June 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

gsmopen one-way voice issue
by Jeff Chua 24 Sep '17

24 Sep '17

I'm configured freeswitch with gsmopen but can't seem to figure where why I only get voice out from SIP-client to mobile-phone but not voice from mobile to SIP client. When I tried to call from mobile-phone to IVR extension 5000 and press 2 for the standard echo test (9196), I could hear the echo back. When I tried to call 9195 for the five second delay echo test, I couldn't anything. I've add these settings in vars.xml as well. <X-PRE-PROCESS cmd="set" data="media_mix_inbound_outbound_codecs=true"/> <X-PRE-PROCESS cmd="set" data="inbound-late-negotiation=false"/> <X-PRE-PROCESS cmd="set" data="inbound-zrtp-passthru=false"/> <X-PRE-PROCESS cmd="set" data="disable-transcoding=false"/> <X-PRE-PROCESS cmd="set" data="inherit_codec=false"/> <X-PRE-PROCESS cmd="set" data="bypass_media=false"/> <X-PRE-PROCESS cmd="set" data="proxy_media=false"/> What am I missing? Thanks, Jeff

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FreeSWITCH on Raspberry board
by youssef elouam 20 Sep '17

20 Sep '17

hi all ; I am trying install freeSWITCH in raspberry board (Raspbian OS). but when I run "make && make install" I get as result : make: *** No targets. Stop. could you please help fix this issue. thanks in advance Bestregards --------------------------Youssef ELOUAM.

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Dockerfile for FreeSWITCH
by Brian West 18 Aug '17

18 Aug '17

https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/docker If anyone wishes to submit pull requests to improve these please do so at freeswitch.org/jira I would like to work with the community to improve and document the use cases correct. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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Forward fails with deflect and REFER to null
by Mario G 12 Aug '17

12 Aug '17

What are the bold deflect and refer to lines below telling me? This worked until last year, not sure if it’s related to 1.6, I can’t nail down a start date. A ring group rings. When one phone’s FWD button is pressed prior to a call or during a call the call is forwarded to VM extension 2921, and the other phones stopped ringing. This no longer works, the set phone does not ring, the call is not transferred to VM and the other phones keep ringing. I couldn’t find anything on the wiki (it only described using codes to turn FWD on/off), etc. to explain why this stopped working. Any help to get the phone forwarding working again is greatly appreciated. Thanks! Mario G YEALINK T48G DSS button is set to forward to 2921, pressing again turns it off. All other DSS keys such as call park, etc work fine. <extension name="vmonly"> <condition field="destination_number" expression="^2921$">  <action application="lua" data="/usr/local/FreeSwitch-scripts/known-people.lua CALLIN"/> <action application="answer"/> <action application="playback" data="${lua_vmmsg}"/>  <action application="set" data="skip_greeting=true"/> <action application="voicemail" data="default ${domain_name} 100"/> </condition> </extension> This log shows the only lines that contain 2921 in bold: 2016-11-19 16:27:26.673609 [INFO] mod_dialplan_xml.c:637 Processing Mario G <15051234567>->2921 in context public Dialplan: sofia/internal/15051234567(a)11.123.1.10 parsing [public->unloop] continue=false Dialplan: sofia/internal/15051234567(a)11.123.1.10 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/15051234567(a)11.123.1.10 Regex (PASS) [unloop] ${sip_looped_call}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/15051234567(a)11.123.1.10 Action deflect(${destination_number}) 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:286 (sofia/internal/15051234567(a)11.123.1.10) State Change CS_ROUTING -> CS_EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/15051234567(a)11.123.1.10) State ROUTING going to sleep 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/15051234567(a)11.123.1.10) Running State Change CS_EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/15051234567(a)11.123.1.10) State EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] mod_sofia.c:198 sofia/internal/15051234567(a)11.123.1.10 SOFIA EXECUTE 2016-11-19 16:27:26.673609 [DEBUG] switch_core_state_machine.c:328 sofia/internal/15051234567(a)11.123.1.10 Standard EXECUTE EXECUTE sofia/internal/15051234567(a)11.123.1.10 deflect(2921) 2016-11-19 16:27:26.673609 [DEBUG] sofia.c:8349 Process REFER to [(null)@2921]

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Adding video to audio call
by Gauri Kshirsagar 19 Jul '17

19 Jul '17

Hi, I am using Freeswitch version 1.9.0. I can make video calls. But when I try to add video to audio call it does not work. A makes audio call to B. Call is established. A adds video . ReINVITE sent to freeswitch has video added in SDP.Freeswitch sends 200 OK response for this INVITE to A which has video in SDP. But there is no ReINVITE being sent to B. I tried enabling renegotiate-codec-on-reinvite in vars.xml and also internal.xml Is this supported? If so is some other configuration required. Regards,

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Error sending queries to pgsql
by Antonio Silva 14 Jul '17

14 Jul '17

Hi all, i use pgsql in core and from time to time i see critical messages like fail to send query, example: [CRIT] switch_pgsql.c:255 Failed to send query (update sip_authentication set expires='1498726568',last_nc=364 where nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to database: server closed the connection unexpectedly This was recently, i did an update to current master the previous version was from April, not sure if it could be an error on FS o some other issue on my box.. PG is installed on the same server and the only thing i see from pg is "postgres[2236]: FATAL: invalid frontend message type 21", PG is installed on the same server, running on /dev/shm with the same prio as FS and the process never stopped. anyone has experience this error before? any idea what it could be the cause? Thanks, -- Saludos / Regards / Cumprimentos, António silva

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from Asterisk sip.conf to FS
by Valter Nogueira 06 Jul '17

06 Jul '17

I have an app that runs in Asterisk and I am trying port it to FS. Below, I pasted a small sip.conf that sums up all my needs. 1. We register out in a VOIP_PROVIDER 2. An external asterisk register in our asterisk as CLIENT01 3. Calls from CLIENT01 route to VOIP_PROVIDER 4. Yealink's, with ACL restrictions, also uses VOIP_PROVIDER to external calls 5. VOIP PROVIDER and Yealinks use one NIC (eth0) 6. CLIENT01 uses another NIC (eth1) In Asterisk all configs are keep together in sip.conf. However, in FS it seems I should spread out things in different configs parts. So: Should Yealink's accounts go into /usr/local/freeswitch/conf/directory/default? Should I put VOIP_PROVIDER in sofia's external profile? Should I create an additional external profile to CLIENT01? Where should I define CLIENT01 account to allow it register in? In /usr/local/freeswitch/conf/directory/default? Thanks for any help. Valter === SIP.CONF === [general] disallow=all context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls alwaysauthreject=yes allowguest=no ;voip provider register => 185999:1010@xxx.xxx.xxx.xxx:5060 [VOIP_PROVIDER] type=peer context=FDTRONCO_94 host=xxx.xxx.xxx.xxx port=5060 username=185999 authuser=185999 authname=185999 secret=XXXXXXXXXXXXXX fromdomain=xxx.xxx.xxx.xxx fromuser=185999 insecure=port,invite canreinvite=no nat=no disallow=all allow=ulaw:30 dtmfmode=rfc2833 ignoreregexpire=yes language=pt_BR call-limit=9999 ;EXTERNAL ASTERISK ACCOUNT - CALLS GOES THRU VOIP_PROVIDER [CLIENT01] type=peer context=FDTRONCO_60 host=dynamic port=5060 username=CLIENT01 authuser=CLIENT01 authname=CLIENT01 secret=XXXXXXX insecure=port,invite canreinvite=no nat=yes disallow=all allow=g729 dtmfmode=rfc2833 ignoreregexpire=yes language=pt_BR call-limit=440 ;yealinks ip phones [1000] type=friend secret=XXXX host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.0.0/16 username=1000 context=FASTDIALER_RAMAIS callerid=testefone <1000> requirecalltoken=no nat=no canreinvite=no qualify=yes disallow=all allow=alaw call-limit=2 dtmfmode=RFC2833 language=pt_BR Callgroup=1 pickupgroup=1 [2012] type=friend secret=XXXX host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.0.0/255.255.0.0 username=2012 context=FASTDIALER_RAMAIS callerid=Telefone sem fio <2012> requirecalltoken=no nat=no canreinvite=no qualify=no disallow=all allow=alaw call-limit=2 dtmfmode=RFC2833 language=pt_BR Callgroup=1 pickupgroup=1

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Verto calls
by Agustí Ubalde 06 Jul '17

06 Jul '17

Hi all, I am trying to call from Verto extension to another Verto extension. *Both are successfully registered* (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): *EXECUTE verto.rtc/1001 bridge()* Regards, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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Conditional call forward
by Ashwin Rath 06 Jul '17

06 Jul '17

Hi I have an extension which has call forward setup BUT i would like the call forward to work only when dialed from a certain number and not from another numbers. Can this be achieved ? -- Ashwin Kumar Rath

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VOIP-GSM Gateway configuration Error
by Deepika Yadav 06 Jul '17

06 Jul '17

Hi, I have been trying for a while the configuration of a VOIP-GSM gateway that I have bought but the registration fails every time. Error at Freeswitch console 2017-06-27 16:27:10.191913 [WARNING] sofia_reg.c:2906 Can't find user [*testadmin(a)192.168.2.71 <testadmin(a)192.168.2.71>*] from *192.168.2.69* You must define a domain called '192.168.2.71' in your directory and add a user with the id="*testadmin*" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2017-06-27 16:27:14.071916 [ERR] sofia_reg.c:2447 MySIP Failed Registration with status Request Timeout [408]. failure #1 2017-06-27 16:27:14.111913 [WARNING] sofia_reg.c:505 MySIP Failed Registration [408], setting retry to 30 seconds. Here *'192.168.2.71*' is the local IP of the Freeswitch server and *192.168.2.69* that of the gateway attached to the LAN port of the Freeswitch server. I have put SIP UserID in the gateway admin panel as "*testadmin*" alongwith its passoword. my external SIP profile is as follows: <include> <gateway name="MySIP"> <param name="username" value="testadmin"/> <param name="realm" value="192.168.2.69"/> <param name="from-user" value="testadmin"/> <param name="from-domain" value="192.168.2.69"/> <param name="password" value="****"/> <param name="extension" value="5888"/> <param name="proxy" value="192.168.2.69"/> <param name="expire-seconds" value="120"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> </gateway> I have tried number of permutations and combinations but the gateway is not getting registered. Regards Deepika https://www.iiitd.edu.in/~deepikay/

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