- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_verto: Invalid Method, Missing Method or Permission Denied / Error code": -32601
by Emrah 22 Jan '16

22 Jan '16

Hi all, Looked too easy to be true… I’m all set up with FS and the Verto demo. I use commercial certificates, the cA is already in my distrib, and everything seemed to work well. I can log in from the Verto page… Except when I attempt to make a call, it miserably fails with error code": -32601. I did my Google homework with no luck. Some stuff seem to point to the CA needing to be imported, but not sure where I could have gone wrong there. There is hardly any verbose… But a debug output is here: http://dmn.io/fs-verto-error.txt The error message is: Invalid Method, Missing Method or Permission Denied I’m not sure what to investigate here. Your suggestions are welcome! Running FreeSWITCH Version 1.6.5-11-d5520a6~64bit Thanks

4 16

Is there some setting that stops configuraton from being loaded from sofia.conf.xml?
by vfclists . 22 Jan '16

22 Jan '16

I am experiencing a strange issue on my system. It seems that configurations in sofia.conf.xml will not load correctly. It is a configuration that contains the entirety of internal.xml and external.xml in the sofia.conf.xml file, bypassing the files on the sip_profiles/external and for some reason it wasn't loading. However when I copy the profile sections for the internal and external profiles into the sip_profiles directory, it works normally. Is there some setting that can cause this? -- Frank Church ======================= http://devblog.brahmancreations.com

3 4

recv_refuse but normal_clearing / no failover?
by Avi Marcus 21 Jan '16

21 Jan '16

This leg was marked normal_clearing so it didn't try failing over. sip_hangup_disposition was recv_refuse but that didn't try the next carrier in my bridge string.. since it was cause 16/normal, I presume. Any suggestion on dealing with these kind of bugged calls? I've told the carrier... (running FreeSWITCH Version 1.6.5-11-d5520a6~64bit (-11-d5520a6 64bit)) *SIP/2.0 487 Request Terminated* Via: SIP/2.0/UDP 80.85.85.46;received=80.85.85.46;rport=5060;branch=z9hG4bK2jKN0erN6acjg To: <sip:001110144208202xxxx@sbc.voxbeam.com>;tag=3662377546-339101 From: "972535927644" <sip:97253592xxxx@80.85.85.46>;tag=6NBQDBN3Bg8HQ Call-ID: 47aa3d9c-3af3-1234-64a8-aec02ee4871b CSeq: 86358306 INVITE Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL Contact: <sip:callee@108.59.2.133;did=19b.0a235804> *Reason: Q.850;cause=16* P-Charging-Vector: icid-value=" 7fa379cc7fa379b-0004-00c3-0000-0000(a)10.11.100.185 ";icid-generated-at=10.11.102.24;term-ioi=italtel.com Content-Length: 0

1 0

WebRTC / Voice- and/or Video-Call to freeSwitch / which JavaScript-Library should I take
by Thorsten Göllner 21 Jan '16

21 Jan '16

Hi there, I am new to WebRTC. I just installed my first FreeSWITCH-Server (1.6.6) and SIP-Calls work fine. Now I want to implement a simple Button on our homepage to establish a voice and/or video-call with a fixed extension in our dialplan. I do *not* need a complete Sip-Client-Interface etc. I just want to implment a simple "Call-Button" :) Do you have any good suggestion, which JavaScript-Libarary I should use in that case? Thanks in advance Thorsten

2 1

[WARNING] switch_core_media.c:3261 RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....
by Karl Schmidt 21 Jan '16

21 Jan '16

OK this warning from when a polycom is going on hold. Is it the phone setup or freeswitch that is wrong? Any breadcrumbs? -- -------------------------------------------------------------------------------- Link to our website and get free US-48 shipping on your next order. Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB https://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-3089 Those that have knowledge don't predict. Those that predict, don't have knowledge. Lao tzu --------------------------------------------------------------------------------

2 1

Getting multicast paging to work
by Varsha Agarwal 20 Jan '16

20 Jan '16

Hi All, I am trying to get multicast paging to work with the default configuration in FreeSwitch. I am using 239.192.0.2 multicast IP Adress on port 8000. I have configured a phone with that multicast address and phone, however when I try to page, I just get dead air. I see some ZRTP negotiation failures in Freeswitch logs. Can someone tell me what could be wrong 479243ec-a237-481b-a94d-af6707cf2b07 2016-01-15 18:24:22.421859 [DEBUG] sofia.c:6614 Channel sofia/internal/1002(a)172.18.18.237 entering state [ready][200] 2016-01-15 18:24:22.481945 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21355 size=144. Stream 2:CLEAR:START 479243ec-a237-481b-a94d-af6707cf2b07 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:5856 Correct ip/port confirmed. 479243ec-a237-481b-a94d-af6707cf2b07 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:3548 Starting timer [soft] 160 bytes per 20ms 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: START SESSION INITIALIZATION. sID=3. 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: ZID=616331323132656430393363. 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: Loading User's profile: 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: allowclear: OFF 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: autosecure: ON 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: disclose_bit: OFF 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: signal. role: Unknown 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: TTL: 4294967295 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: SAS schemes: 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 B256 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 B32 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: Ciphers: 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 AES3 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 AES1 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: PK schemes: 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 EC25 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 DH3k 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 DH2k 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 Mult 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: ATL: 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 HS32 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: Hashes: 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 S256 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: Session initialization - DONE. sID=3. 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: ATTACH NEW STREAM to sID=3: 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp]: Stream ID=0 UNKNOWN switching <NONE> ---> <ACTIVE>. 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: Empty slot was found - initializing new stream with ID=3. 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: Preparing ZRTP Hello according to the Session profile. 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp main]: ATTACH NEW STREAM - DONE. 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp engine]: START STREAM ID=3 mode=CLEAR state=ACTIVE. 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp]: Stream ID=3 CLEAR switching <ACTIVE> ---> <START>. 2016-01-15 18:24:22.642176 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63956 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:22.702262 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63957 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:22.722291 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21356 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:22.802406 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63958 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:22.922579 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21357 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:23.002694 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63959 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:23.122867 [DEBUG] switch_rtp.c:1357 [ zrtp engine]: WARNING! HELLO have been resent 5 times without a response. Raising ZRTP_EVENT_NO_ZRTP_QUICK event. ID=2 2016-01-15 18:24:23.122867 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21358 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:23.202982 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63960 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:23.323155 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21359 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:23.403270 [DEBUG] switch_rtp.c:1357 [ zrtp engine]: WARNING! HELLO have been resent 5 times without a response. Raising ZRTP_EVENT_NO_ZRTP_QUICK event. ID=3 2016-01-15 18:24:23.403270 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63961 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:23.523443 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21360 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:23.603558 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63962 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:23.723731 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21361 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:23.803846 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63963 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:23.924019 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21362 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:24.004134 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63964 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:24.124307 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21363 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:24.204422 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63965 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:24.314580 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21364 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:24.394696 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63966 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:24.514868 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21365 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:24.594984 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63967 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:24.715156 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21366 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:24.795272 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63968 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:24.915444 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21367 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:24.995560 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63969 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:25.115732 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21368 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:25.195848 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63970 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:25.316020 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21369 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:25.396136 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63971 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:25.516308 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21370 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:25.596424 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63972 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:25.716596 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21371 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:25.796712 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63973 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:25.916884 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503275614 seq=21372 size=144. Stream 2:CLEAR:START 2016-01-15 18:24:25.997000 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63974 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:26.117172 [DEBUG] switch_rtp.c:1357 [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=2 2016-01-15 18:24:26.117172 [DEBUG] switch_rtp.c:1357 [ zrtp]: Stream ID=2 CLEAR switching <START> ---> <NOZRTP>. 2016-01-15 18:24:26.197288 [DEBUG] switch_rtp.c:1357 [ zrtp utils]: Send <HELLO> ssrc=1503386214 seq=63975 size=144. Stream 3:CLEAR:START 2016-01-15 18:24:26.397576 [DEBUG] switch_rtp.c:1357 [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=3 2016-01-15 18:24:26.397576 [DEBUG] switch_rtp.c:1357 [ zrtp]: Stream ID=3 CLEAR switching <START> ---> <NOZRTP>. Thanks, Varsha

4 16

How to customize voicemail greeting number ?
by Rutu Patel 20 Jan '16

20 Jan '16

Hi All, I am using voicemail module with odbc. I have a DID(123456789) in freeswitch and it's routed to local extension(1000). When the extension does not available it goes to voicemail and plays that number(1000) is not available.Is it possible to change the number in playback? I want to play DID(123456789) is not available instead of destination number(1000). Any suggestions will be appreciated. -- Thanks, Rutu Patel <http://www.inextrix.com>

5 11

The "awesome FreeSWITCH book"
by Steve Matzura 20 Jan '16

20 Jan '16

Before I lay my money down, does anyone know if this is useable with alternative format outputs such as braille displays and/or speech synthesis via the Kindle Reader on either the OSX or iOS platforms? Windows Kindle is a bit of a pile, sometimes the Mac and iOS versions work a lot better. There is also an older version of it on Bookshare (http://www.bookshare.org) and I'm wondering if just because it's three years old, is it out-of-date? If it is, what can I do to help move the process along for getting an updated edition into Bookshare?

3 7

Reloading XML's
by Steve Matzura 20 Jan '16

20 Jan '16

I'm just getting started with Freeswitch. I edited freeswitch.xml and added one command to set my external IP address as explained in the book. What I can't find is how to cause Freeswitch to rescan that file, or any XML file that gets changed. Do I have to restart the service to do that?

4 6

How to transfer call from one extension to another extension ?
by Ankit Doshi 20 Jan '16

20 Jan '16

Hi All, Anyone please help me out for transfer a call ? let me give you an example : i have 3 extension 1- caller(1000)2- receiver(1001)3- transfer(1002) I am dialing from 1000 and call is received by 1001 then push transfer button and press 1002 for transfer but its not going to transfer.I have set below parameters in sip profile but still its not going to work 1. manual-redirect=true,2. disable-transfer =false Can anyone please help me out for this ? Any suggestion will be appreciated . Best Regards, Ankit Doshi

1 0