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Re: [Freeswitch-users] Calling stored procedure in freeswitch database or using luasql
by Shahzad_Sab 24 Oct '13

24 Oct '13

i am also trying to do and if u got any idea do reply me too regards shahzad bhatti -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Calling-stored-procedure-in-f… Sent from the freeswitch-users mailing list archive at Nabble.com.

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bind_digit_action constantly reporting syntax error
by ik 24 Oct '13

24 Oct '13

Hello, I'ved created the following dialplan (fs v1.2.14) : <context name="bridging_capture"> <extension name="beidgin"> <condition field="destination_number" expression="(^.*$)" > <action application="set" data="hangup_after_bridge= false" /> <action application="digit_action_set_realm" data="bridging_capture" /> <action application="set" data="bridge_pre_execute_bleg_app=bind_digit_action"/> <action application="set" data="bridge_pre_execute_bleg_data=bridging_capture"/>  <action application="bind_digit_action" data="bridging_capture,~200\*[0-9]{3,10},exec:execute_extension,digits,self,peer"/> <action application="bridge" data="user/2002" />  </condition> </extension> <extension name="digits">  <condition field="destination_number" expression="^.*$"> <action application="hangup" /> </condition> </extension> </context> bind_digit_action constantly reporting Syntax Error, USAGE <realm>,<digits|~regex>,< string>[,<value>][,<dtmf target leg>][,<event target leg>] The original aim of the program is the capture the numbers after 200*, and bridge leg-a to the new number, but first disconnecting current leg-b. When using execute_extension with "exec_after_bridge_app=execute_extension "exec_after_bridge_arg=digits ${bridging_capture}" It makes leg-b constantly ring after the call hangup. So does the transfer action that is commented out. So what am I doing wrong here ? Thanks, Ido

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Re: [Freeswitch-users] codec transcodation not working
by neeraj.p 24 Oct '13

24 Oct '13

Here is the full log when leg A calls leg B http://pastebin.com/0JAZQpe7 Here is the log when leg B calls leg A http://pastebin.com/8Q5abKEC I can see different errors in these two cases . Regards, Neeraj

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Forwarding SIP sessions
by Alex Lake 24 Oct '13

24 Oct '13

Is it possible to get Freeswitch to send a re-invite message in order to do the equivalent of "I'm not accepting this call, try xx.xx.xx.xx"?

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issue of cdr_mongodb
by erhuabushuo 24 Oct '13

24 Oct '13

established connection, after finished call, when it try to insert cdr to mongodb, I got this error '2013-10-23 17:21:29.745772 [ERR] mod_cdr_mongodb.c:393 mongo_insert: (error code 15)'

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mod_shout in Windows distribution?
by Rafal Gwizdala 24 Oct '13

24 Oct '13

Hi, is there a reason why mod_shout is not included with Windows binary distribution? Best regards RG

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Parsing from sip header
by Miha 24 Oct '13

24 Oct '13

Hi, I am sending to FS some attribute in header which I have added on proxy side. How can I parse this header, to get this data and send it further? I can not see it in varibles which are printed with info application. I added: Moved: 38618108751. BR, Miha

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Re: [Freeswitch-users] codec transcodation not working
by neeraj.p 24 Oct '13

24 Oct '13

Hey Steven Here is the result of command - sofia status profile internal I can see speex in the list. freeswitch@127.0.0.1:5222@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 172.16.138.74,172.16.138.74 Dialplan XML Context public Challenge Realm auto_from RTP-IP 172.16.138.74 SIP-IP 172.16.138.74 URL sip:mod_sofia@172.16.138.74:5060 BIND-URL sip:mod_sofia@172.16.138.74:5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMU,GSM,PCMA,speex@8000h@20i CODECS OUT PCMU,GSM,PCMA,speex@8000h@20i TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1 --Neeraj

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Individual party recording
by Siri MM 24 Oct '13

24 Oct '13

Hi, Is it possible for me to record each party of a call individually? If I were to transfer the call a few times to different extensions, I would like store each party's conversation separately - is this feasible? Thanks!

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loopback_bowout
by Peter Steinbach 24 Oct '13

24 Oct '13

Hello, I have a strange behaviour with forwarding a call from an external gateway to another number on the same gateway. * when I forward a call from an internal number to an external number via loopback, I firstly get 4 channels. After answering the call, I have 2 channels remaining and audio is fine * when I forward a call from an*_*external_ number to an external number via loopback, I firstly get 4 channels. After answering the call, I still have 4 channels remaining and audio is _not_ there. Freeswitch is dialling via "external" profile via port 5080. I also tried to set "loopback_bowout=true", but this did not change anything. The dialstring is: <action data="{sip_from_uri=0610xxxxxxx@sip.domain.de}sofia/gateway/SIP-68xxxxx/017xxxxx@sip.domain.de" application="bridge" /> The dialplan for both scenarios is the same. Anybody had the same issue and knows how to overcome this? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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