July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ClueCon 2011 Update - Sofitel Full, Alternate Hotel Arrangements Being Made
by Michael Collins 27 Jul '11

27 Jul '11

Hello! Thank you all for supporting ClueCon 2011! We've maxed out the Sofitel, so we're looking at another hotel for our attendees. If you are coming to ClueCon and need a hotel room please contact Brian West ASAP at marketing(a)cluecon.com or 877-742-CLUE. We need to know how many rooms we can guarantee to another hotel in order to get discounted rates. Time is of the essence, so please contact us right away. Thanks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Re: [Freeswitch-users] Gateway real-time
by chrisbware＠interfree.it 27 Jul '11

27 Jul '11

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FreeSWITCH Conference Call Today
by Michael Collins 27 Jul '11

27 Jul '11

Hey all, We are having a simple FreeSWITCH conf call today: http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_27 I want to talk about mod_fifo specifically and call queuing in general. If you use mod_fifo at all please join us. There have been a lot of updates to mod_fifo that make it more "ACD-like" but the wiki page needs some serious attention. Please hop on and help us get the page updated. Thanks! -MC

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Can record_min_sec be used in record dial plan tool?
by Adam Kelloway 27 Jul '11

27 Jul '11

Hi there, In regards to http://wiki.freeswitch.org/wiki/Variable_record_min_sec : I can't seem to get record_min_sec to work when using the record dial plan tool. It doesn't detect that a recording is too short and therefore doesn't delete the file. A look at the code seems to suggest that it is only checked when you are using the media_bug_answer_req feature and/or the record_session dial plan tool. Is this true? Is there any way I can enable this for the record dial plan tool? My extension does the following: ... <action application="set" data="record_min_sec=2"/> <action application="record" data="/tmp/${uuid}.wav 30 200"/> ... I also tried the following, with no difference in results: ... <action application="set" data="RECORD_MIN_SEC=2"/> <action application="record" data="/tmp/${uuid}.wav 30 200"/> ... Thanks, Adam

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Intercept taking too long
by Alex Crow 27 Jul '11

27 Jul '11

Hi, I have set up a system with 6 snom 370s and FS and I'm seeing a long delay (up to 3 seconds) when trying to do a pickup using the BLF. FS receives the **<dest> as soon as the button is pressed but it seems the delay is in tearing down the call to the original destination. This is much longer than it took on an Elastix install with 6 Grandstream GXP2000s. Is there anything I can do to speed this up? It's causing some confusion for our users as they are pressing the pickup button and then start greeting the caller straight away, not realising it's going to take another couple of seconds for them to arrive. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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NAT and Freeswitch
by Narendra Sirugudi 27 Jul '11

27 Jul '11

__________________________ | | |________ ____|_____ ___________ || | | | 74.1.1.1 74.1.1.2 | | || SIP UA |---------------------| NAT Server|------------------------------------------| Freeswitch | ||________|10.10.10.10 |__________| Public Network |___________| | | 74.1.1.0 | Private Network 10.10.10.0 | |__________________________| Hi, In the above topology, the typical issue one faces is that the SDP(rtp) IP&port in SIP invite sent from the SIP UA to Freeswitch will contain the private Ip address of the SIP UA. Assuming source NAT, the typical solution is to use a STUN server in the public network, which will help the SIP UA to discover the public IP&port (for its private IP&port). The UA then sends the Invite with the SDP(RTP) IP&port set to the discovered public IP&port. (I believe Freeswitch does not have a STUN server within itself.) But suppose that we do not want to use the STUN solution. Does Freeswitch itself offer any solutions in this case ? thanks, --naren

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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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