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John M | 25 May 2013 10:01
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Re: dbh:query - insert id

Hi Daniel,

Thanks for your description, it is much appreciated. :-)

5 word one liners from people too lazy to explain properly would really be best if they didn't reply at all.

Cheers, thanks again.

-Jm



-----Original Message-----
From: Daniel Ivanov <sertys-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org>
To: FreeSWITCH Users Help <freeswitch-users-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b@public.gmane.org>
Sent: Sat, May 25, 2013 5:57 pm
Subject: Re: [Freeswitch-users] dbh:query - insert id

It is true that the luasql driver is overly basic and poorly documented . Unfortunately mysql doesn't support RETURNING clause like pgsql and oracle. You should however try SELECT LAST_INSERT_ID(); right after the insert query. I cannot guarantee it works due to the unknown nature(to me that is) of the luasql transaction handling, but it should keep a transaction open for as long as a db handler lives.
On May 25, 2013 7:03 AM, "John M" <j_mj-YDxpq3io04c@public.gmane.org> wrote:
Hi Seven Du,

I'd really like to know if this is possible too, couldn't find it documented anywhere.

Instead of being cryptic, if you know the answer won't you please help by explaining what the RETURNING clause is and how to use it?

Does it somehow return mysql_insert_id()?

How should we use it.

You help is invaluable and is contributing to the freeswitch community.

-Jm


-----Original Message-----
From: Seven Du <dujinfang <at> gmail.com>
To: FreeSWITCH Users Help <freeswitch-users-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b@public.gmane.org>
Sent: Sat, May 25, 2013 12:52 pm
Subject: Re: [Freeswitch-users] dbh:query - insert id

Maybe try the RETURNING clause ?
On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote:
Hello All

How to get the id value after insert a record a record using dbh:query

table_a - columns.

id - auto increment
field1
field2


dbh:query("insert into table_a ( field1,field2) values ('11','Test')")


After insert how to get the table_a - id value for the inserted record?

Thanks
Lloyd

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xiaofengcanyuexp-9Onoh4P/yGk@public.gmane.org | 25 May 2013 04:22
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How fs handle the sip-T message?

Hi, support team

I know FS can pass out SS7 info via properity header.
While, how it handles an incoming SIP-T message with SS7 info encapsulated in "content-type". Will it
parse it or discard it?
	

Thanks
Windy
-------------------
2013-05-25

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xiaofengcanyuexp-9Onoh4P/yGk@public.gmane.org | 25 May 2013 03:18
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How to attche custom "content-type" in Freeswitch SIP message

Dear support,

I''m trying to encapsulate my private "application/isup" in the SIP Msg. Normally, it should like below example.

It firstly addresses   "Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR". And then can
write private "content-type" like "application/sdp".
Now I can see the application/sdp is encapsulated via variable "switch_r_sdp".
Is there anyway to encapsulated other customized "content-type"? 

Appreciate to get your reply.
---------------------------------------------------------------------------------------------------------------
Here is an example of SIP message which encapsulated "applicaiton/sdp" and "application/isup".

INVITE sip: 87896677@...;user=phone;SIP/2.0
From: "Caller"<sip:678923001@...>
To: <sip:87896677@...>;user=phone
Call-ID: QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP@...
Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR
MIME-Version: 1.0
Content-Length: 433
--QRLVLNKeKxWDHAuwlEkR
Content-Type: application/sdp
User-Agent: ENSR2.5.46.6-IS2-RMRG36-RG20-CPO487
Content-Length: 142

v=0
o=- 1706944438 1706944438 IN IP4 192.168.1.105
s=ENSResip
t=0 0
m=audio 6793 RTP/AVP 0
a=rtpmap:0 PCMU/8000
--QRLVLNKeKxWDHAuwlEkR
Content-Type: application/isup; version=ansi;base=ansi00
Content-Disposition: signal; handling=optional

01 00 60 01 0a 03 05 0b 02 c0 90 06 03 10 78 98 66 77 0a 07 83 13 76 98 32 00 0f 00 
--QRLVLNKeKxWDHAuwlEkR--

	

Thanks
-------------------
2013-05-25

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Lloyd Aloysius | 25 May 2013 02:14
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dbh:query - insert id

Hello All

How to get the id value after insert a record a record using dbh:query

table_a - columns.

id - auto increment
field1
field2


dbh:query("insert into table_a ( field1,field2) values ('11','Test')")


After insert how to get the table_a - id value for the inserted record?

Thanks
Lloyd
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mbo | 24 May 2013 20:51
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WebRTC

I've seen there have been a couple of discussions about WebRTC support earlier this year. Are there any news
on that?

Thanks

Markus
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Guillermo Ruiz Camauer | 24 May 2013 19:25
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Assertion when Transcoding

I am getting the following assertion when attempting to record a G.729 call:
 
freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed.
 
The system has a Sangoma D100 card installed.  The call is established using the G.729 codec, and after playing some prompts, I am attempting to record an answer:
 
freeswitch-NcCO+2TxUIY@public.gmane.org8.9.151:5000 <at> internal> show codecs
type,name,ikey
codec,ADPCM (IMA),mod_spandsp
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_spandsp
codec,G.726 16k,mod_spandsp
codec,G.726 16k (AAL2),mod_spandsp
codec,G.726 24k,mod_spandsp
codec,G.726 24k (AAL2),mod_spandsp
codec,G.726 32k,mod_spandsp
codec,G.726 32k (AAL2),mod_spandsp
codec,G.726 40k,mod_spandsp
codec,G.726 40k (AAL2),mod_spandsp
codec,GSM,mod_spandsp
codec,LPC-10,mod_spandsp
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Sangoma G729,mod_sangoma_codec
18 total.
.
.
.
2013-05-24 12:33:31.071446 [ALERT] switch_core_session.c:2739 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE]
2013-05-24 12:33:31.071446 [ALERT] switch_ivr.c:650 sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC]
2013-05-24 12:33:31.111446 [ALERT] switch_core_session.c:2724 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC]
2013-05-24 12:33:31.111446 [INFO] mod_native_file.c:94 Opening File [/mnt/TEMP/beep-7.G729] 8000hz
2013-05-24 12:33:31.231447 [ALERT] switch_core_session.c:2739 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE]
2013-05-24 12:33:31.231447 [ALERT] switch_ivr.c:650 sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC]
2013-05-24 12:33:31.271451 [ALERT] switch_core_session.c:2724 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC]
2013-05-24 12:33:31.291450 [ALERT] switch_core_io.c:446 sofia/crossfonetrunk/52764439 receive message [TRANSCODING_NECESSARY]
freeswitch-Q0ErXNX1Rub9dkYlTf3LZg@public.gmane.org:5000 <at> internal> freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed.

FreeSwitch dies after this.  Note that this does not happen if I restrict everything to G711 (ULAW).
 
Any thoughts?
--
Guillermo Ruiz Camauer
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mbo | 24 May 2013 17:44
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Freeswitch versioning / tagging /releasemanagement

On the wiki I can see that the latest stable freeswitch release is 1.2.9. If I look into git (git tag -l -n1), I
can see that there are many tags with a higher version, but older tag date.

v1.3.0          1.3.0 release
v1.3.1          Tagging version 1.3.1
v1.3.10         tag v1.3.10
v1.3.11         tag v1.3.11
v1.3.12         Retag v1.3.12
v1.3.13         tag v1.3.13
v1.3.14         tag v1.3.14
v1.3.15         tag v1.3.15
v1.3.16         tag v1.3.16
v1.3.17-final   tag v1.3.17-final
v1.3.2          Tagging version 1.3.2
v1.3.3          Tagging 1.3.3
v1.3.4          release v1.3.4
v1.3.5          release v1.3.5
v1.3.6          tag v1.3.6
v1.3.7          tag v1.3.7
v1.3.8          tag v1.3.8
v1.3.9          tag v1.3.9
v1.5.0          tag v1.5.0

What about those tags? How is the release management organized? 

Thanks

Markus
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juned | 24 May 2013 08:59
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voicemail is not working

Hi All,

I am newbie to FS. so as a startup i have installed FS in mu local system to
test out the basic functionality and features. so i have registered default
users 1000 and 1001 in softphone( twinkle
<http://mfnboer.home.xs4all.nl/twinkle/>  ). registration was successful and
calls was also fine but when i tried to check voicemail then it didn't
worked.

what i did to test it out voicemail is, did call to 1001 and let it rang so
after 30 second if no answer is there then voicemail will be activated but
calls are released after 30 seconds.

As per documentation i came to know that by default voicemail is activated
in default extensions is it so ? or i am missing something.

Please point me to right direction, i want to have such a dialplan in which
user can leave voicemail in cases of busy,unavailable and not answering.

Thanks & Regards
Juned 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/voicemail-is-not-working-tp7591043.html
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Clinton Goudie-Nice | 24 May 2013 01:22
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ESL using bridge app doesn't return which gateway was used

When you make a bridge command using esl, where you specify multiple gateways or sip dials separated by or bars, you can't figure out which gateway was used.

For example, if you bridge to something like this:

sofia/gateway/SBC-GW2/+18019600000|sofia/gateway/SBC-GW1/+18019600000

The call could be bridged to either GW2 or GW1. 

When the CHANNEL_BRIDGE event is returned, you can see the original string in variable_current_application_data, and you may be able to infer the destination based on IP address, but nothing clearly says what gateway is used.

If you turn on the all events firehose, you can see the CHANNEL_CREATE event come over the socket, and it does contain variable_sip_gateway_name with the actual name of the gateway, however I can't devise a way to access that data using the org.freeswitch.esl.client library, and even if I could, I still don't want all events for this system.

Is it possible to get this information returned in any meaningful way through the ESL layer, either by an api command to query, or the setting of a variable that will give me back which gateway a bridge was performed through?

If none of that is possible, this sounds worthy of filing a bug to return the variable_sip_gateway_name in the CHANNEL_BRIDGE event.


Thanks for the help,


Clint



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Flavio Goncalves | 23 May 2013 22:38
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Strange issue with late negotiation

Hello, 

I'm having a really strange issue on billing related to late negotiation. The call end up hanged with no BYE.  

There is a dialer sending thousands of calls through and OpenSIPS to a termination based on FreeSwitch. I have a SIP call flow as below. Some times during the day, when the volume is high, the UAC drops some 200Ok  and FS send a reinvite in the middle of the Initial transaction.

----INVITE ---------------   Proxy --INVITE ---------------> FS
<----200 OK --------------   Proxy <---200 OK- ------------- FS
<--REINVITE---------------   Proxy <-REINVITE--------------- FS  
---481 leg does not exit->   Proxy ---481------------------> FS
<-- ACK (REINVITE)--------   Proxy <-ACK(REINVITE)---------- FS
----CANCEL --------------->  Proxy ---CANCEL --------------> FS 
<---200 Ok ----------------  Proxy <----200 Ok ------------- FS

FS is sending a reinvite before ACK comes from the client.

I have two questions:

1) Is it valid to send a reinvite in the middle of an existing transaction?

According to the RFC3261 Section 14. 

Note that a UAC MUST NOT initiate a new INVITE transaction within a dialog while another INVITE transaction is in progress in either direction. 1. If there is an ongoing INVITE client transaction, the TU MUST wait until the transaction reaches the completed or terminated state before initiating the new INVITE. 2. If there is an ongoing INVITE server transaction, the TU MUST wait until the transaction reaches the confirmed or terminated state before initiating the new INVITE.

But it is confusing because just below it says the opposite. 

However, a UA MAY initiate a regular transaction while an INVITE transaction is in progress. A UA MAY also initiate an INVITE transaction while a regular transaction is in progress.

2. Shouldn't FS send a BYE after sending a 200Ok and not receiving the ACK?

If a UAS generates a 2xx response and never receives an ACK, it SHOULD generate a BYE to terminate the dialog.

Best regards, 

Flavio E. Goncalves

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Nathan Neulinger | 24 May 2013 14:43
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Re: Dialplan not executing on continue_on_fail=true

I don't think that's going to do what you want... (May be wrong.)

I think that continue_on_fail is only going to apply to the rules for the received call on this extension,
not the 
received call on the outgoing leg.

i.e. there are no dialplan rules in effect for the outgoing call that you initiated, and that's where the
failure is 
occurring. For these dialplan rules, I think the only failure would be if your IVR (I assume that's was
ash.pl is) 
didn't answer.

Like I said, not certain of this, maybe some else can chime in, but I think you're going to have to handle that
failure 
as a part of your originate on the outbound call. Something like putting

	originate {api_hangup_hook=perl hook.pl}sofia/.....

Where you cause the call to take place.

-- Nathan

On 05/24/2013 07:37 AM, Ashish gautam wrote:
> I am generating an outgoing call through mod_event_socket and then transferring it to this dialplan.
>
> On Fri, May 24, 2013 at 5:57 PM, Nathan Neulinger <nneul@...
<mailto:nneul@...>> wrote:
>
>     I may be misunderstanding - but where are you causing it to ring a device?
>
>     You've told it to internally answer the call, and then not do anything.  There's no bridging to an actual extension.
>
>     Only thing I see that would happen is it running perl/ash.pl <http://ash.pl>, unclear if it would in term execute
>     hook.pl <http://hook.pl> when that script finished (I don't know what that behavior is expected to be).
>
>     -- Nathan
>
>
>     On 05/24/2013 07:17 AM, Ashish gautam wrote:
>
>         Hi,
>
>         I have a dialplan as follows:
>
>         <include>
>         <extension name="public_did">
>         <condition field="destination_number" expression="^(47673501)$">
>         <action application="answer"/>
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="set" data="api_hangup_hook=perl hook.pl <http://hook.pl> <http://hook.pl>"/>
>
>         <action application="set" data="session_in_hangup_hook=__true"/>
>         <action application="perl" data="perl/ash.pl <http://ash.pl> <http://ash.pl>"/>
>         </condition>
>         </extension>
>         </include>
>
>         when the called party does not pick up the phone or is busy, the dialplan does not proceed and hook.pl
>         <http://hook.pl> <http://hook.pl>
>
>         does not get executed.
>
>         Please help
>         --
>         Ashish Gautam
>
>         IVR Developer
>
>         Nucleus Microsystems (Pvt.) Ltd.
>
>
>
>     --
>     ------------------------------__------------------------------
>     Nathan Neulinger nneul@... <mailto:nneul@...>
>     Missouri S&T Information Technology    (573) 612-1412
>     System Administrator - Architect
>
>
>
>
> --
> Ashish Gautam
>
> IVR Developer
>
> Nucleus Microsystems (Pvt.) Ltd.
>
> Ph. 011 47574758

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul@...
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect

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Gmane