- FreeSWITCH-users - FreeSWITCH Mailing Lists

api:execute vs session:execute
by John Nash 09 Feb '15

09 Feb '15

While reading documentation I saw that to call another lua script from inside a lua script we use .. api = freeswitch.API(); reply = api:executeString("luarun another.lua"); But we can also run another lua script using session:execute. I was wondering what is the difference, which one should be preferred?

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Re: [Freeswitch-users] Setting Headers in a LUA mod_curl call
by Roland Selmer 09 Feb '15

09 Feb '15

Is it possible to set arbitrary headers on a LUA mod_curl call like you would using the -H flag on a standard CURL? i.e. the equivalent of this curl -H “token: 1234" http://192.168.100.100:5000/v2.0/tokens for this kind of call session:execute("curl", "https://api.acme.com”) <——How does one specify a token header of 1234?

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Clear path in contact header when using TLS
by Emrah 09 Feb '15

09 Feb '15

Hi all, My experience with FS and TLS has been rather mixed so far. It's been a little inconsistent in keeping NAT sessions up and users discoverable. One thing I've noticed is that FS advertises the wrong information in option packets. The following is what I receive over my TLS session which is working on port 443. 1.2.3.4:443 -(SIP over TLS)-> 10.0.0.99:51132 OPTIONS sip:53178246@10.0.0.99:56494;transport=tls;received=5.6.7.8:51132 SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:443;branch=z9hG4bK6Kv171Q3U5rrD Route: <sip:53178246@5.6.7.8:51132>;transport=tls Max-Forwards: 70 From: <sip:mod_sofia@1.2.3.4:5060>;tag=Q6XDFHeUUrcHD To: <sip:user@domain.com> Call-ID: 0a052f23-34a8-4158-8c88-fd2a70ffb561_c2RhaSoOYBR6jfJe4ndLoTTKJMrO2gMv CSeq: 71498568 OPTIONS Contact: <sip:mod_sofia@1.2.3.4:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Do you see the From and Contact line? How can I rewrite this? Port 5060 has nothing to do there. Thanks and all the best, Emrah

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LIN16@16000h Freeswitch <-> Asterisk
by Antonio Eugenio Burriel 08 Feb '15

08 Feb '15

Hi, I've a FXO running Asterisk 1.8.32.1 registered to Freeswitch 1.4.15~1-1~wheezy+1. Majority of codecs (G722, ALAW, GSM, etc) work well, except for LIN16@16000h (SLIN16 in Asterisk). LIN16@8000h works great: v=0 o=root 739945760 739945761 IN IP4 192.168.1.2 s=Asterisk PBX 1.8.32.1 c=IN IP4 192.168.1.2 t=0 0 m=audio 18760 RTP/AVP 10 101 a=rtpmap:10 L16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2015-02-08 17:49:01.525330 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [L16:10:8000:20:0:1]/[L16:70:8000:20:128000:1] 2015-02-08 17:49:01.525330 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [L16:70:8000:20:128000:1] ++++ is saved as a match 2015-02-08 17:49:01.525330 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [L16:10:8000:20:0:1]/[G722:9:8000:20:64000:1] 2015-02-08 17:49:01.525330 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [L16:10:8000:20:0:1]/[PCMA:8:8000:20:64000:1] But with LIN16@16000h codec can't be matched, even being on Freeswitch allowed list!: v=0 o=root 202962363 202962364 IN IP4 192.168.1.2 s=Asterisk PBX 1.8.32.1 c=IN IP4 192.168.1.2 t=0 0 m=audio 13330 RTP/AVP 118 3 0 8 101 a=rtpmap:118 L16/16000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2015-02-08 15:53:13.645340 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [L16:118:16000:20:0:1]/[L16:70:16000:20:256000:1] 2015-02-08 15:53:13.645340 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [GSM:3:8000:20:13200:1]/[L16:70:16000:20:256000:1] 2015-02-08 15:53:13.645340 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[L16:70:16000:20:256000:1] 2015-02-08 15:53:13.645340 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[L16:70:16000:20:256000:1] With a ptime of 20ms, packet size is well below my MTU. Any hints?

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Can I use a domain name in an ACL list?
by Szeto, Steven 06 Feb '15

06 Feb '15

Suppose I have a switch with DNS hostname of "myswitch.company.com" and an IP address of 10.11.12.13. If I want to allow this switch's phones to call FreeSwitch, I have to update the acl.conf.xml file with an entry that looks like this: <list name="domains" default="deny"> <node type="allow" cidr="10.11.12.13/32" /> </list> What I would like to do is use the DNS hostname of the switch instead. So the above entry should look something like this: <list name="domains" default="deny"> <node type="allow" host="myswitch.company.com" /> </list> When I try the above, the calls are rejected by FreeSwitch. When creating its Access Control List, is there a way to tell FreeSwitch to do a hostname lookup when hosts are specified? Regards, Steve -- This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

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How to use FreeSWITCH system scheduler?
by Борисов, Дмитрий / Dmitriy Borisov 06 Feb '15

06 Feb '15

Hi! I want to schedule a FreeSWITCH event from lua script. How can I do this? sched_api is too hard way to do this, in my opinion. -- with best regards, Dmitriy Borisov

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[USER_NOT_REGISTERED]
by Mitchelle Johnson 06 Feb '15

06 Feb '15

Hi, I get this error when I am trying to call from 1016 to 1018 through FusionPBX. The FusionPBX/freeswitch server run on 192.168.0.103 and I am making a call from my machine at 192.168.0.106. The error logs can be found on http://pastebin.com/22MJ39HG Please help!!! Thanks, Mitchelle

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Call from "Client without TLS" to "Client with TLS"
by srdrztrk 06 Feb '15

06 Feb '15

Hi all, I googled so much, unfortunately found nearly nothing about it. I've got two different clients. One of them supports TLS and registers Freeswitch via TLS. The other one doesn't support TLS and registers Freeswitch via UDP. I can make calls from TLS to UDP direction. But making calls from UDP to TLS direction is a problem. What do i have to do for dialing from UDP to TLS direction? Is it possible? If it is, how can i edit the configuration? Any example will be appreciated. Thx.

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2708 anno create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
by Oguzhan Tuncer 06 Feb '15

06 Feb '15

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 06 Feb '15

06 Feb '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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