- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] (no subject)
by Dave Bracken 27 Jul '11

27 Jul '11

Don’t waste your time!.. http://bamboo-nails.com/important.php?diID=13fu0

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Dailplan question?
by Dmitry Saratsky 27 Jul '11

27 Jul '11

Is there any way to bridge/transfer call to dialplan logic? Asterisk example: Local/XXXXXXXXXXX@outbound-allroutes For example, if I have 3 external trunks to sip providers and DISA service, I want to bridge the call automatically to the regex suitable gateway in dialplan I mean, how can I pass the call to known number (without specifying trunk/gateway, so the relevant gateway will be chosen by dialplan logic) Thanks, Dmitry.

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Dailplan question?
by Dmitry Saratsky 26 Jul '11

26 Jul '11

Is there any way to bridge/transfer call to dialplan logic? Asterisk example: Local/XXXXXXXXXXX@outbound-allroutes For example, if I have 3 external trunks to sip providers and DISA service, I want to bridge the call automatically to the regex suitable gateway in dialplan I mean, how can I pass the call to known number (without specifying trunk/gateway, so the relevant gateway will be chosen by dialplan logic) Thanks, Dmitry.

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Combining AND and ELSE in freeswitch conditions
by Alex Lake 26 Jul '11

26 Jul '11

Hello all, I'm a Freeswitch newbie struggling to get my head around the best way to do some TOD/BH routing. I've got a LUA script that will work out whether or not we're in a bank holiday, but I wanted to do the combinatorial logic within Freeswitch dial plan. Something like: if (Weekday AND (NOT BankHoliday) AND InWorkingHours) { Transfer to mainIVR } else { Transfer to voicemail } What is the most elegant way to achieve this? The Wiki pages don't appear to explain how one can do ANDs with ELSE, as the anti-action will only get executed if it's only the last condition that fails. Thanks!

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Cisco CUCM 6.1 and Freeswitch SIP trunk
by David Martin 26 Jul '11

26 Jul '11

All, Firstly apologies if this is not the correct place to ask, but I am having problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk (I only need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am getting the following sofia trace: http://pastebin.com/g9VmvxFe Can anyone point me in the right direction? I have also tried enabling MTP on the Cisco side, but this does not make any difference. Thanks in advance. David

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(no subject)
by Sanja Angelova 26 Jul '11

26 Jul '11

Dear all! I am new to FreeSwitch and I am trying to map a number for example 654321 to sip adress like this 1005(a)10.1.1.1 with configuring a local DNS server. When I write enum 654321 in fs_cli it gives me: Offered Routes: Order Pref Service Route ============================================================================== 10 100 E2U+sip sofia/${use_profile}/1005(a)10.1.1.1 and seems to be fine. But when I make a call from my soft phone, the FreeSwitch ends up my call and it gives me error 503 service unavailable. I have no idea what the problem is, can you please help me? Thank you in advance

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sipx conference API
by Jayakrishnanu 26 Jul '11

26 Jul '11

Hi Folks Is there any way to add a person in to existing conference? > I configured sipx and I successfully made calls through call controller > API > but when I tried to implement conference web service API I could get the > response from the argument xml_list,volume_in,kick,lock but when I tried > to dial a number in to the conference using "curl --digest -k -X PUT > https://userid:Password@hostname:8085/conference/conference- > name/dial/XXXXX/dial-number".I couldnt get what is the > <endpoint_module_name> for the same. Can anybody give me the correct url for dial service in conference and how can I figure what is the endpoint-name in my system? -- View this message in context: http://old.nabble.com/sipx-conference-API-tp32129940p32129940.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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rxfax on b-leg
by slash＠nat-tele.com 26 Jul '11

26 Jul '11

Hello! I wonder if there is a way to make rxfax recieve fax on outgoing call? The idea is that registered FS user dials out, than tone_detect is used to detect fax tone and start rxfax. The difference is that rxfax should be run on B-leg, instead of A-leg in incoming fax call scenario. I've tried to make a test config with outgoing call, rxfax & tone_detect, but when I dial out and remote user starts sending fax to me, FS detects a tone correctly, starts rxfax, but on my leg (A-leg, i hear fax tones) and B-leg (with real fax machine) is hanged up. Are there are solutions to explain somehow to rxfax that it should be run on B-leg, not on A-leg? Thanks in advance! With best regards, Kirill.

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SDP Append
by Ricardo Martinez 25 Jul '11

25 Jul '11

Hello. I’m trying to append two distinct SDP fields in the outgoing call leg from Freeswitch. Right now I’m only capable to attach only one : <extension name="add-annexb=no" continue="true"> <condition field="${switch_r_sdp}" expression="/(.*)(a=fmtp:18 annexb=no)(.*)/s"> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> </condition> </extension> Because when I’m try to attach a second one, it rewrites the first one. Thanks. Ricardo Martinez.-

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Re: [Freeswitch-users] Can't get variable set in dial plan
by Joshua Foshee 25 Jul '11

25 Jul '11

That fixed it. The fix was to change hours to ${hours} in the condition field. Thanks, Josh From: Steven Ayre [mailto:steveayre@gmail.com] Sent: Monday, July 25, 2011 3:40 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Can't get variable set in dial plan Variables are not fields. Fields are a specific set of attributes of the call... you can however check variables in a condition, it just has a slightly different syntax. Try this. <extension name="public_did"> <condition field="destination_number" expression="^(.*)$"> <action inline="true" application="set" data="hours=1"/> </condition> <condition field="${hours}" expression="^1$"> <action application="answer"/> <action application="ivr" data="GAC-BH"/> <action application="hangup"/> <anti-action application="answer"/> <anti-action application="ivr" data="GAC-BH"/> <anti-action application="hangup"/> </condition> </extension> Also note I changed your regex of "1" to "^1$". The first matches any string containing a digit 1 including 10, 21, 100, 200351. The latter matches the start (^) and end ($) of the string so will only match the exact string of 1. -Steve On 25 July 2011 20:54, Joshua Foshee <Joshua.Foshee(a)logixcom.com> wrote: I am trying to set a variable in the dial plan and use it on a condition. I have pushed the variable setting as close to the condition for testing and this is what I get. Why does it not see the variable set? <extension name="public_did"> <condition field="destination_number" expression="^(.*)$"> <action inline="true" application="set" data="hours=1"/> </condition> <condition field="hours" expression="1"> <action application="answer"/> <action application="ivr" data="GAC-BH"/> <action application="hangup"/> <anti-action application="answer"/> <anti-action application="ivr" data="GAC-BH"/> <anti-action application="hangup"/> </condition> </extension> Here is the console output Dialplan: FreeTDM/1:7/405790xxx8 Regex (PASS) [public_did] destination_number(405790xxx8) =~ /^(.*)$/ break=on-false Dialplan: FreeTDM/1:7/405790xxx8 Action set(hours=1) INLINE EXECUTE FreeTDM/1:7/405790xxx8 set(hours=1) 2011-07-25 12:05:25.191357 [DEBUG] mod_dptools.c:1063 FreeTDM/1:7/405790xxx8 SET [hours]=[1] Dialplan: FreeTDM/1:7/405790xxx8 Regex (FAIL) [public_did] hours() =~ /1/ break=on-false Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action answer() Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action ivr(GAC-BH) Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action hangup() _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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