July 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

simonics & freeswitch inbound call immediately disconnects when answered
by jungle boogie 02 Aug '17

02 Aug '17

Hello All, I want to have my google voice telephone number registered with freeswitch via simonics.com. I have my account already setup and I've used it with sip credentials without freeswitch so I know it's setup correctly. The problem I'm having is that as soon as the call is answered, it's immediately disconnects. /usr/local/freeswitch/conf/sip_profiles/external/googlev.xml <include> <gateway name="GVGW"> <param name="username" value="GV12132935740"/> <param name="from-domain" value="gvgw.simonics.com"/> <param name="password" value="thepassword"/> <param name="extension" value="12130000000"/> <param name="proxy" value="gvgw.simonics.com"/> <param name="register" value="true"/> <param name="register-transport" value="tcp"/> </gateway> </include> /usr/local/freeswitch/conf/dialplan/public/04_inbound_did.xml <include> <extension name="gv_public_did"> <condition field="destination_number" expression="^12130000000$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="2001 XML default"/> </condition> </extension> </include> Here's a link to the call log: https://pastebin.freeswitch.org/view/79a0e071 Anyone notice something wrong as to why the call would immediately disconnect when answered? Thanks!

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Question about DNS handling on FS
by Jospeh Waite 01 Aug '17

01 Aug '17

Hi Guys Got a question, I’m wondering if anyone can answer. I am wondering how FS handles DNS resolution. I assume that if I set the TTL on the DNS entry to say 30 seconds that FS will accept that value and do a DNS lookup every 30 seconds? Is that correct. My second question is one a cll is in motion to a DNS defined endpoint, i.e. call bridged to a SIP endpoint by domain name, would it keep the IP for the call or the domain name? So would it re-DNS query when it comes to sending the BYE message at end of call if it was over 30 seconds, assuming the above 30sec TTL or would it keep the name/IP mapping for the whole call? Regards

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Reinvite Issue / no SDP
by Stefan Kainz 1012 01 Aug '17

01 Aug '17

Hello, We're using Freeswitch 1.4.18 on Debian 8. We ran into a problem with internal call transfers, when the PBX sends a re-invite without SDP offer. The Problem is as follows: 1) PSTN -> Freeswitch -> PBX -> Call is answered on extension 100, Audio in both directions 2) Extension 100 does an unattended transfer to extension 101 2.1) PBX sends a re-invite without SDP. Freeswitch offers SDP in OK message 3) PBX acknoledges the OK message 4) Freeswitch is sending media as announced in SDP Offer, but PBX doesnt "receive" the audio. Also, no media from the PBX. Since we're a little desperate, we already tried the following settings: 1) 3pcc-enable true/false/proxy 2) inbound-late-negotiation is set to true 3) renegotiate-codec-on-reinvite true / false 4) disable-transfer true / false 5) enable-soa true / false ( it acutally does work when set to false, but the call gets hangup after a few seconds of audio - ack timeout ) We also tried to enable bypass_media, so that freeswitch doesnt mess with SDP. So, the SDP Offer is generated by a Sonus Media Gateway - still no luck. Any of you had some experience with such behavior? We would be very thankful for every advice! have a nice day! Stefan Kainz

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DTMF not working from yealink
by Ketan Kothari 31 Jul '17

31 Jul '17

Hello All, I have installed freeswitch 1.6.19 and trying to login in voicemail box 7777 that time DTMF is not working from my yealink and zoiper phone. In same freeswitch when i called from Grandstrem phone its DTMF working fine. Please give me suggesiton to resolve DTMF issue from yealink. Working_DTMF_Grandstream_logs https://pastebin.com/DMnDKjrR Not working_DTMF_Yealink_logs https://pastebin.com/iYYi6Kta

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Working example for presence_in
by Peter Steinbach 30 Jul '17

30 Jul '17

Hallo, I want to make a BLF lamp on the phone turn red/green/flash based on day/night info According to https://freeswitch.org/confluence/display/FREESWITCH/PRESENCE+IN+event+exam… I am sending here a presence info for 200 (ringing) sendevent PRESENCE_IN proto: sip from: 200(a)my.domain.com login: 200(a)my.domain.com event_type: presence alt_event_type: dialog Presence-Call-Direction: outbound answer-state: early via ESL an get a reply +OK 74a91ebc-7aaa-4f04-8fd1-5158836f594d Fs_cli shows freeswitch(a)my.domain.com> /events plain PRESENCE_IN +OK event listener enabled plain RECV EVENT Command: sendevent PRESENCE_IN proto: sip from: 200(a)my.domain.com login: 200(a)my.domain.com event_type: presence alt_event_type: dialog Presence-Call-Direction: outbound answer-state: early Event-UUID: 7a320e1e-8455-4804-b878-66c83914976e Event-Name: PRESENCE_IN Core-UUID: 81230f28-a930-4777-bdc5-d3df670c62ff FreeSWITCH-Hostname: my.domain.com FreeSWITCH-Switchname: my.domain.com FreeSWITCH-IPv4: 192.168.1.9 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2017-07-30 23:23:11 Event-Date-GMT: Sun, 30 Jul 2017 21:23:11 GMT Event-Date-Timestamp: 1501449791174168 Event-Calling-File: mod_event_socket.c Event-Calling-Function: parse_command Event-Calling-Line-Number: 2256 Event-Sequence: 88094 I also see a Notify message to 209 (as 209 was subscribing to 200) NOTIFY sip:209@192.168.178.121:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.178.9;rport;branch=z9hG4bKXKDD7j4B9SBeD. Max-Forwards: 70. From: <sip:200@my.domain.com;user=phone>;tag=svXxoq1TseLS. To: <sip:209@my.domain.com;user=phone>;tag=1852934565. Call-ID: 470672824-5060-5(a)192.168.178.121. CSeq: 911343401 NOTIFY. Contact: <sip:200@192.168.178.9:5060>. User-Agent: Freeswitch. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: dialog. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.win$ Subscription-State: active;expires=3214. Content-Type: application/dialog-info+xml. Content-Length: 156. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="123" state="full" entity="sip:200@my.domain.com"> </dialog-info> But the message sent is always the same (state=full), and does not reflect the answer state in the message (early, confirmed, terminated). Anybody has a clue, what's wrong here or has a better working example? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Is there Sampling rate conversion CPU penalty?
by Muhammad Naseer Bhatti 29 Jul '17

29 Jul '17

Hi, I am streaming a wav file who’s sampling rate is 44100 and FreeSWITCH says sample rate 44100 doesn't match requested rate 8000. So I think it’s doing the conversion here. If so, is there any CPU performance penalty while doing the conversion? What’s the best practice here should I change the sampling rate of the file or just leave it like that? -- Sent with Airmail

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Freeswitch 1.6.19: make current fails after updating
by chris 28 Jul '17

28 Jul '17

Hello List, my problem is that the build after updating fails. This is on a Mac Mini 2012 running OSX 10.11.6, XCode 7.3 and up to date brew. I follow the instructions from <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Manual+Installat…>. The previous installed version is: FreeSWITCH Version 1.6.17+git~20170421T211727Z~ab1f8eae62~64bit (git ab1f8ea 2017-04-21 21:17:27Z 64bit) mini11:freeswitch chris$ pwd /usr/local/src/freeswitch mini11:freeswitch chris$ git pull && make current ... config.status: libs/xmlrpc-c/xmlrpc_amconfig.h is unchanged config.status: executing depfiles commands config.status: executing libtool commands cd src/mod/languages/mod_lua/lua && make clean make[1]: *** No rule to make target `clean'. Stop. make: *** [lua-reconf] Error 2 mini11:freeswitch chris$ ls -la src/mod/languages/mod_lua/lua total 0 drwxr-xr-x 3 chris admin 102 21 Jul 00:14 . drwxr-xr-x 19 chris admin 646 21 Jul 00:15 .. drwxr-xr-x 34 chris admin 1156 25 Mär 22:18 .deps Complete output from build is on <http://www.gcjd.org/temp0717/freeswitchbuild.txt> I did an update before some time ago the same way and it worked flawless. How can I proceed without loosing my config? Thanks for any advice, Chris

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Sofia Recover
by Joseph Waite 28 Jul '17

28 Jul '17

Hi Guys I am setting up Freeswitch HA and have it working, and freeswithc is performing recover when I issue sofia recover I am using a shared IP so the IP doesn’t change, however FreeSwitch issues a re-invite when it performs the recovery. This is not needed in a shared IP situation and breaks some clients/carriers. Is there a setting to tell freeswitch to just re-start media and not send the re-invite. Regards

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Double NAT
by Thomas Chen 27 Jul '17

27 Jul '17

i am trying this set up (double NATs) SIP CLIENT (10.1.2.x) ===( NAT1 )===>[ FS1 ]===>( NAT2 )===>[ FS2 ] i am trying to make a call from SIP CLIENT... to FS2 (twilio) for FS1 configuration... $${domain} is set to IP address on NAT1 (10.1.2.1) and ext-rtp-ip and ext-sip-ip are set to IP address on NAT2 (192.168.0.x) side... the "sofia global siptrace on" is showing that requests are sent from FS1 sent to FS2, and FS1 is not getting any response... is there any info regarding double NATs ??? thanks

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Txfax not hanging up after FAX complete
by Daniel Greenwald 26 Jul '17

26 Jul '17

We are seeing cases where FS doesn't hang up after the FAX exchange complete. Version is: 1.6.18+git-20170621T132243Z~6e79667c0a~64bit (git 6e79667 2017-06-21 13:22:43Z 64bit) Any thoughts?

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