May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Mod callcenter with off-hook agents
by Alex Pierry 09 May '16

09 May '16

Hi guys, is there a way to implement mod callcenter with off-hook agents? Thanks, Alex. <http://pt.slideshare.net/marketingteclan> <http://blog.teclan.com.br/>

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WebRTC mod verto paid consulting
by Arsen Manukyan 09 May '16

09 May '16

Hello We need implement WebRTC So we need paid consulting for our system intial setup

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Manually set Contact user in gateway registration
by Jeff Pyle 09 May '16

09 May '16

Hello, I'm curious if it's possible to manually specify the user value in the Contact of an outbound gateway registration. In the documentation and examples I see where extra parameters can be set, but not the base user itself. - Jeff

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FreeSWITCH server as proxy or redirect or registrar or all together
by amani mansour 09 May '16

09 May '16

Hi all , I would like to ask you some questions: -Is there a book written with the frensh language ? -I need to know is there an active FreeSWITCH GUI ? I find freeWIKI but is not active . -FreeSWITCH is an VOIP server or SIP server . So i want to know ,in FreeSWITCH, where is the : Proxy server ,Registrar server ,and redirect server ,are they the moduls ? Please I need your help . Best Regards Amani

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Chatplan Invalid profile
by Cj B 09 May '16

09 May '16

Hi all, I’m trying to create a lua script to send an SMS message between phones and I keep getting this “invalid profile” error. I’m trying follow the “Sending a Message from a script” on the mod_sms confluence page. I found an old mailing list thread that said we need to "alias the domain or IP to the profile”. But I’m not exactly sure how to do that. Any hints? http://pastebin.com/uL3NpmMN <http://pastebin.com/uL3NpmMN> Thanks, Cj B

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Invalid file format with Codec AMR
by Pradeepa 09 May '16

09 May '16

Hi, I want to play amr file in freeswitch IVR. But I get Invalid file format [amr] error. Any suggestion to solve this problem? freeswitch log ------------------- 2016-05-05 22:36:02.151586 [INFO] mod_dptools.c:1260 Sending early media 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [AMR:120:8000:20:0]/[AMR:96:8000:20:12200] 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3190 Set Codec sofia/internal/1000(a)10.169.112.170 AMR/8000 20 ms 160 samples 12200 bits 2016-05-05 22:36:02.151586 [DEBUG] switch_core_codec.c:111 sofia/internal/ 1000(a)10.169.112.170 Original read codec set to AMR:96 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:5451 Set 2833 dtmf send/recv payload to 101 2016-05-05 22:36:02.151586 [DEBUG] mod_erlang_event.c:156 Sending event CODEC to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3449 AUDIO RTP [sofia/internal/1000(a)10.169.112.170] 10.169.112.170 port 22020 -> 192.168.0.174 port 4006 codec: 120 ms: 20 2016-05-05 22:36:02.151586 [DEBUG] mod_erlang_event.c:156 Sending event CODEC to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.151586 [DEBUG] switch_rtp.c:2040 Starting timer [soft] 160 bytes per 20ms 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3716 Set 2833 dtmf send payload to 101 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3722 Set 2833 dtmf receive payload to 101 2016-05-05 22:36:02.151586 [DEBUG] sofia_glue.c:3749 sofia/internal/ 1000(a)10.169.112.170 Set rtp dtmf delay to 40 2016-05-05 22:36:02.151586 [NOTICE] sofia_glue.c:4360 Pre-Answer sofia/internal/1000(a)10.169.112.170! 2016-05-05 22:36:02.151586 [DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_PROGRESS_MEDIA to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.151586 [DEBUG] switch_channel.c:3368 (sofia/internal/ 1000(a)10.169.112.170) Callstate Change RINGING -> EARLY 2016-05-05 22:36:02.151586 [DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_CALLSTATE to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.151586 [DEBUG] mod_sofia.c:2891 Ring SDP: v=0 o=FreeSWITCH 1462490942 1462490943 IN IP4 10.169.112.170 s=FreeSWITCH c=IN IP4 10.169.112.170 t=0 0 m=audio 22020 RTP/AVP 120 101 a=rtpmap:120 AMR/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2016-05-05 22:36:02.151586 [DEBUG] switch_core_session.c:871 Send signal sofia/internal/1000(a)10.169.112.170 [BREAK] 2016-05-05 22:36:02.711592 [DEBUG] switch_core_session.c:1151 Send signal sofia/internal/1000(a)10.169.112.170 [BREAK] 2016-05-05 22:36:02.731593 [DEBUG] switch_ivr.c:614 sofia/internal/ 1000(a)10.169.112.170 Command Execute playback(abc_test/en/test.amr) EXECUTE sofia/internal/1000(a)10.169.112.170 playback(abc_test/en/test.amr) 2016-05-05 22:36:02.731593 [DEBUG] mod_erlang_event.c:156 Sending event CHANNEL_EXECUTE to attached session 69a98c32-134c-11e6-a976-6b5931cb8fb7 2016-05-05 22:36:02.731593 [ERR] switch_core_file.c:150 Invalid file format [amr] for [/usr/local/abc/sounds/en/us/callie/abc_test/en/test.amr]! amr prompt details ------------------------- Audio Format : AMR Format/Info : Adaptive Multi-Rate Format profile : Narrow band Duration : 16s 140ms Bit rate mode : Constant Bit rate : 12.8 Kbps Channel(s) : 1 channel Sampling rate : 8 000 Hz Bit depth : 13 bits Stream size : 25.2 KiB (100%) Thanks.

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Error linking
by Luis Daniel Lucio Quiroz 09 May '16

09 May '16

It is clear I am missing something. Any clue? *** Warning: Linking the shared library libfreeswitch.la against the *** static library libs/libvpx/libvpx.a is not portable! Making all in . make[2]: Entering directory `/builddir/build/BUILD/freeswitch-1.6.8' CC libfreeswitch_la-switch_version.lo CXXLD libfreeswitch.la *** Warning: Linking the shared library libfreeswitch.la against the *** static library libs/libvpx/libvpx.a is not portable! CC freeswitch-switch.o CCLD freeswitch ./.libs/libfreeswitch.so: undefined reference to `__cxa_pure_virtual' ./.libs/libfreeswitch.so: undefined reference to `__gxx_personality_v0' ./.libs/libfreeswitch.so: undefined reference to `operator new(unsigned long)' ./.libs/libfreeswitch.so: undefined reference to `operator delete(void*)' ./.libs/libfreeswitch.so: undefined reference to `vtable for __cxxabiv1::__class_type_info' collect2: error: ld returned 1 exit status make[2]: Leaving directory `/builddir/build/BUILD/freeswitch-1.6.8' make[1]: Leaving directory `/builddir/build/BUILD/freeswitch-1.6.8' -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH

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Dialed number after transfer
by Varsha Agarwal 08 May '16

08 May '16

Hi, I have a Call manager transferring call to FS server. I want FS server to transfer the call to a voice application on say extension 5001. When the call is transferred to voice application, voice application needs to retrieve the original dialed number used to make the call but since this call is transferred from FreeSwitch, original number gets lost and I get 5001 as dialed number. Is it possible to retain the dialed number? Thanks, Varsha

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Basic configuration falis
by Shlomi Schwartz 08 May '16

08 May '16

Hi, I'm struggling with some basic configuration. I'm trying to make a call using Verto Communicator. I have added 2 extensions (1000, 1008), and 2 inbound routes: - condition: destination_number (1008) - action: set effective_caller_id_name=user 1#${caller_id_name} - action:transfer 1008 XML my.ip.address But I always get a voice recording saying the extension is not available, I'm a lost. - *Disclaimer: I'm using FusionPBX on top Freeswitch.* -- This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this on behalf of the addressee you must not use, copy, disclose or take action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply email and delete this message. Thank you.

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Configure media till 180 ring is received
by Abhishake Agarwal 08 May '16

08 May '16

Hi, I need beep ring until 180 ring is not received or before early media. Please help me resolving this issue. Thanks Abhishake

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