December 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference timer synchronization with T1 FreeTDM - how?
by Bob Hartwig 20 Dec '13

20 Dec '13

Looking at the source for mod_conference.c, I see that the timer_name is hard-coded to "soft", i.e. the 1000 Hz. system clock. How can this work with a T1-based FreeTDM system, where the FreeTDM channels are presumably synchronized to the T1 clock source? I'm seeing some weirdness with my FreeTDM / Dahdi system, where only FreeTDM channels are hearing distortion on conference, and I'm almost certain that this is the culprit, but I have no idea how to proceed to fix it. I'm pretty new to FreeTDM, so I'm hoping that I'm missing something simple. Any thoughts? Thanks, Bob

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ESL Outbound for driving IVRs
by Alex Balashov 19 Dec '13

19 Dec '13

Hi, Is there any practical way to take drive IVRs dynamically - via an API - while taking advantage of the more high-level features of the 'ivr' module? As I understand it, there's no way to pass in the IVR configuration parameters, i.e. the <menu> tags and their parameters, via ESL. At least, I can't find 'ivr' API commands anywhere. It seems to me that the only way to do this dynamically would be to mod_xml_curl to dynamically generate the XML config data in those sections. Can anyone confirm or deny that? It would certainly be nice to be able to do this purely through ESL outbound instead. Of course, at the end of the day, all these IVR actions can be manually scripted out through dialplan applications and orchestrated through ESL that way. But that implies a fair amount of wheel invention, on the basis that the 'ivr' module provides higher-level wrapping for a lot of it. I'd like to use it if possible. Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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in case of failure multiple condition and action bridge not working how to dial from different gateways according prefix numbers?
by netandtech 19 Dec '13

19 Dec '13

Thank you for that ! That helps a lot . I am having another question In case of failure can I add another condition below with another bridge to another gateway? What I want to achieve is in case of a gateway failure the call to continue with another gateway .. Also I am wondering how to make dial tone when someone open the handset and when a numbers is dialed how to hear the normal dialing tones/ Thank you in advanced! From: Andrew Cassidy [mailto:andrew@cassidywebservices.co.uk] Sent: Thursday, December 19, 2013 12:44 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] multiple condition and action bridge not working how to dial from different gateways according prefix numbers? Try adding break="on-true" to this tag: <condition field="destination_number" expression="^69(\d+)$"> -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director <http://www.cassidywebservices.co.uk/static/emailsig.png> <mailto:info@cassidywebservices.co.uk> T 03300 100 960 <mailto:info@cassidywebservices.co.uk> F 03300 100 961 <mailto:info@cassidywebservices.co.uk> E andrew(a)cassidywebservices.co.uk <mailto:andrew@cassidywebservices.co.uk> <mailto:info@cassidywebservices.co.uk> W www.cassidywebservices.co.uk <http://www.cassidywebservices.co.uk>

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Can API commands return something other than CSV?
by Fraser Redmond 19 Dec '13

19 Dec '13

I'm connecting to FS from PHP over a socket, and parsing results using the php function str_getcsv() In my dialplan I have: <action application="playback" data="tone_stream://%(250,250,600,120)"/> When I run "api show channels" if a channel is currently playing that tone, then within the returned results I get these 3 headers and values: <snip>,application,application_data,dialplan,<snip> <snip>,playback,tone_stream://%(250,250,600,120);loops=3,XML,<snip> So the data-separator is the comma, but the data also contains commas without being escaped in any way. Is there any way to change the results to be returned in json? Or change the delimiter? I've made a regex replace to get around this particular case by matching parentheses... But I can see this happening again with some other value that has a comma outside of parentheses, and would rather get the results in a better format if possible. (In the wiki, I can see that CDRs and events can be sent in json, and the serialize command for events, but none of those would help this particular case.) Cheers, Fraser

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Changing callee_id_name in a call
by Markus Carlstedt 19 Dec '13

19 Dec '13

Hi, I am trying to change the effective_callee_id_name when in a call. I tried using: uuid_setvar <uuid> effective_callee_id_name Testing I can see that the variable is set via: uuid_getvar <uuid> effective_callee_id_name But the info is not sent to the telephone, atleast not what I could see. / Markus

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multiple condition and action bridge not working how to dial from different gateways according prefix numbers?
by netandtech 19 Dec '13

19 Dec '13

From: netandtech [mailto:bill_vus@hotmail.com] Sent: Wednesday, December 18, 2013 5:28 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] FW: how to dial from different gateways according prefix numbers? From: netandtech [mailto:bill_vus@hotmail.com] Sent: Wednesday, December 18, 2013 5:27 PM To: freeswitch Subject: how to dial from different gateways according prefix numbers? The following is the extension number that I use in the default.xml What I want to achieve is to Route the calls for landlines to different gateways and calls to mobiles from another gateway. <extension name="1002">   <condition field="destination_number" expression="^69(\d+)$"> <action application="transfer" data="sofia/gateway/myvoip/$0"/> </condition> <condition field="destination_number" expression="^22810(\d+)$"/> <action application="transfer" data="sofia/gateway/voipdito2/$0"/> </extension>

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issue of freeswitch working with doubango webrtc clients
by Vincent Xia 19 Dec '13

19 Dec '13

im testing the interoperability of freeswitch and doubango sip clients including boghe and imsdroid, that both have the setting of media profile: default or webrtc, setting to default the call is fine as from or to the boghe/imdroid client, but setting to webrtc, these clients could only make outgoing calls, when receiving calls they respond the fs invite message with 488 "bad content", the fs console says the call failed due to incompitalbe destination: 2013-12-17 18:03:50.158267 [NOTICE] switch_ivr_originate.c:2699 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION] 2013-12-17 18:03:50.158267 [INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION the instruction document from doubango says for the webrtc clients to work with the "legacy" sip network a webrtc2sip module is required to as a sip proxy and webrtc breaker, but now freeswitch supports webrtc and sip over websocket, is a webrtc breaker still mandatory?

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Call quality statistics from CDR reports?
by d.campbell 18 Dec '13

18 Dec '13

Greetings freeswitch community, Freeswitch is an amazing piece of work, and I have been able to use it with great success in large-scale scenarios. Much thanks to all the contributors! One feature I'm not finding is a report on call quality for SIP/RTP calls. I see that the CDRs contain several potentially relevant values including: rtp_audio_in_jb_packet_count rtp_audio_in_flush_packet_count rtp_audio_in_largest_jb_size Unfortunately, there is no documentation on exactly what these values mean. Would it be possible to compute packet loss and jitter statistics on the inbound call from these numbers or others? Much thanks, Doug Campbell

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overriding default timer values
by Grant Bagdasarian 18 Dec '13

18 Dec '13

Hello, The SIP RFC defines some timers and their default values. Is it possible to override these values in sip profiles or somewhere else? T1 500ms default Section 17.1.1.1 RTT Estimate T2 4s Section 17.1.2.2 The maximum retransmit interval for non-INVITE requests and INVITE responses T4 5s Section 17.1.2.2 Maximum duration a message will remain in the network Timer A initially T1 Section 17.1.1.2 INVITE request retransmit interval, for UDP only Timer B 64*T1 Section 17.1.1.2 INVITE transaction timeout timer Timer C > 3min Section 16.6 proxy INVITE transaction bullet 11 timeout Timer D > 32s for UDP Section 17.1.1.2 Wait time for response 0s for TCP/SCTP retransmits Timer E initially T1 Section 17.1.2.2 non-INVITE request retransmit interval, UDP only Timer F 64*T1 Section 17.1.2.2 non-INVITE transaction timeout timer Timer G initially T1 Section 17.2.1 INVITE response retransmit interval Timer H 64*T1 Section 17.2.1 Wait time for ACK receipt Timer I T4 for UDP Section 17.2.1 Wait time for 0s for TCP/SCTP ACK retransmits Timer J 64*T1 for UDP Section 17.2.2 Wait time for 0s for TCP/SCTP non-INVITE request retransmits Timer K T4 for UDP Section 17.1.2.2 Wait time for 0s for TCP/SCTP response retransmits

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Restrict SIP registration
by mehroz 18 Dec '13

18 Dec '13

Hi All, We already know that we can allow/deny a SIP end point to get registered to FreeSWITCH on the basis of IP , but what about if we want it to filter "User-Agent" field in REGISTER packets to be the key for registration. Since , we can also use TLS based registration but some soft phones does not have that capability. In that case , is there any way of passing only specified USER AGENT SIP client? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Restrict-SIP-registration-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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