Re: [Freeswitch-users] How do I know the destination profile name?

[Yehavi Bourvine]

Thanks Mike! However, this doesn't fully solve my problem. When using
sofia_contact() indeed it works ok with finding the destination's profile.
However, it breaks the BLFs...

When calling *sofia/sip_profile/local-user%local-do**main* the BLF works ok.
When calling 
sofia_contact(*sofia/sip_profile/local-u...@local-domain*)
BLF doesn't work (nothing is sent to the watching phone).

Any more clues???

 Thanks! __Yehavi:

2009/11/20 Michael Jerris 

> check out sofia_contact function.  If you use this in combination with
> binding profiles together so they are one table I think this should work
> right.
>
> Mike
>
> On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote:
>
> > Brian West wrote:
> >>
> >> Why do you need to know the destination profile like that?  You get to
> >> pick that on your own so you should already know that before hand.
> >>
> >>
> >> /b
> >>
> >> On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote:
> >>
> >>
> >>> Hi
> >>> We have more then one profile. To make a call I have to enter : bridge
> >>> sofia/profile/num...@ip
> >>> The problem is when I use : "${use_profile}" I am getting the caller
> >>> profile, and I need the destination profile.
> >>>
> >>> How do I get this information?
> >>>
> >>
>  > Thanks for your answer.
> >
> > The problem is when I call to that number that the phone hook to other
> server, I cannot make the call.
> > Is there is a variable that can tell me the destination profile?
> > Lets say the other profile called "ph1" I have to dial
> > sofia/ph1/xx...@host to make the call. Is there other way to do that?
>
>
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[Freeswitch-users] ATA that supports TLS/SRTP w FS

[Mark Campbell-Smith]

HI All,

Has anyone got some recommendations on which ATA to buy that supports
TLS and SRTP?

Thanks!

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[Freeswitch-users] IP1001 Setup

[David V. Fansler]

-Original Message- From: Michael Collins Sent: Nov 21, 2009 4:51 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated 
On Sat, Nov 21, 2009 at 4:10 AM, Dave Stevenson  wrote:
Sorry for the extended forum thread on this subject - This really IS thelast post !I have now got the ATA to work without the dialplan fix provided by Michael.After I'd implemented the "fix", I had more of an idea of what the problemwas and was better able to go through the Polycom VPA-11 setup screensthrough its web interface to see if there were any options that might havehad a bearing on the problem.Under Configuration   VoIP       Non-Line Config           General Parameters               VoIP GeneralThere is an option to "Append UserId" - the Default Value is Yes.That was where the "extra characters" were coming from.Setting this option to No, makes the ATA behave more as expected,RegardsDave
Nice work! I can understand why it was easier to look for this AFTER we did the band-aid solution and not before. :)You next task is to visit http://wiki.freeswitch.org and sign up. Please add the setup details for this device. I already created stub to get you going:http://wiki.freeswitch.org/wiki/Interop_List#PluscomThanks!-MC

David V. Fansler
S/V Annabelle
www.dv-fansler.com
dfans...@dv-fansler.com

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Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated

[Michael Collins]

On Sat, Nov 21, 2009 at 4:10 AM, Dave Stevenson
wrote:

> Sorry for the extended forum thread on this subject - This really IS the
> last post !
>
> I have now got the ATA to work without the dialplan fix provided by
> Michael.
>
> After I'd implemented the "fix", I had more of an idea of what the problem
> was and was better able to go through the Polycom VPA-11 setup screens
> through its web interface to see if there were any options that might have
> had a bearing on the problem.
>
> Under Configuration
>VoIP
>Non-Line Config
>General Parameters
>VoIP General
>
> There is an option to "Append UserId" - the Default Value is Yes.
>
> That was where the "extra characters" were coming from.
>
> Setting this option to No, makes the ATA behave more as expected,
>
> Regards
> Dave
>
> Nice work! I can understand why it was easier to look for this AFTER we did
the band-aid solution and not before. :)

You next task is to visit http://wiki.freeswitch.org and sign up. Please add
the setup details for this device. I already created stub to get you going:
http://wiki.freeswitch.org/wiki/Interop_List#Pluscom

Thanks!
-MC
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Re: [Freeswitch-users] tcp call misses sip message

[RobertT]

Yep, I use proxy media. First it started with 1.0.4 release, then I've
updated a week or two ago with the latest svn trunk, not sure what was the
rev number.
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[Freeswitch-users] Help Freeswitch with Voipuser Gateway

[Sam Abekah-Mensah]

I need help as I cannot receive calls through VOIPUSER. This is a 
learning setup Attached are my conf files. What is wrong with them ?  
When I dial from a landline I get a continuous beep.


Attached are my gateway and the conf file to transfer. Sopfia Status is 
my screen message. I can see a FAIL and cannot make head or tail of all 
that message. Hopefully anyone using voipuser or in fact any of you 
clever folks can make sense of this.


Thanks for your time.

2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate Buggy RFC2833 
Mode!
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec Compare 
[PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec Compare 
[PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec 
sofia/external/nob...@213.166.5.133 PCMA/8000 20 ms 160 samples
2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 
101
2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455 
(sofia/external/nob...@213.166.5.133) State Change CS_NEW -> CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send signal 
sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 
(sofia/external/nob...@213.166.5.133) Running State Change CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 
(sofia/external/nob...@213.166.5.133) State INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83 
sofia/external/nob...@213.166.5.133 SOFIA INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111 
(sofia/external/nob...@213.166.5.133) State Change CS_INIT -> CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send signal 
sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481 
(sofia/external/nob...@213.166.5.133) State INIT going to sleep
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398 
(sofia/external/nob...@213.166.5.133) Running State Change CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484 
(sofia/external/nob...@213.166.5.133) State ROUTING
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130 
sofia/external/nob...@213.166.5.133 SOFIA ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78 
sofia/external/nob...@213.166.5.133 Standard ROUTING
2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315 Processing 
anonymous->abeka in context public
Dialplan: sofia/external/nob...@213.166.5.133 parsing [public->unloop] 
continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (PASS) [unloop] 
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) [unloop] 
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing [public->outside_call] 
continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Absolute Condition [outside_call]
Dialplan: sofia/external/nob...@213.166.5.133 Action set(outside_call=true)
Dialplan: sofia/external/nob...@213.166.5.133 parsing [public->call_debug] 
continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) [call_debug] 
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/nob...@213.166.5.133 parsing 
[public->public_extensions] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) [public_extensions] 
destination_number(abeka) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing [public->public_did] 
continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) [public_did] 
destination_number(abeka) =~ /^(5551212)$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing 
[public->s...@sip.voipuser.org] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) 
[...@sip.voipuser.org] destination_number(abeka) =~ /08715042951/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing 
[public->inbound-ab...@sip.voipuser.org]] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL) 
[inbound-ab...@sip.voipuser.org]] destination_number(abeka) =~ /[08444846450]/ 
break=on-false
2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:114 
(sofia/external/nob...@213.166.5.133) State Change CS_ROUTING -> CS_EXECUTE
2009-11-21 22:07:15.704513 [DEBUG] switch_core_session.c:932 Send signal 
sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:484 
(sofia/external/nob...@213.166.5.133) State ROUTING going to sleep
2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:398 
(sofia/external/nob...@213.166.5.133) Running State Change CS_EXECUTE
2009-11-21 22:07:15.704513 [DEBUG] switch_core_state_machine.c:491 
(sofia/external/nob...@213.166.5.133) State EXECUTE
2009-11-21 22:07:15.706658 [DEBUG] mod_sofia.c:173 
sofia/external/no

Re: [Freeswitch-users] Problem while playing more than 10 voice files using playback

[Anthony Minessale]

cant you use the execute_complete events to tell when your playback is done
or var is set?


On Sat, Nov 21, 2009 at 3:22 AM, Thangappan.M wrote:

> Dear all,
>
>  I am in the process of implementing IVR using event outbound
> socket (async mode).
>  I have implemented using Perl language.
>
>   I did the following steps:
>=> Set the playback_delimiter variable
>=> Set the playback_sleep_val variable
>=> Set the event lock as true
> => Set the freeswitch ( my own)  variable as zero
> => Wait in the loop until the variable is been set as
> zero
>=>  Playback the voice files ( Here I combined the voice
> files with the delimiter value if more than one voice files are there)
>=> Set the freeswitch(my own) variable as true ( This is
> used to identify whether the voice files are played
>  successfully).
>=> Wait in the loop until the variable is been set as
> one.
>=> Set the Event lock as false
>
>=> Trying to get the DTMF digits ( Have a assurance
> that  all the voice files are played).
>
>The problem is,
>
>  The above steps are working fine when the voice file count is
> lesser than or equal to 10. After the voice files are played only the
> variable(my own freeswitch) is set. Based on the variable I am doing further
> things.
>
>  But when I tried to give the voice files count of more than 10
> the variable has been set while starting to play back the first voice file
> itself . Because of this I am not able to proceed further.
>
>   *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?*
>
> *NOTE*: I also referred mod_file_string documentation. In that they
> specified 128 files can be used to play back the voice files using
> playback_delimiter option.
>
> Please help me?
> Thanks in advance.
>
>
> --
> Regards,
> Thangappan.M
>
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Re: [Freeswitch-users] Using odbc in FS core

[Anthony Minessale]

we had the code slightly out of order, you should update to latest trunk for
the right version.

The test of 2 deletes is to see if your odbc driver will fail when trying to
execute 2 statements at once so I can properly
fail over to sqlite because transactions are mandatory for a database
running core in odbc.


On Sat, Nov 21, 2009 at 6:02 AM, Mike Tkachuk  wrote:

> Hello,
>
>  Looks like the issue is not in multi statements in one request.
>  Manually creating DB schema helped and everything started up.
>  I will continue testing
>
>  Also in code I see such construction:
>  > switch_cache_db_execute_sql(dbh, "begin;delete from channels where
> hostname='';delete from channels where hostname='';commit;", &err);
>  Anyone can explain why to do such delete twice and in transaction?
>
>  Thanks.
>
>
>
> Saturday, November 21, 2009 1:41:06 PM, you wrote:
>
> MT> Hello Folks,
>
> MT>  I'm interesting in completely moving away from sqlite and use
> MT>  postgresql everywhere including core ( switch_core.c )
>
> MT>  All other applications can use odbc without issues (sofia, limit,
> MT>  fifo etc), but as I see in core only sqlite3 supported.
>
> MT>  I correctly set 'core-db-dsn' parameter, but looks like the problem
> MT>  that latest psqlodbc_08_04_0100 don't support multiple statements in
> MT>  one request that is often used in switch_core_sqldb.c:
>
> >> sql = switch_mprintf(
> >>   "update channels set uuid='%q' where uuid='%q' and hostname='%q';"
> >>   "update calls set caller_uuid='%q' where caller_uuid='%q' and
> hostname='%q';"
> >>   "update calls set callee_uuid='%q' where callee_uuid='%q' and
> hostname='%q'",
> >>   switch_event_get_header_nil(event, "unique-id"),
> >>   ... SKIP ...
>
> MT>  So, does anyone have any clue how to us postgresql in the FS core?
>
> MT>  Thanks.
>
> MT> --
> MT> Mike Tkachuk
>
>
>
> --
> Mike Tkachuk
>
>
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Re: [Freeswitch-users] tcp call misses sip message

[Brian West]

Well since we aren't a proxy we wouldn't resend the one we receive...  
what svn rev and are you using proxy media?

/b

On Nov 21, 2009, at 7:28 AM, RobertT wrote:

> Attached is graphical representation of SIP message flow. You can  
> see that for some reason FS doesn't resend to callee an ACK message  
> recieved from caller.
>
> Regards, RobertT


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Re: [Freeswitch-users] tcp call misses sip message

[RobertT]

Attached is graphical representation of SIP message flow. You can see that
for some reason FS doesn't resend to callee an ACK message recieved from
caller.

Regards, RobertT
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Re: [Freeswitch-users] Problems with Voicemail

[Peter P GMX]

I installed all sounds from SVN, but

usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA

isn't there. I checked another, older installation and couldn't this
file either.

I think that freeswitch tries to build a sound path for the file to be
played, and some parts of the path are missing.
I expect it would play a recorded message at that time in
/usr/local/freeswitch/storage/voicemail/default/${domain} and the
defined format is "wav" not pcma.

I also set "storage_dir" explicitely in the voicemail configs,but this
also didn't help.

Best regards
Peter


Brian West schrieb:
> I'm going to venture to guess maybe the file was recorded in a  
> different codec and NOT pcma?
>
> /b
>
> On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote:
>
>   
>> 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening / 
>> usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA
>> 
>
>
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>   

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Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated

[Dave Stevenson]

Sorry for the extended forum thread on this subject - This really IS the 
last post !

I have now got the ATA to work without the dialplan fix provided by Michael.

After I'd implemented the "fix", I had more of an idea of what the problem 
was and was better able to go through the Polycom VPA-11 setup screens 
through its web interface to see if there were any options that might have 
had a bearing on the problem.

Under Configuration
VoIP
Non-Line Config
General Parameters
VoIP General

There is an option to "Append UserId" - the Default Value is Yes.

That was where the "extra characters" were coming from.

Setting this option to No, makes the ATA behave more as expected,

Regards
Dave






- Original Message - 
From: "Dave Stevenson" 
To: 
Sent: Saturday, November 21, 2009 1:20 AM
Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial 
Out,but can receive calls - Issue Closed


> OK,
>
> Just to close this one out, I've just spent some time on IRC and Michael
> Collins very quickly helped me get this sorted.
>
> I am using a cheap ATA which has a couple of "issues".
>
> I should probably have included it in the original post, but the ATA is a 
> :-
>
> Pluscom SIP VoIP ATA - Model VPA-11.
>
> Quote from the ATA Manual ...
>
> "If a default dial plan string is not required, the Default Dial Plan 
> String
> field on the General configuration page (Section 4.1.2)
> can be left empty, in which case the default dial pattern to accept all
> dialed digits will be incorporated.
> The default dial pattern, [0-9*]>#[0-9*].e[0-9*].ft4, is transparent to 
> the
> user and will not be displayed on the General
> configuration page"
>
> I had already (or so I thought) deleted the default ATA dialplan as I
> suspected that it was causing problems.
>
> As it turns out though, even with an apparently blank dialplan configured,
> the ATA was inserting some characters after the dialed number which were
> confusing the number handling in FreeSwitch.
>
> Michael quickly spotted the problem from my Pastebin dump and took about 
> 10
> seconds to come up with a fix !
>
> Adding the following section to the top of the default FreeSwitch dialplan
> strips these extra characters off the string that FreeSwitch sees
>
>
> 
> 
>  
>
>  
> 
> Thanks a lot Michael !
>
> regards
> Dave
>


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Re: [Freeswitch-users] Using odbc in FS core

[Mike Tkachuk]

Hello,

 Looks like the issue is not in multi statements in one request.
 Manually creating DB schema helped and everything started up.
 I will continue testing

 Also in code I see such construction:
 > switch_cache_db_execute_sql(dbh, "begin;delete from channels where 
 > hostname='';delete from channels where hostname='';commit;", &err);
 Anyone can explain why to do such delete twice and in transaction?

 Thanks.



Saturday, November 21, 2009 1:41:06 PM, you wrote:

MT> Hello Folks,

MT>  I'm interesting in completely moving away from sqlite and use
MT>  postgresql everywhere including core ( switch_core.c )

MT>  All other applications can use odbc without issues (sofia, limit,
MT>  fifo etc), but as I see in core only sqlite3 supported.

MT>  I correctly set 'core-db-dsn' parameter, but looks like the problem
MT>  that latest psqlodbc_08_04_0100 don't support multiple statements in
MT>  one request that is often used in switch_core_sqldb.c:

>> sql = switch_mprintf(
>>   "update channels set uuid='%q' where uuid='%q' and hostname='%q';"
>>   "update calls set caller_uuid='%q' where caller_uuid='%q' and 
>> hostname='%q';"
>>   "update calls set callee_uuid='%q' where callee_uuid='%q' and 
>> hostname='%q'",
>>   switch_event_get_header_nil(event, "unique-id"),
>>   ... SKIP ...

MT>  So, does anyone have any clue how to us postgresql in the FS core?

MT>  Thanks.

MT> --
MT> Mike Tkachuk



--
Mike Tkachuk


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[Freeswitch-users] Using odbc in FS core

[Mike Tkachuk]

Hello Folks,

 I'm interesting in completely moving away from sqlite and use
 postgresql everywhere including core ( switch_core.c )

 All other applications can use odbc without issues (sofia, limit,
 fifo etc), but as I see in core only sqlite3 supported.

 I correctly set 'core-db-dsn' parameter, but looks like the problem
 that latest psqlodbc_08_04_0100 don't support multiple statements in
 one request that is often used in switch_core_sqldb.c:

> sql = switch_mprintf(
>   "update channels set uuid='%q' where uuid='%q' and hostname='%q';"
>   "update calls set caller_uuid='%q' where caller_uuid='%q' and 
> hostname='%q';"
>   "update calls set callee_uuid='%q' where callee_uuid='%q' and 
> hostname='%q'",
>   switch_event_get_header_nil(event, "unique-id"),
>   ... SKIP ...

 So, does anyone have any clue how to us postgresql in the FS core?

 Thanks.

--
Mike Tkachuk


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Re: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app

[Artem Shiyanov]

Anthony,
>>As soon as you call uuid_bridge you are transferring both legs of the call
to bridge to each other.
>>This means your java app must exit so the channels can connect to each
other.

I didn't know that. Now my java app is exiting upon the onHangup() call so
everything has become "ok". Thank you much.
I'll add note to the wiki about this issue.

Artem



On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> Your "annoying behaviour" is the exact behavior you should be getting
> considering what you told FS to do.
>
> As soon as you call uuid_bridge you are transferring both legs of the call
> to bridge to each other.
> This means your java app must exit so the channels can connect to each
> other.
>
> remember that you hangup hook can be called when the channel is transferred
> not only when it hangs up.
> you have to test which is happening based on the input to your callback.
>
>
> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote:
>
>> Hi there!
>>
>> I've got annoying FS behavior:
>> There are 2 channels executing the same Java application (application
>> itself is an IVR). If I try to bridge them with uuid_bridged then both
>> channels are killed. Here is a log from FS console:
>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165
>> (sofia/internal/1...@192.168.147.130) State Change CS_EXECUTE ->
>> CS_HIBERNATE
>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called
>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done playing
>> file
>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done playing
>> file
>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal
>> sofia/internal/1...@192.168.147.130 [BREAK]
>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167
>> (sofia/internal/1...@master.agent.starpoundtech.net) State Change
>> CS_EXECUTE -> CS_HIBERNATE
>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called
>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2
>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output:
>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de
>>
>> freeswi...@localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG]
>> switch_core_session.c:933 Send signal
>> sofia/internal/1...@master.agent.starpoundtec
>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal
>> sofia/internal/1...@192.168.147.130 [BREAK]
>>
>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream
>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]
>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal
>> sofia/internal/1...@master.agent.starpoundtech.net [BREAK]
>>
>> (FS version is 1.0.4)
>>
>> Any thoughts?
>>
>>
>> Artem
>>
>>
>>
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>
>
> --
> Anthony Minessale II
>
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[Freeswitch-users] Problem while playing more than 10 voice files using playback

[Thangappan.M]

Dear all,

 I am in the process of implementing IVR using event outbound socket
(async mode).
 I have implemented using Perl language.

  I did the following steps:
   => Set the playback_delimiter variable
   => Set the playback_sleep_val variable
   => Set the event lock as true
=> Set the freeswitch ( my own)  variable as zero
=> Wait in the loop until the variable is been set as
zero
   =>  Playback the voice files ( Here I combined the voice
files with the delimiter value if more than one voice files are there)
   => Set the freeswitch(my own) variable as true ( This is
used to identify whether the voice files are played
 successfully).
   => Wait in the loop until the variable is been set as
one.
   => Set the Event lock as false

   => Trying to get the DTMF digits ( Have a assurance that
all the voice files are played).

   The problem is,

 The above steps are working fine when the voice file count is
lesser than or equal to 10. After the voice files are played only the
variable(my own freeswitch) is set. Based on the variable I am doing further
things.

 But when I tried to give the voice files count of more than 10
the variable has been set while starting to play back the first voice file
itself . Because of this I am not able to proceed further.

  *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?*

*NOTE*: I also referred mod_file_string documentation. In that they
specified 128 files can be used to play back the voice files using
playback_delimiter option.

Please help me?
Thanks in advance.


-- 
Regards,
Thangappan.M
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