June 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dialplan: destination_number: regex question
by Nate 06 Jul '17

06 Jul '17

Good day/evening everyone, Apologies for not being able to figure this out on my own. I've been searching and trying for several days to get inbound/outbound working but have yet to see success. At this stage I need help determining a proper regex expression for handling New Zealand phone numbers. For instance, there are three different ways of expressing numbers here in NZ: International: +64 22 333 4444 non-local: 022 333 4444 local: 333 4444 A couple questions related to the SIP proxy: If the SIP proxy is located in Hong Kong, but the phone number is a New Zealand number, does the location of the proxy have any impact on the number of characters in the string for inbound/outbound calls? Again, apologies for the rudimentary nature of these questions, but having nearly exhausted all other options (docs, searches, IRC), I am now spending a large amount of time guessing and trial and error without any progress. Many thanks for any feedback at all. Nate

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Switching profiles upon 3xx Redirection
by Rahul MathuR 05 Jul '17

05 Jul '17

Hello guys, Thanks for a wonderful product ! I'm loving it. I am working on a situation where FS is installed on a centos7 having 2 ip address internal and public. On internal, I receive INVITE and create a b-leg and send to a Redirect server. This server sends 3xx and populates Contact header. When I receive it, I can see that external profile has already been set. And I can't change it to send it to another profile. Is there a way to do that ? My external profile listens on internal ip. I tried different options mentioned in confluence page but to no avail. Am I missing anything or trying to accomplish something impossible ? Thanks in anticipation. Rahul

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Limit
by Shaun Stokes 04 Jul '17

04 Jul '17

Hi All, In our environment we restrict the number of current channels (using limit) per extension, however using attended transfer allows extensions to exceed the limit since leg a ends once the transfer completes which resets the limit to 0. I believe the solution is to apply the limit on leg b, however leg b is initiated via a bridge (with-in our outbound LUA script). How can we apply the limit to leg b, or is there a better solution? Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes(a)itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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mod_xml_curl performance questions
by Michael Avers 01 Jul '17

01 Jul '17

Hello, I'm planning on moving all our local lua-generated XML to use mod_xml_curl over HTTPS. However, the primary http server is not in the same data center as our FS instances. Ping time is about 50ms. Is this going to be an issue? Is it typical to run xml_curl hosted elsewhere? The web app itself is written in Go and is very fast. Just wondering about network latency aspect of it all. Thanks -Mike

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Audio issue in Amazon EC2 server
by Ketan Kothari 01 Jul '17

01 Jul '17

Hi All, I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 <https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2>* Once we tried to call outbound call getting *no audio in both side* also FreeSWITCH passing my server local IP as RTP-IP to gateway side, =========================== 2017/06/29 12:57:06.579360 128.22.54.32:5060 -> 88.22.45.123:5060 v=0 o=FreeSWITCH 123456456 1245210017 IN IP4 *192.168.1.101* s=FreeSWITCH c=IN IP4 *192.168.1.101* t=0 0 m=audio 31010 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ============== 128.22.54.32 = Amazon EC2 server public IP *192.168.1.101* = Amazon EC2 server local IP 88.22.45.123 = Gateway provider. Could you please help me on audio issue in local-ip selected as RTP_IP and passing to gateway side?

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Use FreeSWITCH to solve a problem and win a prize!
by Anthony Minessale 30 Jun '17

30 Jun '17

This year at the ClueCon Coder Games there is a category for the most innovative use of FreeSWITCH to solve a problem. The prize is a bar top arcade. Get started now! Check out the other challenges and prizes on our web site: http://cluecon.com/coder-games.html -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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DTMF 2833 not working when switching to dynamic payload type 96
by John Narduchi 30 Jun '17

30 Jun '17

Having some trouble getting DTMF to work where an INVITE comes into FreeSWITCH offering a=rtpmap:101 telephone-event/16000. FreeSWITCH responds with a=rtpmap:96 telephone-event/8000 but then is unable to change the payload type to 96. This invite is a bit out of the ordinary as it does not specify telephone-event/8000. INVITE SDP *m=audio 63244 RTP/AVP 9 0 101* *b=AS:80* *a=rtpmap:101 telephone-event/16000* *a=fmtp:101 0-15* *a=ptime:20* *a=rtcp-xr:voip-metrics* *a=sendrecv* //fs logs after INVITE *2017-06-29 19:28:47.475013 [DEBUG] switch_core_media.c:4192 Set telephone-event payload to 101@16000* *2017-06-29 19:28:47.475013 [DEBUG] switch_core_media.c:4596 sofia/external/1231231234(a)X.X.X.X Set 2833 dtmf send payload to 101 recv payload to 101* FreeSWITCH then replies with the following SDP. *m=audio 33734 RTP/AVP 0 96* *a=rtpmap:0 PCMU/8000* *a=rtpmap:96 telephone-event/8000* *a=fmtp:96 0-16* *a=silenceSupp:off - - - -* *a=ptime:20* When the digits come in to FS they are type 96, confirmed via tcpdump however FreeSWITCH doesn't recognize them and this entry shows up in the logs when digits are entered. *2017-06-29 19:29:02.975006 [WARNING] switch_core_media.c:2563 Could not change to payload type 96, ignoring...* *2017-06-29 19:29:02.975006 [DEBUG] switch_core_media.c:2540 alternate payload received (received 96, expecting 0)* Sofia.conf.xml and external.xml are vanilla. Have tested liberal-dtmf=true but no luck. Tested on version 1.6.10 and 1.6.18. Was hoping someone might have a setting to try or any suggestions? Thanks!

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may be fix DNS?
by Sergey Safarov 30 Jun '17

30 Jun '17

[safarov@safarov ~]$ ping files.freeswitch.org PING files.freeswitch.org(2607:f348:1021::7 (2607:f348:1021::7)) 56 data bytes >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=1 Destination unreachable: No route >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=2 Destination unreachable: No route

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best way to fork media streams for a bridged call?
by Dave Horton 29 Jun '17

29 Jun '17

If I have a bridged call and want to duplicate the two media streams and send each in a separate (one-way, sendonly) SIP dialog to a remote uri, what would be the best way to do that? It looks like it could possibly be achieved using the eavesdrop application (?), or is there an alternate preferred/tested method?

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FS service issues (systemd)
by Rick Jarvis 29 Jun '17

29 Jun '17

Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install]

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