December 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP.js / mobile devices / speakers
by Thorsten Göllner 07 Dec '16

07 Dec '16

Hi, this is maybe not the best maillinglist for my problem but I did not find a better one :) I am playing around with SIP.js and freeswitch as backend. Works fine so far. But is it possible to use the inernal speaker of a mobile device when using SIP.js? If I use it with my android phone, always the external (loud) speaker is active but I just want to use it like a normal voice call (holding my device near my ears). Is there any option to switch speakers perhaps? Thanks in advance, Thorsten

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Bridge to gateway with prefix alters Destination number
by Gregor Nanger 07 Dec '16

07 Dec '16

I have one funny challenging problem and do not know how to solve it. I am bridging call to gateway and provider requires that I add prefix. For example: }sofia/gateway/myprovider/aa15551231234 This is everything ok and it works, but then I get update display on softphone from 5551231234 to aa15551231234 and in call logs I get same number: aa15551231234. And then it looks like user dialed aa15551231234 . How to tell freeswitch not to update display and call log? I tried with ignore_update_display, but with no luck. Best regards, Gregor

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Re- FreeSWITCH 1.6.13 on CentOS 6.8 and mod_flite build issue
by Andrew Keil 07 Dec '16

07 Dec '16

To FreeSWITCH users, I understand that there is a need to move to CentOS 7 for FreeSWITCH, however my client cannot do this currently. I have tried to get mod_flite to build inside FreeSWITCH 1.6.13 on CentOS 6.8 I get the following when running FreeSWITCH make after ./configure: making all mod_flite make[3]: Entering directory `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite' Makefile:800: *** You must install libflite-dev to build mod_flite. Stop. make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod/asr_tts/mod_flite' make[2]: *** [mod_flite-all] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[1]: *** [mod_flite] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch/src/mod' make: *** [mod_flite] Error 2 Since there is no libflite-devel package for CentOS 6 I did the following successfully: cd /usr/src wget http://files.freeswitch.org/downloads/libs/flite-2.0.0.tar.gz tar -xzvf flite-2.0.0.tar.gz cd flite-2.0.0 ./configure --enable-shared --prefix=/usr/local make && make install This worked OK and setup everything inside /usr/local/lib However when I try to run ./configure and make inside the FreeSWITCH source directory again I get the same error above. Any ideas? Kind Regards, Andrew Keil Visytel Pty Ltd [visytellogo] www.visytel.com<http://www.visytel.com/> Mobile: +61 (0) 429 076 431 Skype: visytel This e-mail message may contain confidential and commercially sensitive information. If you are not the intended recipient, any use, interference with, disclosure or copying of this material is unauthorised and prohibited. If you receive this e-mail message by mistake, please contact Visytel Pty Ltd on +61 429 076 431.

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originate and origination_uuid
by Tihomir Culjaga 06 Dec '16

06 Dec '16

Hello everyone, i'd like to clarify one small thing: in an originate with multiple dialstrings separated by "|" ( meaning sequential dialing ) ... if i set origination_uuid as session variable, can i have the same uuid on every session/channel. something like that: bgapi originate {originate_timeout=60} [origination_uuid=c88a737e-d713-4e47-be56-e384df470b44]sofia/gateway/gatewayA/1222333444| [origination_uuid=c88a737e-d713-4e47-be56-e384df470b44]sofia/gateway/gatewayB/1222333444| [origination_uuid=c88a737e-d713-4e47-be56-e384df470b44]sofia/gateway/gatewayC/1222333444 &application so in other words, when executing originate command, FS, will generate a separate thread for the call and within the thread it will generate these sessions. So, is it safe to set the same uuid on all session that are being dialed sequentially? Do sessions start and end sequentially ? Appreciate any answer on this. Regards, Tihomir.

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Freeswitch CA certs
by Vladyslav Zakhozhai 06 Dec '16

06 Dec '16

Hi, I wonder why freeswitch does not use system's CA certificates for SSL/TLS verification? I have Let's Encrypt certificate on the web secrive side and I have troubles with http_cache over https (server's cert cant be verified). In freeswitch'es wiki I've read ( https://wiki.freeswitch.org/wiki/Mod_http_cache#Installing) For HTTPS support, grab the latest CA certs from http://curl.haxx.se/ca/cacert.pem and install in /usr/local/freeswitch/conf. An older copy is also available in freeswitch/src/mod/applications/mod_http_cache/conf/cacert.pem. First, what is haxx.se? I do not know and it is not an authority for me. So do I need to bundle cacers myself? Why do I need to do it (for verification of SSL/TLS of external service with real certificate) if I already have it on my system? Tell me please where I am wrong? Thanks in advance. -- С уважением, Владислав Захожай

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T.38 RTP Poll failed with error: 9 [Bad file descriptor]
by Peter Steinbach 06 Dec '16

06 Dec '16

Hello, I have a strange error when proyy-media T38 faxes. Freeswitch is behind NAT (internal 192.168.12.248, external 5.10.xxx.yyy), other, receiving Server is also a Freeswitch A connection is made via G.711A, we hear the machines negociating, then we receive a T.38 reinvite from the remote Freeswitch server. The local ATA 30@ reponds with T38 SDP, Freeswitch is passing it to the remote site (and there the error occurs). See error in bold: 2016-12-01 15:28:55.534228 [DEBUG] sofia.c:6937 Channel sofia/internal/30(a)192.168.12.248 entering state [completing][200] 2016-12-01 15:28:55.534228 [DEBUG] sofia.c:6944 Duplicate SDP v=0 o=30 8002 8001 IN IP4 192.168.12.99 s=SIP Call c=IN IP4 192.168.12.99 t=0 0 m=image 5012 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:500 a=T38FaxMaxDatagram:500 a=T38FaxUdpEC:t38UDPRedundancy 2016-12-01 15:28:55.534228 [DEBUG] sofia.c:6937 Channel sofia/internal/30(a)192.168.12.248 entering state [ready][200] 2016-12-01 15:28:55.554501 [DEBUG] mod_sofia.c:2088 Responding with 200 [OK] 2016-12-01 15:28:55.554501 [DEBUG] switch_core_media.c:8955 sofia/external/0403801xxxxxxx image media sdp: v=0 o=FreeSWITCH 1480572959 1480572961 IN IP4 5.10.xxx.yyy s=FreeSWITCH c=IN IP4 5.10.xxx.yyy t=0 0 m=image 29544 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:500 a=T38FaxMaxDatagram:500 a=T38FaxUdpEC:t38UDPRedundancy *2016-12-01 15:28:55.554501 [ERR] switch_rtp.c:6420 Poll failed with error: 9 [Bad file descriptor]* 2016-12-01 15:28:55.554501 [DEBUG] switch_ivr_bridge.c:712 sofia/external/0403801xxxxxxx ending bridge by request from read function 2016-12-01 15:28:55.554501 [DEBUG] switch_ivr_bridge.c:785 BRIDGE THREAD DONE [sofia/external/0403801xxxxxxx] 2016-12-01 15:28:55.554501 [NOTICE] switch_ivr_bridge.c:889 Hangup sofia/external/0403801xxxxxxx [CS_EXCHANGE_MEDIA] [NORMAL_CLEAR Some background Info: * Freeswitch runs in LXC environment * Freeswitch runs as non-priviledged user. What is the cause of this? Maybe a problem with port 29544? But port 29544 was already previously used with G.711A. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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log X-AUTH-IP instead of source IP
by Roman Dissauer 06 Dec '16

06 Dec '16

Hi, I’m using Kamailio as a transparent SIP Proxy in front of my Freeswitch which is configured to use Digest Auth for the clients. I already set up X-AUTH-IP headers on Kamailio and apply-proxy-acl in internal FS profile. To use fail2ban I need to get X-AUTH-IP information instead of source IP in the log messages: --> "SIP auth challenge (INVITE|REGISTER) on sofia profile 'internal' for [xxxxxx] from ip x.x.x.x" (proxy IP) Is there a way to get the original source IP into the SIP auth challenge log message? Thank you for your help!! Best Regards, Roman

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RE- FreeSWITCH originate api command question
by Andrew Keil 06 Dec '16

06 Dec '16

To FreeSWITCH Users, I have a need to have the FreeSWITCH originate API command to return ASAP (and not wait for 200 OK SDP (ie. CONNECT)). My FreeSWITCH version is the latest production release 1.6.13. This is easy to test from fs_cli: originate {call_timeout=45,ignore_early_media=true,return_ring_ready=true}sofia/gateway/mygateway/xxxxxxxxxx;user=phone 5000 Where: xxxxxxxxxx = outbound phone number After looking at my SIP Trunk provider's SIP Flow the following happens: Tx: INVITE SDP Rx: 100 Trying Rx: 183 Session Progress SDP Tx: RTP {early media} Rx: 180 Ringing Rx: 200 OK SDP Tx: ACK Rx: BYE Tx: 200 OK Based on the above flow, I understand that since 183 SDP is received then the 180 Ringing is not used by FreeSWITCH (based on reading other comments on this topic), therefore what settings should I use inside the originate command in order for the originate API call to return and start service 5000 flow when 183 SDP is received. My questions: 1) Should ignore_early_media=true or ignore_early_media=false or ignore_early_media=ring_ready for originate to return when 183 SDP is received? 2) Will return_ring_ready=true have no affect based on the above call flow? 3) Are there any other settings that I should use to enable originate to return when 183 SDP is received? Kind Regards, Andrew Keil

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A primitive way of getting VoLTE recordings in FreeSwitch. Looking for better ideas.
by Oivvio Polite 05 Dec '16

05 Dec '16

I'm building a call recording tool (outgoing calls only) where the prime considerations are 1) audio quality and 2) ease of use. During the last years many mobile carriers have enabled VoLTE calling that operate at higher bitrates with better audio quality than traditional telephony. A common patterns seems to be that VoLTE is only enabled when the phone handsets on both ends supports it and both parties are on the same mobile carrier. To get that sweet HD audio in my call recordings I can't just run FreeSwitch with a regular SIP trunking provider. I've googled extensively and can't find any SIP trunking provider that claims VoLTE capability. A primitive way recording VoLTE calls with FreeSwitch would be to cook something together with mod_portaudio kind of like this: The user dials into FreeSwitch with a some custom HD audio client, probably using WebRTC. We capture this call leg at the server. Hooked up to the FreeSwitch box we have 3 audio cards. Connected to each audio card is a VoLTE capable cellphone on a diffrent mobile carrier (We pick the three most popular). The phones connect via the cellphone jacks. Each phone is also running a custom app that listens (via Wifi) for instructions from our FreeSwitch box. When the call arrives to FreeSwitch we check what carrier the the user is dialing. Hopefully one of our 3 connected phones also use that carrier. If so when instruct that phone to call the destination number and we connect the phone via mod_portaudio as the second leg of our call. This scheme as kind of Rube Goldberg and has many limitation. 1. If we want to support simultaneous calls on the same carrier we need more phones. 2. We loose audio quality in the recording when going from digital to analog. 3. Cellphones error prone, subject to freezes and what not. This is not something you want to put in a datacenter. So I'm wondering what other options might be for recording VoLTE calls. kind regards, Oivvio Polite

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E-mail a voicemail message
by William Colls 05 Dec '16

05 Dec '16

I am setting up the functions/procedures that will e-mail a voice mail message to a designated individual, and I have two questions. 1. Does FreeSWITCH read the message template each time it sends a message, or is the template loaded into memory at startup, and that copy used to build the message? 2. We use nullmailer as our MTA, and like freeswitch it does not run with root privilage. When the message file is written to the queue, it is written with privilage rw-------, and owner is freeswitch. Nullmailer can't read the message because it has insufficiant privilage. At the moment, we get around the problem by starting nullmailerwith UID of freeswitch, and that works, but it feels wrong to me. Is there anyway to set umask for freeswitch, so that the message is written with rw-rw----. Since both freeswitch and nullmaile are members of the group daemon, nullmailer would then be able to read the message queue while still running with its proper UID? -- William Colls & Associates VoIP Telephone Services Digital PBX and Telephone Systems

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