you just throw in a
Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a
second or two of delay be an issue (or does it still not work)?

-Josh



On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote:

 Dear friends
 I filed a bug
 https://issues.asterisk.org/jira/browse/ASTERISK-23656
 but I am wondering if somebody can figure a workaround. I am stuck
 trying to deliver an application.
 The case is this: A Record is executed and an immediate Playback
 follows. Asterisk returns an error, saying that the file does not
 exist, but a few seconds later, it does.
 It does not help if after the Record application I do SHELL(sync).
 Asterisk has not flushed the file out to the OS and it already
 returned. Maybe the application record should have a parameter about
 this behavior. For some application is fine, for some others is not.

 --
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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Help with a bug

[Josh Metzger]

As a second possible solution, instead of Record, could you use
MixMonitor, then run StopMixMonitor and THEN do your Playback?  That
should definitely make sure the recording file is closed and the file
handle released.

-Josh


On Wed, Apr 23, 2014 at 2:29 PM, Josh Metzger joshdmetz...@gmail.comwrote:

 How many seconds later does the file show up?  Can you just throw in a
 Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a
 second or two of delay be an issue (or does it still not work)?

 -Josh



 On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote:

 Dear friends
 I filed a bug
 https://issues.asterisk.org/jira/browse/ASTERISK-23656
 but I am wondering if somebody can figure a workaround. I am stuck
 trying to deliver an application.
 The case is this: A Record is executed and an immediate Playback
 follows. Asterisk returns an error, saying that the file does not
 exist, but a few seconds later, it does.
 It does not help if after the Record application I do SHELL(sync).
 Asterisk has not flushed the file out to the OS and it already
 returned. Maybe the application record should have a parameter about
 this behavior. For some application is fine, for some others is not.

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Help with a bug

[Eric Wieling]

Doesn't MixMonitor use sox to combine the incoming and outgoing recordings?   
If so, I'd expect MixMonitor to add MORE delay, not less.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Metzger
Sent: Wednesday, April 23, 2014 2:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with a bug

As a second possible solution, instead of Record, could you use MixMonitor, 
then run StopMixMonitor and THEN do your Playback?  That should definitely 
make sure the recording file is closed and the file handle released.


-Josh



On Wed, Apr 23, 2014 at 2:29 PM, Josh Metzger joshdmetz...@gmail.com wrote:


How many seconds later does the file show up?  Can you just throw in a 
Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a second 
or two of delay be an issue (or does it still not work)?


-Josh




On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote:


Dear friends
I filed a bug
https://issues.asterisk.org/jira/browse/ASTERISK-23656
but I am wondering if somebody can figure a workaround. I am 
stuck
trying to deliver an application.
The case is this: A Record is executed and an immediate Playback
follows. Asterisk returns an error, saying that the file does 
not
exist, but a few seconds later, it does.
It does not help if after the Record application I do 
SHELL(sync).
Asterisk has not flushed the file out to the OS and it already
returned. Maybe the application record should have a parameter 
about
this behavior. For some application is fine, for some others is 
not.

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Re: [asterisk-users] Help with a bug

[Josh Metzger]

That's the case with Monitor (apparently), but MixMonitor grabs both
ends of the call.  On a system I ran with lots of MixMonitor recording,
Asterisk renamed / moved the recording file when a call completed, and that
happened without any delay at all.  Only one file was created for the
entire call.


On Wed, Apr 23, 2014 at 2:39 PM, Eric Wieling ewiel...@nyigc.com wrote:

 Doesn't MixMonitor use sox to combine the incoming and outgoing
 recordings?   If so, I'd expect MixMonitor to add MORE delay, not less.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Metzger
 Sent: Wednesday, April 23, 2014 2:35 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Help with a bug

 As a second possible solution, instead of Record, could you use
 MixMonitor, then run StopMixMonitor and THEN do your Playback?  That
 should definitely make sure the recording file is closed and the file
 handle released.


 -Josh



 On Wed, Apr 23, 2014 at 2:29 PM, Josh Metzger joshdmetz...@gmail.com
 wrote:


 How many seconds later does the file show up?  Can you just throw
 in a Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even
 a second or two of delay be an issue (or does it still not work)?


 -Josh




 On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote:


 Dear friends
 I filed a bug
 https://issues.asterisk.org/jira/browse/ASTERISK-23656
 but I am wondering if somebody can figure a workaround. I
 am stuck
 trying to deliver an application.
 The case is this: A Record is executed and an immediate
 Playback
 follows. Asterisk returns an error, saying that the file
 does not
 exist, but a few seconds later, it does.
 It does not help if after the Record application I do
 SHELL(sync).
 Asterisk has not flushed the file out to the OS and it
 already
 returned. Maybe the application record should have a
 parameter about
 this behavior. For some application is fine, for some
 others is not.

 --

 _
 -- Bandwidth and Colocation Provided by
 http://www.api-digital.com --
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 every Thurs:
http://www.asterisk.org/hello

 asterisk-users mailing list
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Re: [asterisk-users] Help with a bug

[Steve Edwards]

On Wed, 23 Apr 2014, CDR wrote:


The case is this: A Record is executed and an immediate Playback
follows. Asterisk returns an error, saying that the file does not
exist, but a few seconds later, it does.


A simple test:

exten = *,n,record(foo.wav)
exten = *,n,playback(foo)

works as expected for me with Asterisk 11.8.1.

I notice in the console log you uploaded, you have a file name of 
'180-industry:sln'


The syntax for record says 'filename.format' not 'filename:format'

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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files

[Ioan Indreias]

Hello Steve,

Have you tried to send the automated call to your dialplan instead of the
phone?

For example, instead of calling SIP/aastra_phone call
Local/aastra_phone@auto-answer-context and tweak auto-answer-context from
your dialplan as needed.

HTH,
Ioan


On Tue, Jan 28, 2014 at 6:56 PM, Steve McCann srmcc...@gmail.com wrote:

 Hello All,

 I've asked this on the asterisk-dev list, so sorry for cross-posting. So
 far I'm not sure how to accomplish this without looking at the source code
 or looking at some other way to get around this issue.

 I'm trying to have an automated call to an Aastra SIP phone and have the
 call auto-answeredby the phone. I know that a SIP call placed to the phone
 can be auto-answered if a certain SIP header is added to the call. I am
 able to apply the SIP headers manually and get that working (using
 SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but for
 call files, I don't seem to be able to edit any of the sip headers - there
 is only basic customizations allowed to setup the calls.

 Does anyone know how I could place automated outgoing calls that would
 have the proper sip headers added to it that would allow the call to be
 auto-answered?

 I've also posted this question to the forums here:
 http://forums.asterisk.org/viewtopic.php?f=1t=89190

 Many thanks,
 Steve

 

 http://www.stevemccann.net


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[asterisk-users] [HELP]: Auto-answering calls placed from call files

[Steve McCann]

Hello All,

I've asked this on the asterisk-dev list, so sorry for cross-posting. So
far I'm not sure how to accomplish this without looking at the source code
or looking at some other way to get around this issue.

I'm trying to have an automated call to an Aastra SIP phone and have the
call auto-answeredby the phone. I know that a SIP call placed to the phone
can be auto-answered if a certain SIP header is added to the call. I am
able to apply the SIP headers manually and get that working (using
SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but for
call files, I don't seem to be able to edit any of the sip headers - there
is only basic customizations allowed to setup the calls.

Does anyone know how I could place automated outgoing calls that would have
the proper sip headers added to it that would allow the call to be
auto-answered?

I've also posted this question to the forums here:
http://forums.asterisk.org/viewtopic.php?f=1t=89190

Many thanks,
Steve



http://www.stevemccann.net
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Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files

[Gareth Blades]

On 28/01/14 16:56, Steve McCann wrote:

Hello All,

I've asked this on the asterisk-dev list, so sorry for cross-posting. 
So far I'm not sure how to accomplish this without looking at the 
source code or looking at some other way to get around this issue.


I'm trying to have an automated call to an Aastra SIP phone and have 
the call auto-answeredby the phone. I know that a SIP call placed to 
the phone can be auto-answered if a certain SIP header is added to the 
call. I am able to apply the SIP headers manually and get that working 
(using SIPAddHeader(Alert-Info: info=alert-autoanswer) in the 
dialplan, but for call files, I don’t seem to be able to edit any of 
the sip headers - there is only basic customizations allowed to setup 
the calls.


Does anyone know how I could place automated outgoing calls that would 
have the proper sip headers added to it that would allow the call to 
be auto-answered?


I've also posted this question to the forums here: 
http://forums.asterisk.org/viewtopic.php?f=1t=89190 
http://forums.asterisk.org/viewtopic.php?f=1t=89190


Many thanks,
Steve




So I take it in the call file you have it set to call Dial(SIP/something) ?
If rather than dialling the sip destination immediately you dialled a 
local channel then it could add the custom header and then initiate the 
dial to the sip destination.



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Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files

[Matthew Jordan]

On Tue, Jan 28, 2014 at 10:56 AM, Steve McCann srmcc...@gmail.com wrote:
 Hello All,

 I've asked this on the asterisk-dev list, so sorry for cross-posting. So far
 I'm not sure how to accomplish this without looking at the source code or
 looking at some other way to get around this issue.


 I'm trying to have an automated call to an Aastra SIP phone and have the
 call auto-answeredby the phone. I know that a SIP call placed to the phone
 can be auto-answered if a certain SIP header is added to the call. I am able
 to apply the SIP headers manually and get that working (using
 SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but for
 call files, I don't seem to be able to edit any of the sip headers - there
 is only basic customizations allowed to setup the calls.

 Does anyone know how I could place automated outgoing calls that would have
 the proper sip headers added to it that would allow the call to be
 auto-answered?

 I've also posted this question to the forums here:
 http://forums.asterisk.org/viewtopic.php?f=1t=89190

 Many thanks,
 Steve


This isn't a development question, as it doesn't relate to the actual
Asterisk source code itself. Cross-posting across the -dev and -users
lists isn't helpful either, as pretty much everyone who is subscribed
to the asterisk-dev list is also subscribed to the asterisk-users
list.

As SIPAddHeader is a dialplan application and not a dialplan function,
it cannot be used from a call file. One approach to performing an
outbound call that requires SIPAddHeader - and that doesn't rely on
undocumented behaviour - is to use the call file to create a Local
channel in the dialplan that dials the SIP channel, and use
SIPAddHeader from there. A quick Google indicates others have used a
similar approach in the past as well [1].

[1] http://lists.digium.com/pipermail/asterisk-users/2008-January/204375.html

Matt

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files

[Tech Support]

Although I haven't tried this for this particular example, instead of
using a .call file, you could probably originate a call using Ryan Bullock's
Asterisk::AMI PERL module
http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.8. It's one of the most
valuable tools that I have and I've written literally hundreds of PERL
scripts using it. You should check it out. It's got good documentation and
examples to go along with it. I also use the AGISpeedy FastAGI package
written in PERL and there's also an AGISpeedy package written in php that is
also a valuable tool.
Regards;
John

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Tuesday, January 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [HELP]: Auto-answering calls placed from call
files

On Tue, Jan 28, 2014 at 10:56 AM, Steve McCann srmcc...@gmail.com wrote:
 Hello All,

 I've asked this on the asterisk-dev list, so sorry for cross-posting. 
 So far I'm not sure how to accomplish this without looking at the 
 source code or looking at some other way to get around this issue.


 I'm trying to have an automated call to an Aastra SIP phone and have 
 the call auto-answeredby the phone. I know that a SIP call placed to 
 the phone can be auto-answered if a certain SIP header is added to the 
 call. I am able to apply the SIP headers manually and get that working 
 (using
 SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but 
 for call files, I don't seem to be able to edit any of the sip headers 
 - there is only basic customizations allowed to setup the calls.

 Does anyone know how I could place automated outgoing calls that would 
 have the proper sip headers added to it that would allow the call to 
 be auto-answered?

 I've also posted this question to the forums here:
 http://forums.asterisk.org/viewtopic.php?f=1t=89190

 Many thanks,
 Steve


This isn't a development question, as it doesn't relate to the actual
Asterisk source code itself. Cross-posting across the -dev and -users lists
isn't helpful either, as pretty much everyone who is subscribed to the
asterisk-dev list is also subscribed to the asterisk-users list.

As SIPAddHeader is a dialplan application and not a dialplan function, it
cannot be used from a call file. One approach to performing an outbound call
that requires SIPAddHeader - and that doesn't rely on undocumented behaviour
- is to use the call file to create a Local channel in the dialplan that
dials the SIP channel, and use SIPAddHeader from there. A quick Google
indicates others have used a similar approach in the past as well [1].

[1]
http://lists.digium.com/pipermail/asterisk-users/2008-January/204375.html

Matt

--
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com  http://asterisk.org

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Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

[Brandon Coale]

On 1/9/2014 12:12 PM, Jeremy Kister wrote:

On 1/8/2014 9:12 PM, Brandon Coale wrote:

However, I am not able to get app_swift to compile.  I am running
Asterisk 11.6.0 and CentOS 6.4 64-bit.

I am wondering if anyone else out there has been able to get app_swift
working with Asterisk 11 and could share any tricks they used to get it
installed?


can you pastie your configure and make ?

I don't have Cepstral6 but did submit tweaks to the code that should 
have made it Cepstral6 compatible.


Also since you recently spent money with Cepstral, they'll help you. 
They've got at least one guy who understands the app_swift code and 
was working on forking it as an official version.


Thanks to Jeremy Kister's advice, I was able to get this compiled. He 
suggested these two commands:

yum update -y
yum install asterisk-devel

These commands upgraded my Asterisk to 11.7.0 and installed the Asterisk 
development files that I was missing.  After I did that, the make worked 
beautifully:


[root@dialer app_swift]# make


 ____
(_)  / __)  _
_      ___ _ _ _ _ _| |__ _| |_
   ( |  _ \|  _ \ /___) | | | (_   __|_   _)
   / ___ | |_| | |_| |   |___ | | | | | | || |_
   \_|  __/|  __/ () |___/ \___/|_| |_| \__)
 |_|   |_|

gcc -I/opt/swift/include -I/usr/include -g -Wall -fPIC -D_SWIFT_VER_6 
-D_AST_VER_11   -c -o app_swift.o app_swift.c
gcc -shared -Xlinker -x -o app_swift.so -L/opt/swift/lib -L/usr/lib 
-lswift -lceplang_en -lceplex_us app_swift.o


  
  *  Run 'make install' to install the app_swift module. *
  

When I ran the make install, I did get install: cannot create regular 
file `/usr/lib/asterisk/modules': No such file or directory, but I just 
changed the line SYS_LIB_DIR=/usr/lib in the Makefile to

SYS_LIB_DIR=/usr/lib64 and it worked fine.

Just wanted to post this resolution in case it helps someone else out there.

Brandon


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Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

[Brandon Coale]

On 1/13/2014 6:13 PM, Justin Killen wrote:

Another option is to use an MRCP server like UniMRCP along with the Cepstral plugin.  
One very nice thing about this approach is that there is less 'cepstral version' 
- 'asterisk version' dependency, which is a problem with the current 
app_swift module (each app_swift version is designed to work on specifically one 
version of asterisk and one version of cepstral).

Cepstral provides details here: http://www.cepstral.com/en/telephony/mrcp
Information on the open-source uniMRCP can be found here: 
http://www.unimrcp.org/
Information for connecting asterisk to uniMRCP can be found here (although it 
seems to be having issues ATM): 
http://code.google.com/p/unimrcp/wiki/asteriskUniMRCP



-Justin

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Friday, January 10, 2014 1:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

Wouldn't it be easier in your case to pay somebody to do the job? I doubt that 
it would take
more than a couple of minutes to compile, install, and configure the package. 
Maybe some things
need to get adjusted as the author has abandoned the project (at least there is 
no longer a
project web page) and the latest sources are about 2 years old.

Building from sources is not that difficult, but if you don't have a proper 
configure script you
are responsible that all prerequisites are met, which can be time consuming if 
you don't know
your distro well enough. Here, there is no configure script and some things, 
which might be
invalid for your machine, are hand coded inside the Makefile. Nothing 
spectacular, but you could
end up asking a lot more questions that have nothing to do with asterisk.

jg


Justin,

Thank you very much for the information, that is great to learn that I 
have that option as well.


Brandon



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Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

[Justin Killen]

Another option is to use an MRCP server like UniMRCP along with the Cepstral 
plugin.  One very nice thing about this approach is that there is less 
'cepstral version' - 'asterisk version' dependency, which is a problem with 
the current app_swift module (each app_swift version is designed to work on 
specifically one version of asterisk and one version of cepstral).

Cepstral provides details here: http://www.cepstral.com/en/telephony/mrcp
Information on the open-source uniMRCP can be found here: 
http://www.unimrcp.org/
Information for connecting asterisk to uniMRCP can be found here (although it 
seems to be having issues ATM): 
http://code.google.com/p/unimrcp/wiki/asteriskUniMRCP



-Justin

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Friday, January 10, 2014 1:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

Wouldn't it be easier in your case to pay somebody to do the job? I doubt that 
it would take 
more than a couple of minutes to compile, install, and configure the package. 
Maybe some things 
need to get adjusted as the author has abandoned the project (at least there is 
no longer a 
project web page) and the latest sources are about 2 years old.

Building from sources is not that difficult, but if you don't have a proper 
configure script you 
are responsible that all prerequisites are met, which can be time consuming if 
you don't know 
your distro well enough. Here, there is no configure script and some things, 
which might be 
invalid for your machine, are hand coded inside the Makefile. Nothing 
spectacular, but you could 
end up asking a lot more questions that have nothing to do with asterisk.

jg

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Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

[jg]

Wouldn't it be easier in your case to pay somebody to do the job? I doubt that it would take 
more than a couple of minutes to compile, install, and configure the package. Maybe some things 
need to get adjusted as the author has abandoned the project (at least there is no longer a 
project web page) and the latest sources are about 2 years old.


Building from sources is not that difficult, but if you don't have a proper configure script you 
are responsible that all prerequisites are met, which can be time consuming if you don't know 
your distro well enough. Here, there is no configure script and some things, which might be 
invalid for your machine, are hand coded inside the Makefile. Nothing spectacular, but you could 
end up asking a lot more questions that have nothing to do with asterisk.


jg

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Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

[Jeremy Kister]

On 1/8/2014 9:12 PM, Brandon Coale wrote:

However, I am not able to get app_swift to compile.  I am running
Asterisk 11.6.0 and CentOS 6.4 64-bit.

I am wondering if anyone else out there has been able to get app_swift
working with Asterisk 11 and could share any tricks they used to get it
installed?


can you pastie your configure and make ?

I don't have Cepstral6 but did submit tweaks to the code that should 
have made it Cepstral6 compatible.


Also since you recently spent money with Cepstral, they'll help you. 
They've got at least one guy who understands the app_swift code and was 
working on forking it as an official version.


--

Jeremy Kister
http://jeremy.kister.net./


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Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

[Brandon Coale]

Jeremy,
 
Thanks very much.  I admit I am new to compiling software on Linux.  I am not 
sure where I would grab the configure?  By make I believe you mean the Makefile 
that I downloaded from https://github.com/darrensessions/app_swift?
 
I went ahead and opened a support ticket with Cepstral, so we can wait until I 
hear back from them if you want.  If we come up with a solution, I will write 
back with what they come up with.
 
Thanks again,
Brandon


- Original Message -
From: Jeremy Kister asterisk...@jeremykister.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Cc: 
Sent: Thursday, January 9, 2014 12:12 PM
Subject: Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

On 1/8/2014 9:12 PM, Brandon Coale wrote:
 However, I am not able to get app_swift to compile.  I am running
 Asterisk 11.6.0 and CentOS 6.4 64-bit.
 
 I am wondering if anyone else out there has been able to get app_swift
 working with Asterisk 11 and could share any tricks they used to get it
 installed?

can you pastie your configure and make ?

I don't have Cepstral6 but did submit tweaks to the code that should have made 
it Cepstral6 compatible.

Also since you recently spent money with Cepstral, they'll help you. They've 
got at least one guy who understands the app_swift code and was working on 
forking it as an official version.

-- 
Jeremy Kister
http://jeremy.kister.net./


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Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

[Brandon Coale]

On 1/9/2014 12:12 PM, Jeremy Kister wrote:

On 1/8/2014 9:12 PM, Brandon Coale wrote:

However, I am not able to get app_swift to compile.  I am running
Asterisk 11.6.0 and CentOS 6.4 64-bit.

I am wondering if anyone else out there has been able to get app_swift
working with Asterisk 11 and could share any tricks they used to get it
installed?


can you pastie your configure and make ?

I don't have Cepstral6 but did submit tweaks to the code that should 
have made it Cepstral6 compatible.


Also since you recently spent money with Cepstral, they'll help you. 
They've got at least one guy who understands the app_swift code and 
was working on forking it as an official version.



Jeremy,

I heard back from Cepstral support, and sadly I was told Unfortunately, 
we do not provide support for app_swift. app_swift is an open source 
program that was not written by Cepstral.


I am new to installing software on Linux from source code.  I am 
wondering if you would mind giving me some details on where I can find 
my configure and make?


My /opt/swift directory looks like this:
[root@dialer swift]# ls -l
total 28
drwxr-xr-x 2 root root 4096 Dec 30 19:47 bin
drwxr-xr-x 2 root root 4096 Dec 30 19:47 doc
drwxr-xr-x 2 root root 4096 Dec 30 20:19 etc
drwxr-xr-x 2 root root 4096 Dec 30 19:47 include
drwxr-xr-x 2 root root 4096 Dec 30 19:47 lib
drwxr-xr-x 2 root root 4096 Dec 30 19:47 sfx
drwxr-xr-x 3 root root 4096 Dec 30 19:47 voices

I downloaded app_swift to /root/app_swift and the directory looks like this:
[root@dialer app_swift]# ls -l
total 48
-rw-r--r-- 1 root root 20073 Dec 31 10:23 app_swift.c
-rw-r--r-- 1 root root 15123 Dec 31 10:23 LICENSE
-rw-r--r-- 1 root root  3210 Jan  8 14:07 Makefile
-rw-r--r-- 1 root root  1701 Dec 31 10:23 README
-rw-r--r-- 1 root root  1072 Dec 31 10:23 swift.conf.sample

When I run the make command, it spews a whole bunch of errors and 
warnings.  The first few lines are:


gcc -I/opt/swift/include -I/usr/src/asterisk-11.6.0/include -g -Wall 
-fPIC -D_SWIFT_VER_6 -D_AST_VER_11   -c -o app_swift.o app_swift.c

In file included from app_swift.c:33:
/usr/src/asterisk-11.6.0/include/asterisk.h:21:33: error: 
asterisk/autoconfig.h: No such file or directory

In file included from /usr/src/asterisk-11.6.0/include/asterisk.h:27,
 from app_swift.c:33:
/usr/src/asterisk-11.6.0/include/asterisk/compat.h:97: error: expected

Please let me know what information I can provide, thanks for your 
patience with a newbie.


Brandon


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[asterisk-users] help with Cepstral 6 and Asterisk 11

[Brandon Coale]

Hello,

I recently purchased the Cepstral 6 text-to-speech engine (swift), and 
am now wondering if I should have bought something else.  I would like 
to use Cepstral text to speech like some people use the Festival() or 
Flite() applications.  For example, when I do a core show application 
flite at the CLI, flite is available to me:


localhost*CLI core show application flite
  -= Info about application 'Flite' =-
[Synopsis]
Say text to the user, using Flite TTS engine
[Description]
 Flite(text[,intkeys]): This will invoke the Flite TTS engine, send a 
text string,
get back the resulting waveform and play it to the user, allowing any 
given interrupt
keys to immediately terminate and return the value, or 'any' to allow 
any number back.



I would like to do the same thing with Cepstral, and am trying to use 
app_swift available from:


https://github.com/darrensessions/app_swift

However, I am not able to get app_swift to compile.  I am running 
Asterisk 11.6.0 and CentOS 6.4 64-bit.


I am wondering if anyone else out there has been able to get app_swift 
working with Asterisk 11 and could share any tricks they used to get it 
installed?


Brandon

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[asterisk-users] Help - DTMF relay in meetme is not reliable

[Rajib Deka]

Hello List,

I am facing some issue while passing DTMF (RFC2833 set globally in
sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two
users tries to pass DTMF simultaneously at the same time from their phones
only one DTMF is detected in asterisk and broadcasted to other users. Other
DTMF lost somewhere. We have tested only with sip phones.

Can someone help me with this, or is there any configuration option that
can resolve this problem? I want asterisk receive the DTMFs send at the
same time and to pass those either by queuing them or by some other means.
We can not use confbridge at this moment as we have developed the
application on meetme. Please help!

Regards,
-- 
Rajib Deka
Sr. Programmer
Siemens Ltd.
Chennai, India
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[asterisk-users] Help with decyphering DND status

[James B. Byrne]

Arch x86_64
OS CentOS-6.4 (freepbx)
Asterisk 11.4
FreePBX 2.11.0.4

Snom870 with FW-8.7.4.8


What I am attempting to do is to set a different background colour for
the BLF vkeys when a station is set to DND.  This is supposedly
accomplished through this setting in the phones provisioning file:

vkey_blue perm=RW
DND
Blue.on
Blue.pickup
Blue.park
Blue.message
/vkey_blue

However, this does not work.  What instead works when DND is set is this:

vkey_blue perm=RW
CONNECTED
Blue.on
Blue.pickup
Blue.park
Blue.message
/vkey_blue

Which makes no sense to me.  However, I infer that somewhere in the
bowels of Asterisk something is set for DND which the Snom interprets
as CONNECTED instead.  It is what this something is and how it is set
that I am trying to understand.

To further this I am trying to discover is exactly what is sent to the
other stations by asterisk when DND is enabled for a station.  Short
of installing wireshark is there any other way to see exactly what
asterisk is sending to the phone?

When I look at the SIP trace logs on the handset when switch dnd on
and off on another handset I see this sort of thing:

Received from udp:192.168.6.9:5060 at 16/7/2013 11:56:14:422 (693 bytes):

NOTIFY sip:41720@192.168.6.120:3072;line=d241fk25 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK56d306e4;rport
Max-Forwards: 70
From: sip:41712@192.168.6.9;;tag=as149ada79
To: sip:41720@192.168.6.9;tag=tyybvtkyiy
Contact: sip:41712@192.168.6.9:5060
Call-ID: 455FE451B390010B19F8CAE0BE485B25-ju0bdbwv885i
CSeq: 191 NOTIFY
User-Agent: FPBX-2.11.0(11.4.0)
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 206

?xml version=1.0?
dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=89
state=full entity=sip:41712@192.168.6.9
dialog id=41712
stateconfirmed/state
/dialog
/dialog-info

Sent to udp:192.168.6.9:5060 at 16/7/2013 11:56:14:426 (300 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK56d306e4;rport=5060
From: sip:41712@192.168.6.9;;tag=as149ada79
To: sip:41720@192.168.6.9;tag=tyybvtkyiy
Call-ID: 455FE451B390010B19F8CAE0BE485B25-ju0bdbwv885i
CSeq: 191 NOTIFY
User-Agent: snom870/8.7.4.8
Content-Length: 0

Received from udp:192.168.6.9:5060 at 16/7/2013 11:56:16:051 (693 bytes):

NOTIFY sip:41720@192.168.6.120:3072;line=d241fk25 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK49b47181;rport
Max-Forwards: 70
From: sip:41712@192.168.6.9;;tag=as149ada79
To: sip:41720@192.168.6.9;tag=tyybvtkyiy
Contact: sip:41712@192.168.6.9:5060
Call-ID: 455FE451B390010B19F8CAE0BE485B25-ju0bdbwv885i
CSeq: 192 NOTIFY
User-Agent: FPBX-2.11.0(11.4.0)
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 206

?xml version=1.0?
dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=90
state=full entity=sip:41712@192.168.6.9
dialog id=41712
stateconfirmed/state
/dialog
/dialog-info

Sent to udp:192.168.6.9:5060 at 16/7/2013 11:56:16:055 (300 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK49b47181;rport=5060
From: sip:41712@192.168.6.9;;tag=as149ada79
To: sip:41720@192.168.6.9;tag=tyybvtkyiy
Call-ID: 455FE451B390010B19F8CAE0BE485B25-ju0bdbwv885i
CSeq: 192 NOTIFY
User-Agent: snom870/8.7.4.8
Content-Length: 0

Sent to udp:192.168.6.9:5060 at 16/7/2013 11:56:16:672 (483 bytes):

SUBSCRIBE sip:41710@192.168.6.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.120:3072;branch=z9hG4bK-f20yldu080fk;rport
From: sip:41720@192.168.6.9;tag=se3w15c5fb
To: sip:41710@192.168.6.9;;tag=as6723ebb5
Call-ID: 455FE4519A6B01B916F8CAE0BE485B25-x08c2euu13zz
CSeq: 63 SUBSCRIBE
Max-Forwards: 70
User-Agent: snom870/8.7.4.8
Contact: sip:41720@192.168.6.120:3072;line=d241fk25;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
Expires: 3600
Content-Length: 0

Received from udp:192.168.6.9:5060 at 16/7/2013 11:56:16:674 (529 bytes):

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.6.120:3072;branch=z9hG4bK-f20yldu080fk;received=192.168.6.120;rport=3072
From: sip:41720@192.168.6.9;tag=se3w15c5fb
To: sip:41710@192.168.6.9;;tag=as6723ebb5
Call-ID: 455FE4519A6B01B916F8CAE0BE485B25-x08c2euu13zz
CSeq: 63 SUBSCRIBE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=510db654
Content-Length: 0

Sent to udp:192.168.6.9:5060 at 16/7/2013 11:56:16:680 (648 bytes):

SUBSCRIBE sip:41710@192.168.6.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.120:3072;branch=z9hG4bK-6lft658u13gn;rport
From: sip:41720@192.168.6.9;tag=se3w15c5fb
To: sip:41710@192.168.6.9;;tag=as6723ebb5
Call-ID: 455FE4519A6B01B916F8CAE0BE485B25-x08c2euu13zz
CSeq: 64 SUBSCRIBE
Max-Forwards: 70
User-Agent: snom870/8.7.4.8
Contact: sip:41720@192.168.6.120:3072;line=d241fk25;reg-id=1
Event: dialog
Accept: application/dialog-info+xml
Authorization: Digest

[asterisk-users] Help me understand these log messages

[Chris Gentle]

OK, I need a bit of help here.  I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules drop for a day or so.
When I logged in today, I found messages like the ones below on my
asterisk console.  Obviously somebody was trying to take advantage of
my carelessness.  So can someone explain what would cause these types
of messages to show up on my console?

I understand that my iptables would have stopped this but I'm just
trying to understand more about the problem.  What other settings
might have stopped this?  Fail2ban was running but there were no
failed registration type messages that would have triggered it.

[May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '00972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '000972595595767' rejected
because extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '011972595595767' rejected
because extension not found in context 'default'.
snip


--
Chris

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Re: [asterisk-users] Help me understand these log messages

[Yves A.]

... an anonyous (not registerted) sip user from 188.161.238.232 was 
trying to initiate a call to

9725955 and so on...
you could enable sip tracing to get more information.

maybe you should change the 'allowguest' option in sip.conf..?

regards,
yves

Am 31.05.2013 23:57, schrieb Chris Gentle:

OK, I need a bit of help here.  I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules drop for a day or so.
When I logged in today, I found messages like the ones below on my
asterisk console.  Obviously somebody was trying to take advantage of
my carelessness.  So can someone explain what would cause these types
of messages to show up on my console?

I understand that my iptables would have stopped this but I'm just
trying to understand more about the problem.  What other settings
might have stopped this?  Fail2ban was running but there were no
failed registration type messages that would have triggered it.

[May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '00972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '000972595595767' rejected
because extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '011972595595767' rejected
because extension not found in context 'default'.
snip


--
Chris

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Re: [asterisk-users] Help me understand these log messages

[Alec Davis]

Top of sip.conf
quote
;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
;   understand the risks of installing Asterisk with the sample
;   configuration. If your Asterisk is installed on a public
;   IP address connected to the Internet, you will want to learn
;   about the various security settings BEFORE you start
;   Asterisk.
;
;   Especially note the following settings:
;   - allowguest (default enabled)
;   - permit/deny/acl - IP address filters
;   - contactpermit/contactdeny/contactacl - IP address filters
for registrations
;   - context - Which set of services you offer various users
;
/quote

In other words: allowguest = yes, is the default.
But in trunk the context for guest is [public], yours started in the
[default] context

Alec 

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
 Chris Gentle
 Sent: Saturday, 1 June 2013 9:57 a.m.
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Help me understand these log messages
 
 OK, I need a bit of help here.  I'm configuring a new 
 Asterisk 11 system and I accidentally let my firewall rules 
 drop for a day or so.
 When I logged in today, I found messages like the ones below 
 on my asterisk console.  Obviously somebody was trying to 
 take advantage of my carelessness.  So can someone explain 
 what would cause these types of messages to show up on my console?
 
 I understand that my iptables would have stopped this but I'm 
 just trying to understand more about the problem.  What other 
 settings might have stopped this?  Fail2ban was running but 
 there were no failed registration type messages that would 
 have triggered it.
 
 [May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '972595595767' rejected 
 because extension not found in context 'default'.
 [May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
 RTP CoS mark 5
 [May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '00972595595767' 
 rejected because extension not found in context 'default'.
 [May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
 RTP CoS mark 5
 [May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '000972595595767' 
 rejected because extension not found in context 'default'.
 [May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
 RTP CoS mark 5
 [May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '011972595595767' 
 rejected because extension not found in context 'default'.
 snip
 
 
 --
 Chris
 
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Re: [asterisk-users] Help me understand these log messages

[Chris Gentle]

OK, I understand now.  I didn't realize allowguest was on by default.
I guess I should read more closely.  Thanks!

On Fri, May 31, 2013 at 5:15 PM, Yves A. yves...@gmx.de wrote:
 ... an anonyous (not registerted) sip user from 188.161.238.232 was trying
 to initiate a call to
 9725955 and so on...
 you could enable sip tracing to get more information.

 maybe you should change the 'allowguest' option in sip.conf..?

 regards,
 yves

 Am 31.05.2013 23:57, schrieb Chris Gentle:

 OK, I need a bit of help here.  I'm configuring a new Asterisk 11
 system and I accidentally let my firewall rules drop for a day or so.
 When I logged in today, I found messages like the ones below on my
 asterisk console.  Obviously somebody was trying to take advantage of
 my carelessness.  So can someone explain what would cause these types
 of messages to show up on my console?

 I understand that my iptables would have stopped this but I'm just
 trying to understand more about the problem.  What other settings
 might have stopped this?  Fail2ban was running but there were no
 failed registration type messages that would have triggered it.

 [May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '972595595767' rejected because
 extension not found in context 'default'.
 [May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
 RTP CoS mark 5
 [May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '00972595595767' rejected because
 extension not found in context 'default'.
 [May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
 RTP CoS mark 5
 [May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '000972595595767' rejected
 because extension not found in context 'default'.
 [May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
 RTP CoS mark 5
 [May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '011972595595767' rejected
 because extension not found in context 'default'.
 snip


 --
 Chris

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-- 
Chris

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