August 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch modems flow control?
by Alex Crow 29 Aug '15

29 Aug '15

Hi, I asked this question a *long* time ago as I was having major problems receiving faxes using HylaFAX and the builtin FS Modems, in that the FS Modems did not have any kind of flow control, which resulted in a repeated backing off of baud rate until the incoming faxes failed with a rate of over 50%. Has there been any change in the softmodem code to add flow control so that this does not happen any more? Best regards Alex

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Unable to get channel variables from directory to dialplan.
by Vladislav Ivanov 29 Aug '15

29 Aug '15

Good Day, I have been trying to set up the sip phone at my home, and i'm having some issues with channel variables. >From what I have read: https://wiki.freeswitch.org/wiki/Routing_with_directory_parameters I should force per-call authentication in your sofia.xml.conf for <variables> to be passed through to the dialplan. But I were unable to find this option in sofia/sofia_default files. https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables - Although here there is nothing about "per-call authentication". And I'm not sure how to proceed. I have this directory xml: <include> <user id="123123"> <params> <param name="password" value="123123"/> </params> <variables> <variable name="accountcode" value="123123"/> <variable name="user_context" value="test_context"/> <variable name="effective_caller_id_name" value="Test Phone"/> <variable name="effective_caller_id_number" value="123123"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> </include> Directory variables should be passed as ${variable} as they are channel variables? I have Dialplan as follows: <extension name="registrar_test"> <condition field="destination_number" expression="^(.*)$"> <action application="log" data="ERR TestMessage [${user_context}]"/> <action application="respond" data="503 Route Successfull"/> </condition> </extension> <extension name="all_reject"> <condition field="destination_number" expression="^(.*)$"> <action application="respond" data="503 No Route"/> </condition> </extension> When I dial the call I get TestMessage [] instead of test_context. Also, wrapping registrar_test extension with <context name="test_context"></context> doesn't help, my call will go to "all_reject" extension. What do I miss? Best Regards, Vlad

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Receiving faxes over sip trunk
by Michael T Mowatt 29 Aug '15

29 Aug '15

HI all, Was wondering if anyone is using FreeSWITCH to receive faxes over a SIP trunk, and what the experience has been like. I know I've heard this can be difficult, but was interested in real experiences. Thanks, Mike

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OPUS, pjsip, csipsimple bug for 1 chanel
by Tim Meade 29 Aug '15

29 Aug '15

We are playing around with opus on iOS and freeswitch. Dialing out from iOS through FS shows this: Invalid SDP for opus. Don't ask.. but it needs a /2 FS seems to fix the invalid opus and the call sounds fantastic. The problem is on sending opus TO the iOS device. FS is sending: a=rtpmap:102 opus/48000/2. Which is correct and true. We are reading that this particular implementation of opus from cSipSimple has a know bug on the /2 it just wants opus/48000. Does anyone know of a fix for either the cSipSimple side or a workaround for FS to just send the opus/48000. ? Thanks Tim

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Unable to continue dialplan if legB fails on early-media with multiple destination
by gdesai 29 Aug '15

29 Aug '15

I need to implement following scenario: 1) Leg A comes to FS. 2) FS answers the call and plays a prompt. 3) FS originates leg B (with two destination) 4) Leg B sends ringback tone for source. 5) If Leg A does not answer, FS tries second destination. Problem is when Leg B does not answer, FS hangs up Leg B. When Leg B hangs up, FS also hangs up Leg A. Instead of keeping Leg A and dialing second destination. My originate dialplan: originate {origination_caller_id_number=18888888888,ignore_early_media=true,park_after_bridge=true}[origination_uuid=uuid,sip_h_Subject='"Direct" 17777777777'][leg_timeout=15,park_after_bridge=true]sofia/gateway/40/5555555555,[leg_delay_start=12]sofia/gateway/40/4444444444'&managed(CallCloudAppServer.Service.OutboundLeg)' Note: When early media is received from Leg B, I am bridging both legs and then park Leg A. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Unable-to-continue-dialplan-i… Sent from the freeswitch-users mailing list archive at Nabble.com.

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JSON-based API calls
by David McNab 29 Aug '15

29 Aug '15

Hi there, How do you do API calls in JSON? Or, how do I get a list of all the API calls which can receive JSON parameters and return JSON return values? For example, in the console, I tried 'show calls', and got a list of calls. I did the same call from via XML-RPC from a python script, and got a single string back, containing what would have been output to the console. But when I tried, from the console: *json {"command":"show calls", "format":"pretty", "data":{}}* I got an invalid response. I also tried: *json {"command":"show", "format":"pretty", "data":"calls"}* but got similar. Is there a general sceme for invoking all the console API commands (and receiving responses) in JSON format? If not, I guess I need to write some parsers to process each command's output string. Cheers David

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The FreeSWITCH 1.4.21 release is here!
by Ken Rice 28 Aug '15

28 Aug '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1JC21dn The FreeSWITCH 1.4.21 release is here! The FreeSWITCH 1.4.21 release is here! This is a routine maintenance release and the resources are located here: Tarballs: http://ift.tt/1JBF6un Packaging: http://ift.tt/1U8FMQN New features that were added: FS-7760 [mod_sofia] Added xml fetch for channels to externally support nightmare transfer depends on channel-xml-fetch-on-nightmare-transfer profile param (default is disabled) Improvements in build system, cross platform support, and packaging: OPENZAP-238 [freetdm] Fix some GSM compilation errors and do a bit of code cleanup OPENZAP-237 [freetdm] Use __func__ instead of __FUNCTION__ to comply with c99 in gcc 5.1 FS-7043 Fixed apr1 unresolved symbols in libfreeswitch.so.1.0.0 FS-7985 [mod_voicemail] Fixed a compilation error on 32-bit PCC platform FS-8047 [mod_basic] Fixed a build error on newer compiler The following bugs were squashed: FS-8028 [mod_shout] fix random sockets being closed regression from FS-7621 FS-8037 [core] zrtp-passthru shouldn’t activate unless the zrtp-hash is in the SDP. FS-7732 [mod_commands] Continue recording with uuid_transfer FS-7721 [core] Fixed a segfault caused when using session:recordFile() and session:unsetInputCallback in a lua script FS-7429 [mod_curl] Fixed to output valid json. NOTE: this may cause issues if you have worked around this invalid json already FS-7761 [core] Fix shutdown races running api commands during shutdown FS-7767 [mod_sofia] Fixed a segfault caused by invalid arguments to sip_dig FS-7778 [mod_sofia] Fixed a bug causing a SQL statement to fail because of a double quote instead of a single quote FS-7754 [freetdm] Fixed a bug relating to single digit dial-regex with analog devices FS-7731 [mod_xml_cdr] Fixed a curl default connection timeout FS-7685 [mod_say_nl] Fix Dutch numbers pronunciation FS-7654 [core] Fixed regressions on eavesdropping on channels playing a file and on channels with unlike rates FS-7866 [core] Fixed a crash when running incorrect var api expansion syntax “eval ${${external_sip_ip}:4}” FS-7872 [mod_verto] Gracefully fail attempting to transfer 1 legged call FS-7891 [mod_spandsp] Allow spandsp dtmf detector to work on rates other than 8k FS-7839 [webrtc] Correct firefox > 38 DTLS behavior to match new EC requirements FS-7917 [mod_sofia] Fixed default config, we really shouldn’t be setting ext-*-ip settings for ipv6 profiles FS-7951 [mod_rayo] Completely clean up mod_rayo if it fails to load FS-7955 [mod_sofia] Fixed a crash caused by invalid contact when using event to send a notify message FS-7975 [mod_voicemail] Fix record-greeting event missing VM-Greeting-Path FS-7969 Fixed a segfault due to pthread_setschedparam() on a thread that has exited FS-7962 [mod_sofia] Fixed sporadic invite/replaces failure FS-7486 [mod_sofia] Fixed the handling of queued requests FS-7750 [mod_commands] Set uuid_setvar to handle arrays

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 28 Aug '15

28 Aug '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Pickup from a Queue - mod_callcenter
by Raphael Lechner 28 Aug '15

28 Aug '15

Hi, We have a customer that want to Pickup from time to time a call from a queue. It seems that picking up a call from a Queue is not intended to be used in combination with a Queue or I’m wrong with that? I created a solution for that customer that store the UUID in a database via a python script before the caller gets into the queue and a function to retrieve that UUID if the caller is still in the queue and to pick up with that UUID. After the call get answered the UUID get deleted. This function is working fine, but maybe somebody knows a better way to picking up calls from a Queue? Thanks in advance, Raphael

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ESL, "181 Call Is Being Forwarded" and voice mail detection
by Marcin S 28 Aug '15

28 Aug '15

Hi, I have a dialer app that uses ESL and freeswitch. I would like to detect when voice mail answers call. For such calls there is a "181 Cal Is Being Forwarded" message in SIP signalization. However, in ESL there is no event fired (maybe beacuse there was 183 and 180 earlier). How can I detect, that there was 181 received? Here is an example sip trace: 1 0.000000 192.168.88.237 -> 89.108.20.42 SIP/SDP Request: INVITE sip:123456789@89.108.20.42, with session description 2 0.018028 89.108.20.42 -> 192.168.88.237 SIP Status: 100 Trying 3 0.418550 89.108.20.42 -> 192.168.88.237 SIP/SDP Status: 183 Session Progress, with session description 4 1.405465 89.108.20.42 -> 192.168.88.237 SIP Status: 180 Ringing 5 29.270672 89.108.20.42 -> 192.168.88.237 SIP Status: 181 Call Is Being Forwarded 6 29.659371 89.108.20.42 -> 192.168.88.237 SIP/SDP Status: 200 OK, with session description 7 29.670833 192.168.88.237 -> 89.108.20.42 SIP Request: ACK sip:3c6d5343fb70df3a18f4dd32e7c9d226@89.108.20.42:5060;transport=udp 8 70.590778 192.168.88.237 -> 89.108.20.42 SIP Request: BYE sip:3c6d5343fb70df3a18f4dd32e7c9d226@89.108.20.42:5060;transport=udp 9 70.625224 89.108.20.42 -> 192.168.88.237 SIP Status: 200 OK

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