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FreeSWITCH as a softphone with PulseAudio under Linux
by Jason White 20 Nov '13

20 Nov '13

I'm interested in running FreeSWITCH as a softphone on Linux systems with PulseAudio installed. This bug is exactly on point (the plan is to update the PortAudio library to solve the issue): http://jira.freeswitch.org/browse/FS-3387 However, the bug hasn't been updated since April. Is anybody currently working on it? If not, is anybody writing a module that would interface with PulseAudio directly rather than through PortAudio? Unfortunately I don't expect to have time to contribute to this for quite a while, if at all, but I can help with testing when there's a solution implemented.

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Multiple attended transfer of an inbound call
by Support Technique 20 Nov '13

20 Nov '13

Hi team, I have Freeswitch setup so when an incoming call is answered, it can be transferred to another extension or into a valet parking lot. That`s fairly simple using the transfer key of the phones. What I am having issue with is if the extension that answered the transferred call (or picked up the parked call) wants to transfer it somewhere else or park it again, things will work until that extension hangs up; meaning I hit the transfer key and make the other extension to ring, it answers but once I hang up, everything hang up so the original incoming call is not connected. I get this in the log: -- The call is coming in 2013-11-19 17:17:13.938118 [NOTICE] switch_channel.c:968 New Channel sofia/internal/5551212(a)192.168.0.10 [579ac220-5168-11e3-9adb-1f655b7322d0] 2013-11-19 17:17:14.018119 [INFO] mod_dialplan_xml.c:557 Processing Benoit <5551212>->gxw01 in context public 2013-11-19 17:17:14.018119 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:399@192.168.0.101:5063 [57a74022-5168-11e3-9aef-1f655b7322d0] 2013-11-19 17:17:14.028112 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:300@192.168.0.102:5060 [57a797d4-5168-11e3-9af9-1f655b7322d0] 2013-11-19 17:17:14.078115 [NOTICE] sofia.c:5666 Ring-Ready sofia/internal/sip:300@192.168.0.102:5060! 2013-11-19 17:17:14.088114 [NOTICE] mod_sofia.c:2746 Ring-Ready sofia/internal/5551212(a)192.168.0.10! 2013-11-19 17:17:14.088114 [NOTICE] switch_ivr_originate.c:521 Ring Ready sofia/internal/5551212(a)192.168.0.10! 2013-11-19 17:17:14.378117 [NOTICE] sofia.c:5666 Ring-Ready sofia/internal/sip:399@192.168.0.101:5063! 2013-11-19 17:17:15.568118 [NOTICE] sofia.c:6326 Channel [sofia/internal/sip:300@192.168.0.102:5060] has been answered 2013-11-19 17:17:15.568118 [NOTICE] switch_ivr_originate.c:3351 Hangup sofia/internal/sip:399@192.168.0.101:5063 [CS_CONSUME_MEDIA] [LOSE_RACE] 2013-11-19 17:17:15.578113 [NOTICE] sofia_glue.c:4259 Pre-Answer sofia/internal/5551212(a)192.168.0.10! 2013-11-19 17:17:15.578113 [NOTICE] switch_ivr_originate.c:3378 Channel [sofia/internal/5551212(a)192.168.0.10] has been answered 2013-11-19 17:17:15.578113 [NOTICE] switch_core_session.c:1517 Session 186 (sofia/internal/sip:399@192.168.0.101:5063) Ended 2013-11-19 17:17:15.578113 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/sip:399@192.168.0.101:5063 [CS_DESTROY] -- Now I transfer to extension 399 2013-11-19 17:17:20.238119 [NOTICE] switch_ivr.c:1801 Transfer sofia/internal/5551212(a)192.168.0.10 to XML[399@public] 2013-11-19 17:17:20.268119 [NOTICE] switch_ivr_bridge.c:685 Hangup sofia/internal/sip:300@192.168.0.102:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-11-19 17:17:20.268119 [INFO] mod_dialplan_xml.c:557 Processing Benoit <4504324671>->399 in context public 2013-11-19 17:17:20.268119 [NOTICE] switch_core_session.c:1517 Session 187 (sofia/internal/sip:300@192.168.0.102:5060) Ended 2013-11-19 17:17:20.268119 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/sip:300@192.168.0.102:5060 [CS_DESTROY] 2013-11-19 17:17:20.368119 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:399@192.168.0.101:5063 [5b6f1824-5168-11e3-9b0d-1f655b7322d0] 2013-11-19 17:17:20.618118 [NOTICE] sofia.c:5666 Ring-Ready sofia/internal/sip:399@192.168.0.101:5063! 2013-11-19 17:17:22.498112 [NOTICE] sofia.c:6326 Channel [sofia/internal/sip:399@192.168.0.101:5063] has been answered 2013-11-19 17:17:29.518117 [NOTICE] switch_channel.c:968 New Channel sofia/internal/399(a)192.168.0.3 [60e39e56-5168-11e3-9b17-1f655b7322d0] -- Then I hit the transfer key on 399 to go to 300 2013-11-19 17:17:29.738117 [INFO] mod_dialplan_xml.c:557 Processing Test <399>->300 in context default 2013-11-19 17:17:29.738117 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:300@192.168.0.102:5060 [61056d4c-5168-11e3-9b2a-1f655b7322d0] 2013-11-19 17:17:29.798118 [NOTICE] sofia.c:5666 Ring-Ready sofia/internal/sip:300@192.168.0.102:5060! 2013-11-19 17:17:29.798118 [NOTICE] mod_sofia.c:2746 Ring-Ready sofia/internal/399(a)192.168.0.3! 2013-11-19 17:17:29.798118 [NOTICE] switch_ivr_originate.c:521 Ring Ready sofia/internal/399(a)192.168.0.3! -- As soon as I hit the transfer key again to finalise the transfer, everything hangs up 2013-11-19 17:17:30.978119 [NOTICE] sofia.c:6381 Hangup sofia/internal/399(a)192.168.0.3 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2013-11-19 17:17:30.978119 [NOTICE] switch_ivr_originate.c:3351 Hangup sofia/internal/sip:300@192.168.0.102:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2013-11-19 17:17:30.978119 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2013-11-19 17:17:30.978119 [INFO] mod_dptools.c:3055 Originate Failed. Cause: ORIGINATOR_CANCEL 2013-11-19 17:17:30.988118 [NOTICE] switch_core_session.c:1517 Session 190 (sofia/internal/sip:300@192.168.0.102:5060) Ended 2013-11-19 17:17:30.988118 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/sip:300@192.168.0.102:5060 [CS_DESTROY] 2013-11-19 17:17:30.988118 [NOTICE] switch_core_session.c:1517 Session 189 (sofia/internal/399(a)192.168.0.3) Ended 2013-11-19 17:17:30.988118 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/399(a)192.168.0.3 [CS_DESTROY] 2013-11-19 17:17:32.028118 [NOTICE] switch_ivr.c:1801 Transfer sofia/internal/5551212(a)192.168.0.10 to XML[300@public] 2013-11-19 17:17:32.048118 [NOTICE] switch_ivr_bridge.c:685 Hangup sofia/internal/sip:399@192.168.0.101:5063 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-11-19 17:17:32.068118 [NOTICE] switch_core_session.c:1517 Session 188 (sofia/internal/sip:399@192.168.0.101:5063) Ended 2013-11-19 17:17:32.068118 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/sip:399@192.168.0.101:5063 [CS_DESTROY] 2013-11-19 17:17:32.068118 [NOTICE] switch_cpp.cpp:637 Hangup sofia/internal/5551212(a)192.168.0.10 [CS_ROUTING] [NORMAL_CLEARING] 2013-11-19 17:17:32.068118 [NOTICE] switch_core_session.c:1517 Session 185 (sofia/internal/5551212(a)192.168.0.10) Ended 2013-11-19 17:17:32.068118 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/5551212(a)192.168.0.10 [CS_DESTROY] Any advice would be appreciated. Benoit

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avmd only works if sound is being sent.
by Andrew 19 Nov '13

19 Nov '13

Andrew I am trying to build an IVR using c# and mod_managed. I have a need to do this in c# to replace a current stack. I start the program using String err = ""; freeswitch.switch_core_set_globals(); uint flags = (uint)(switch_core_flag_enum_t.SCF_USE_AUTO_NAT); switch_status_t res = freeswitch.switch_core_init(flags, switch_bool_t.SWITCH_TRUE, ref err); search_bind = FreeSWITCH.SwitchXmlSearchBinding.Bind(xml_search, switch_xml_section_enum_t.SWITCH_XML_SECTION_CONFIG); event_bind = FreeSWITCH.EventBinding.Bind("SampleClient", switch_event_types_t.SWITCH_EVENT_ALL, null, event_handler,true); freeswitch.switch_core_init_and_modload(flags, switch_bool_t.SWITCH_TRUE, ref err); I am making a call using string Route = "{ignore_early_media=true,origination_caller_id_number=" + CallerIDNumber + ",origination_caller_id_name=" + CallerIDName + ",originate_timeout=20}sofia/softphone/" + NumberToDial; ManagedSession leg_b = new ManagedSession(Route); Then I am setting leg_b.Execute("avmd", "start"); after the call connects. I have found if I have mod_portaudio and have the audio play on my computer the beep will work but I am not making a softphone but a IVR. I think the key is freeswitch.xml but no clue how to fix it. One thing found out is if I am playing audio while the beep happen the even works but if I call and do nothing the beep never happens. Andrew

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SDReporter Integration with FS
by Deon Vermeulen 19 Nov '13

19 Nov '13

Hi I've successfully setup SDReporter, FreeSWITCH to export CDRs using cdr_csv with the Asterisk template. I've setup a cron to "automatically" push the csv file(s) to SDReporter $HOME_DIR/SDReporter/CDRConverter-4.0.8/data/INPUT/Asterisk/1/ I've tried different options in the format of the CDRs ,but no matter what I do I keep getting an " CDRFieldChannelAsterisk(String): wrong Asterisk channel. " ERROR. I've removed the "${channel_name}","${bridge_channel}" variables, but still I get the above error. I currently run FreeSWITCH Version 1.5.6b+git~20131018T052734Z~e0054af96f. Has anyone successfully got SDReporter working with FS? Thanks for any assistance. -- Kind Regards

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Re: [Freeswitch-users] Transferring calls - can Freeswitch instruct the up-stream provider to do the transfer?
by Clive Lansink 19 Nov '13

19 Nov '13

Thanks Brian. I think in this case the upstream provider provides virtual PBX services to my client, but it could equally well be the client's PBX. In this case Freeswitch is not the PBX, and what I want to achieve is that when the IVR application wants to transfer the call, instead of it routing through Freeswitch, I want Freeswitch to tell the provider PBX to take the call back and route it to where it now needs to go. Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Brian West <brian(a)freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Transferring calls - can Freeswitch instruct the up-stream provider to do the transfer? Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Tue, 19 Nov 2013 12:37:35 -0600 yes, see deflect vs redirect on the wiki, deflect sends a refer and the redirect sends a 302, I dont know of any provider up stream that will take a refer and act on it. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On 2013-11-18, at 4:46 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hi everyone. > > I have an IVR application written in Python that utilises Freeswitch to handle all the SIP and VOIP stuff. However in this case, Freeswitch is really only the end point and is not the client's PBX system. The client would like an option that will cause the incoming call to be transferred to a specified destination. The destination is an outside number and not a Freeswitch extension. > > I am playing with this now. I believe I can use the API originate command to originate a call to the specified destination and use the bridge application to bridge them. But even if that works, it would mean the incoming call is still being routed right through Freeswitch and out again to the destination. > > Now is there a way I can tell Freeswitch to issue like a call transfer command to the up-stream provider so Freeswitch can then hang up and the call can be handled elsewhere? > > In the old days, and in fact I have done this in a much earlier version of this same IVR application, I would have sent a flash to the outside world, wait for dialtone then dial the destination number, wait for ring and then hang up. Can anyone advise how would I do something like that now? > > Cheers. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 Error: unable to create file c:\clansink-documents\mail\messages\20131120\m201311200752080018\c:\clansink-documents\mail\messages\20131120\m201311200752080018-001.txt for attachment signature.asc. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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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Re: [Freeswitch-users] Transferring calls - can Freeswitch instruct the up-stream provider to do the transfer?
by Brian West 19 Nov '13

19 Nov '13

yes, see deflect vs redirect on the wiki, deflect sends a refer and the redirect sends a 302, I don’t know of any provider up stream that will take a refer and act on it. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On 2013-11-18, at 4:46 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hi everyone. > > I have an IVR application written in Python that utilises Freeswitch to handle all the SIP and VOIP stuff. However in this case, Freeswitch is really only the end point and is not the client's PBX system. The client would like an option that will cause the incoming call to be transferred to a specified destination. The destination is an outside number and not a Freeswitch extension. > > I am playing with this now. I believe I can use the API originate command to originate a call to the specified destination and use the bridge application to bridge them. But even if that works, it would mean the incoming call is still being routed right through Freeswitch and out again to the destination. > > Now is there a way I can tell Freeswitch to issue like a call transfer command to the up-stream provider so Freeswitch can then hang up and the call can be handled elsewhere? > > In the old days, and in fact I have done this in a much earlier version of this same IVR application, I would have sent a flash to the outside world, wait for dialtone then dial the destination number, wait for ring and then hang up. Can anyone advise how would I do something like that now? > > Cheers. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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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mod_lcr and 302 (Moved temporarily)
by Armando Kumbara 19 Nov '13

19 Nov '13

I have configured the diaplan to make a database query using the lcr module and respond with a 302 (moved temporarily) the destination number with a new prefix to the gateway that is sending the call. The process of manipulatinfg the digits works fine and the 302 message is generated bur the call fails because of a problem in the 302 message below: * SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 192.168.8.23:5060;rport=5060;branch=z9hG4bK-528b3362-c488a182-4cc5 Max-Forwards: 50 From: "Armando Kumbara"<sip:1234567@192.168.8.23 <sip%3A1234567(a)192.168.8.23>>;tag=45826b5f-13c4-528b3362-c488a182-473c To: <sip:9903645879714@95.107.130.71:5080 <http://sip:9903645879714@95.107.130.71:5080>>;tag=vSU04ySty4yvQ Call-ID: 12b6c96c-45826b5f-13c4-528b3362-c488a182-670e(a)192.168.8.23 <12b6c96c-45826b5f-13c4-528b3362-c488a182-670e(a)192.168.8.23> CSeq: 1 INVITE Contact: "unknown" <sip:9902645879714@192.168.8.26:5060 <http://sip:9902645879714@192.168.8.26:5060>> User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131104T192948Z~384ddf1517~32bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=31;text="NORMAL_UNSPECIFIED" Content-Length: 0 Remote-Party-ID: "9903645879714" <sip:9903645879714@192.168.8.25 <sip%3A9903645879714(a)192.168.8.25>>;party=calling;privacy=off;screen=no* Contact: "unknown" <sip:9902645879714@192.168.8.26:5060> should be Contact: "9902645879714" <sip:9902645879714@192.168.8.26:5060>. the dialplan entry for theDB query is the following: < *condition field="destination_number" expression="9903645879714"> <action application="lcr" data="$1 test"/> <action application="capture" data="mynum|${lcr_auto_route}|([^/]*)$"/> <action application="redirect" data="${mynum[1]}@192.168.8.26:5060 <http://192.168.8.26:5060>>"/> </condition>* Basically it should receive the call and query the db, capture the last digits after the "/" from the returned string and return it to the gaetway via the 302 message. Can you please advice me how i can fix this issue? Is there any way i can fix the contact information from the dialplan?

1 0

read_codec and write_codec xml_cdr
by FERNANDO VILLARROEL 19 Nov '13

19 Nov '13

Dear All. I am trying to use xml_cdr for save the cdr in backend (postgresql). My problem is only the answered calls have the read_codec and write_codec codec variables. Any idea why the cancel calls have not this variables on xml_cdr. I appreciate some help.

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xml_curl: initiator type
by vad fabi 19 Nov '13

19 Nov '13

Hi all, How to indentify call initiator type (gateway or locally registered user) then freeswitch sends xml_curl dialplan request ? As i see all post variables almost same for both types.

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Ignore busy signal when dialling out on an FXO port
by Oren Hecht 19 Nov '13

19 Nov '13

Hi, I'm new to freeswitch so please excuse me if I'm not using the right terms. I have freeswitch connected to a POTS line using an FXO port, and I am trying to use it to dial out without success. The POTS line requires me to dial '9' first to get a line for dialing out, and as long as I don't press 9 it gives a 'busy' tone. I am trying to bridge an extension to freetdm port with adding a '9' prefix, but I get "Failed to open channel" and the reason is "NORMAL_CIRCUIT_CONGESTION". When I look at 'ftdm dump' I see: "last_state: BUSY". I guess the reason for failure is that freeswitch hears the busy tone and thinks the line is not free and aborts. I also thought this is related to the early media signaling so I tried adding {ignore_early_media=true} thinking it will ignore the busy tone, but it dodn't help. How can I tell freeswitch to ignore the busy signal and just dial the destaination number, (the '9' prefix will give it a free line)? Thank you very much for the help, Oren Hecht.

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