April 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

WebRTC + Chrome = RTP Socket Error!
by Luiz Fernando Softov 07 Apr '15

07 Apr '15

I'm testing WebRTC + Chrome Using http://sipjs.com When im using FireFox, its work fine, i can make and receive calls with audio and/or video But when i try to use chrome. The other extension rings, but when i answer the call I receive this error, and call hangup 2015-04-07 05:51:39.016628 [ERR] switch_core_media.c:5775 AUDIO RTP REPORTS ERROR: [Socket Error!] 2015-04-07 05:51:39.016628 [NOTICE] switch_core_media.c:5776 Hangup sofia/internal/1001(a)192.168.1.100 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] Has anyone experienced the same problem? -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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Originate_timeout longer than 5 minutes?
by Fredrik Larsson 07 Apr '15

07 Apr '15

Hi, I am sending originate_timeout=600 on ESL but it always terminates after 5 minutes e.g. 300 seconds. I have tried to find out if there is some hardcoded max variable but I can't find it. Do anyone know if there is some other variable I can change to get higher value or is this per design? Regards, Fredrik

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Early Dial / Sip 484
by Paul Atreides 07 Apr '15

07 Apr '15

Hi, does Freeswitch support Early Dial ( Sip 484 ? ), if yes how do i set it up in the dialplan? Thanks

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Sofia.h limits ?
by Abdul Hakeem 06 Apr '15

06 Apr '15

Hello Guys, Could someone shed lights if these lines are hard codec limits and suggestions for alternative parameters ? #define SOFIA_MAX_AL 100 #define MAX_RTP 50 #define SOFIA_MAX_MSG_QUEUE 64 Many thanks, Abdul Hakeem

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FreeSWITCH Week in Review (Master Branch) March 28th-April 3rd
by Ken Rice 06 Apr '15

06 Apr '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1GZCf13 FreeSWITCH Week in Review (Master Branch) March 28th-April 3rd Hello, again. This passed week in the FreeSWITCH master branch we had 6 commits. We have a few great features this week including a new module! First, a new header, DTMF-Source, was added to DTMF event. DTMF-Source can have the following values: APP : injected by application (send_dtmf, etc), ENDPOINT : detected by endpoint signaling (like SIP INFO), INBAND_AUDIO : detected by start_dtmf, spandsp_start_dtmf, etc, RTP : detected as 2833/4733 telephone event, and UNKNOWN : unknown source. The new module is called mod_amqp (Advanced Message Queueing Protocol). This new module will allow FreeSWITCH to natively send events to amqp servers, like rabbitmq, and also subscribe to a channel for a FreeSWITCH API to run locally. This functionality can replace the need for ESL connections. Special thanks-to: Daniel Bryars and Tim Brown from aeriandi.com for publishing the module into ‘contrib’ and William King and Mike Jerris for cleaning and expanding the module to merge into the master branch. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7406 Added DTMF-Source header to DTMF event. DTMF-Source may have the following values: APP : injected by application (send_dtmf, etc) ENDPOINT : detected by endpoint signaling (like SIP INFO) INBAND_AUDIO : detected by start_dtmf, spandsp_start_dtmf, etc RTP : detected as 2833/4733 telephone event UNKNOWN : unknown source. And added deduplicate_dtmf application FS-7060 Adding mod_amqp as an event_handler module, AMQP(Advanced Message Queueing Protocol) is an event bus protocol with more info here http://ift.tt/1iHGPkc mod_amqp implements the client side of the amqp protocol. Currently the module supports two separate, but not mutually exclusive usages: 1. Sending a filtered whitelist of events to an AMQP server. The before the events are sent an AMQP routing header is generated so that downstream subscriptions can subscribe using filters. If the connection to the remote server is interrupted, the module will attempt to reconnect. 2. Subscribing to a ‘control’ channel on the remote AMQP server. When events are received on this channel if they match a configured filter, the commands are run locally as api commands. If the filter for the events being sent to the remote AMQP server include SWITCH_EVENT_API then the results of the api commands will be sent back to the AMQP server. Mod_amqp was also added to the Debian packaging. Thanks-to: Daniel Bryars and Tim Brown

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Failed to load radius handle for digest invite authentication
by Густаво Силва 06 Apr '15

06 Apr '15

Hi everyone! How to solve this problem occurring on Freeswitch with the xml_radius module? 2015-04-06 11:48:25.284825 [ERR] mod_xml_radius.c:633 Failed to load radius handle for digest invite authentication 2015-04-06 11:48:25.284825 [ERR] mod_xml_radius.c:946 Failed to create new accounting_start handle for call: 0fcb1214-4941-477b-b329-2d034623c1e9 My freeradius server is getting this parameters(sniffed by wireshark: 192.168.56.1 - softphone, 192.168.56.105 - Freeswitch/radiusclient, 192.168.56.103 - Freeradius server): Radius Protocol Code: Accounting-Request (4) Packet identifier: 0xf9 (249) Length: 240 Authenticator: 19a3f95c13b4811e35b43d5390b29f8a Attribute Value Pairs AVP: l=6 t=Acct-Status-Type(40): Start(1) Acct-Status-Type: Start (1) AVP: l=38 t=Acct-Session-Id(44): 0fcb1214-4941-477b-b329-2d034623c1e9 Acct-Session-Id: 0fcb1214-4941-477b-b329-2d034623c1e9 AVP: l=6 t=User-Name(1): 1002 User-Name: 1002 AVP: l=12 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=6 t=Freeswitch-Src(4): 1002 Freeswitch-Src: 1002 AVP: l=12 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=6 t=Freeswitch-CLID(2): 1002 Freeswitch-CLID: 1002 AVP: l=12 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=6 t=Freeswitch-Dst(5): 1001 Freeswitch-Dst: 1001 AVP: l=11 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=5 t=Freeswitch-Dialplan(3): XML Freeswitch-Dialplan: XML AVP: l=6 t=Framed-IP-Address(8): 192.168.56.1 Framed-IP-Address: 192.168.56.1 (192.168.56.1) AVP: l=8 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=2 t=Freeswitch-RDNIS(17): Freeswitch-RDNIS: AVP: l=15 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=9 t=Freeswitch-Context(18): default Freeswitch-Context: default AVP: l=12 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=6 t=Freeswitch-Ani(8): 1002 Freeswitch-Ani: 1002 AVP: l=8 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=2 t=Freeswitch-Aniii(9): Freeswitch-Aniii: AVP: l=17 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=11 t=Freeswitch-Source(19): mod_sofia Freeswitch-Source: mod_sofia AVP: l=39 t=Vendor-Specific(26) v=FreeSWITCH(27880) VSA: l=33 t=Freeswitch-Callstartdate(20): 2015-04-06T11:48:25.284825+0400 Freeswitch-Callstartdate: 2015-04-06T11:48:25.284825+0400 AVP: l=6 t=NAS-Port(5): 0 NAS-Port: 0 AVP: l=6 t=Acct-Delay-Time(41): 0 Acct-Delay-Time: 0 AVP: l=6 t=NAS-IP-Address(4): 192.168.56.105 NAS-IP-Address: 192.168.56.105 (192.168.56.105) Regards, Gustavo Silva

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freeswitch.dbh / dbh:query
by Paul Atreides 06 Apr '15

06 Apr '15

Hi, I am sorry I don't get it :/ How do I pull of a single value from this query: "SELECT uuid FROM fifo_intercept ORDER BY time_id DESC LIMIT 1" How can I save the result into the variable fifo_uuid with dbh:query?

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Intercept caller number
by Paul Atreides 06 Apr '15

06 Apr '15

Hi, when I do a global intecept then I dont see the original caller number in the display. I see the dialplan number for the intercept instead. How can I replace it with the original caller number? Thank you

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Transfer/shift calls from one freeswitch to other freeswitch box
by Varghese Paul 06 Apr '15

06 Apr '15

Hi all, Is it possible to transfer/forward calls from one freeswitch box to other box ?. Use case: We wanted to do some maintenance on one freeswitch server and thinking to transfer/shift the current calls in one server to other server. Is there any way to route the calls to other freeswitch server without hanging up the call?. I know we should send Re-INVITE to other users for modifying the RTP IP + RTP port number for the new Freeswitch IP. Is it possible to achieve in freeswitch ? Regards Varghese Paul

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bridge_answer_timeout
by Gabriel Kuri 04 Apr '15

04 Apr '15

I can't seem to get bridge_answer_timeout to work. I have bridge_answer_timeout=10 and ignore_early_media=false set in the dialplan, but it just seems to get ignored and it rings the first extension (100) indefinitely like bridge_answer_timeout isn't getting set. It should ring extension 100 for 10 seconds, then ring extension 102 for 10 seconds, then send the call to voicemail. Here's the dialplan, any ideas? <document type="freeswitch/xml"> <section name="dialplan" description="Dial Plan for FreeSWITCH"> <context name="default"> <extension name="call to 100"> <condition field="uuid" expression="293df60c-d51f-4dc1-93d3-101584774665"> <action application="set" data="effective_caller_id_number=101"></action> <action application="set" data="effective_caller_id_name=Test 101"></action> <action application="set" data="sip_cid_type=pid"></action> <action application="privacy" data="no"></action> <action application="set" data="ignore_early_media=false"></action> <action application="set" data="continue_on_fail=true"></action> <action application="set" data="bypass_media=false"></action> <action application="bridge" data="[bridge_answer_timeout=10]sofia/${external_sip_profile}/100(a)switch.com "></action> <action application="bridge" data="[bridge_answer_timeout=10]sofia/${external_sip_profile}/102(a)switch.com "></action>  <action application="bridge" data="sofia/${external_sip_profile}/ 100(a)voicemail-switch.com"></action> </condition> </extension> </context> </section> </document> Thanks ...

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