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Connecting asterisk to freeswitch
by Ashwin Jain 10 Jul '13

10 Jul '13

Hi, I am trying to connect asterisk to freeswitch. For this, I have added a contact in sip_profiles/users.xml, and asterisk is connecting FS as a sip contact. When I am trying to dial to asterisk contact, asterisk is not getting any SIP packets. Following is the registration information of asterisk contact: Registrations: ================================================================================================= Call-ID: 4e46179c713c03f42d1a6dfe3b104777(a)127.0.0.1 User: 102(a)freeswitch.abcd.co Contact: "user" <sip:102@182.172.122.40> Agent: Asterisk PBX 1.6.2.12 Status: Registered(UDP)(unknown) EXP(2013-07-10 03:09:02) EXPSECS(128) Host: freeswitch IP: 182.172.122.40 Port: 14638 Auth-User: unknown Auth-Realm: freeswitch.abcd.co MWI-Account: 102(a)freeswitch.abcd.co Total items returned: 1 ===================================================================== -- Thanks and Regards, Ashwin Jain

3 2

Firefox WebRTC 'fingerprint' SDP attribute missing 200OK response
by Karl Mack 10 Jul '13

10 Jul '13

All: If possible I would like to inquire why my local build of 1.4.0 B3 does not return a 'fingerprint' SDP attribute in the 200 OK response for a WebRTC origination from Firefox v24.0a1. I completed a similar WebRTC call via Freeswitch website 'https://webrtc.freeswitch.org/sipml5/' and the 'fingerprint' SDP attribute exists. Am I missing an configuration setting? *-- Local Build : freeswitch-1.4.0.beta3 (6/28/2013) --* INVITE sip:9600@example.com SIP/2.0 Call-ID: 1373406010440 CSeq: 1 INVITE From: "Test System" <sip:5275@example.com>;tag=1373406014154 To: <sip:9600@example.com> Via: SIP/2.0/WS B9x0pJyOAiMG.invalid;branch=z9hG4bK1373405991583;rport Max-Forwards: 70 Content-Type: application/sdp User-Agent: TestWebPhone/1.00 Allow: INVITE,ACK,CANCEL,BYE Contact: <sip:5275@B9x0pJyOAiMG.invalid;transport=ws> Content-Length: 588 v=0 o=Mozilla-SIPUA-24.0a1 1327 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 a=ice-ufrag:0dbd1e66 a=ice-pwd:9cc76e3ba9334f2e8369fc5f730d4839 a=fingerprint:sha-256 BE:83:F6:B9:A0:B7:73:0A:26:A0:2A:BC:54:9D:5F:E9:1A:9D:A3:5A:60:66:A7:EF:05:42:31:42:79:B6:C9:BB m=audio 64358 RTP/SAVPF 109 0 8 101 c=IN IP4 192.168.129.83 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:0 1 UDP 2111832319 192.168.129.83 64358 typ host a=candidate:0 2 UDP 2111832318 192.168.129.83 62653 typ host SIP/2.0 200 OK Via: SIP/2.0/WS B9x0pJyOAiMG.invalid;branch=z9hG4bK1373405991583;rport=64794;received=192.168.129.83 From: "Vail Systems" <sip:5275@vailsys.com>;tag=1373406014154 To: <sip:9600@vailsys.com>;tag=QD139mFva9jNg Call-ID: 1373406010440 CSeq: 1 INVITE Contact: <sip:9600@74.63.156.145:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 762 Remote-Party-ID: "Outbound Call" <sip:9600@vailsys.com>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1373356842 1373356843 IN IP4 74.63.156.145 s=FreeSWITCH c=IN IP4 74.63.156.145 t=0 0 a=msid-semantic: WMS Xusp8lc7KzweB2mads8BzqXOOdfmpEjd m=audio 49172 RTP/SAVPF 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=ptime:20 a=rtcp:49173 IN IP4 74.63.156.145 a=ssrc:3722431166 cname:YBsXNuUnUhrmaDmd a=ssrc:3722431166 msid:Xusp8lc7KzweB2mads8BzqXOOdfmpEjd a0 a=ssrc:3722431166 mslabel:Xusp8lc7KzweB2mads8BzqXOOdfmpEjd a=ssrc:3722431166 label:Xusp8lc7KzweB2mads8BzqXOOdfmpEjda0 a=ice-ufrag:WkEmBs0fv5sLzBOn a=ice-pwd:37bGsVTc3ap3EB6q a=candidate:6683873826 1 udp 659136 74.63.156.145 49172 typ host generation 0 a=candidate:6683873826 2 udp 659134 74.63.156.145 49173 typ host generation 0 *---------------------------------------------------------* *-- O&A https://webrtc.freeswitch.org/sipml5/ -- *INVITE sip:9386@webrtc.freeswitch.org SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKcxkVDkhksN02zm8U31EiPCdwiTReMKQw;rport From: "Karl mack"<sip:1000@webrtc.freeswitch.org>;tag=RtwA31FZToUkxXyeZOLy To: <sip:9386@webrtc.freeswitch.org> Contact: "Karl mack"<sips:1000@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;+sip.ice;language="en,fr" Call-ID: eb87cf95-ea84-ae23-d7ee-309431fce563 CSeq: 21413 INVITE Content-Type: application/sdp Content-Length: 1604 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24 Organization: Doubango Telecom v=0 o=Mozilla-SIPUA-24.0a1 2839 1 IN IP4 0.0.0.0 s=Doubango Telecom - firefox t=0 0 a=ice-ufrag:6abd2c98 a=ice-pwd:ee047a97e6b3603180c693e7a065ae16 a=fingerprint:sha-256 B7:71:34:0E:4B:B1:B8:74:7E:02:DD:A0:3D:F4:19:98:DA:BD:B4:D3:85:3A:A6:3E:C2:BE:53:93:7A:17:E7:79 m=audio 51710 UDP/TLS/RTP/SAVPF 109 0 8 101 c=IN IP4 63.210.102.138 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:0 1 UDP 2111832319 192.168.129.83 63474 typ host a=candidate:1 1 UDP 1692467199 63.210.102.138 51710 typ srflx raddr 192.168.129.83 rport 63474 a=candidate:2 1 UDP 1692467199 63.210.102.138 52129 typ srflx raddr 192.168.129.83 rport 63474 a=candidate:0 2 UDP 2111832318 192.168.129.83 63475 typ host a=candidate:1 2 UDP 1692467198 63.210.102.138 50308 typ srflx raddr 192.168.129.83 rport 63475 a=candidate:2 2 UDP 1692467198 63.210.102.138 58034 typ srflx raddr 192.168.129.83 rport 63475 m=video 60914 UDP/TLS/RTP/SAVPF 120 c=IN IP4 63.210.102.138 a=rtpmap:120 VP8/90000 a=sendrecv a=candidate:0 1 UDP 2111832319 192.168.129.83 63476 typ host a=candidate:1 1 UDP 1692467199 63.210.102.138 60914 typ srflx raddr 192.168.129.83 rport 63476 a=candidate:2 1 UDP 1692467199 63.210.102.138 63252 typ srflx raddr 192.168.129.83 rport 63476 a=candidate:0 2 UDP 2111832318 192.168.129.83 63477 typ host a=candidate:1 2 UDP 1692467198 63.210.102.138 59956 typ srflx raddr 192.168.129.83 rport 63477 a=candidate:2 2 UDP 1692467198 63.210.102.138 55695 typ srflx raddr 192.168.129.83 rport 63477 SIP/2.0 200 OK Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=54562;received=63.210.102.138;branch=z9hG4bKcxkVDkhksN02zm8U31EiPCdwiTReMKQw From: "Karl"<sip:1000@webrtc.freeswitch.org>;tag=RtwA31FZToUkxXyeZOLy To: <sip:9386@webrtc.freeswitch.org>;tag=tFH27N15gcQyr Contact: <sip:9386@198.22.64.248:5060;transport=udp> Call-ID: eb87cf95-ea84-ae23-d7ee-309431fce563 CSeq: 21413 INVITE Content-Type: application/sdp Content-Length: 1525 User-Agent: FreeSWITCH-mod_sofia/1.5.3b+git~20130708T181922Z~35fca3d290 Accept: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE Supported: precondition,path,replaces Allow-Events: talk,hold,conference,presence,dialog,line-seize,call-info,sla,include-session-description,presence.winfo,message-summary,refer Content-Disposition: session Remote-Party-ID: screen=no;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1373382476 1373382477 IN IP4 198.22.64.248 s=FreeSWITCH c=IN IP4 198.22.64.248 t=0 0 a=msid-semantic: WMS F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O m=audio 24162 RTP/SAVPF 109 101 a=rtpmap:109 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=ptime:20 a=fingerprint:sha-256 06:6C:87:AA:34:0C:96:4E:37:60:F8:9A:42:C2:6B:D3:07:88:ED:59:EF:CA:B5:56:55:2B:3F:C6:3A:0D:06:F6 a=rtcp:24163 IN IP4 198.22.64.248 a=ssrc:975760054 cname:PSe18QBEBkcVJBzU a=ssrc:975760054 msid:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O a0 a=ssrc:975760054 mslabel:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O a=ssrc:975760054 label:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8Oa0 a=ice-ufrag:7MwKVhaawChGjfkY a=ice-pwd:XpQW5iqxssi1UHMR a=candidate:6695616536 1 udp 659136 198.22.64.248 24162 typ host generation 0 a=candidate:6695616536 2 udp 659134 198.22.64.248 24163 typ host generation 0 m=video 23786 RTP/SAVPF 120 a=rtpmap:120 VP8/90000 a=fingerprint:sha-256 06:6C:87:AA:34:0C:96:4E:37:60:F8:9A:42:C2:6B:D3:07:88:ED:59:EF:CA:B5:56:55:2B:3F:C6:3A:0D:06:F6 a=rtcp:23787 IN IP4 198.22.64.248 a=ssrc:289067359 cname:PSe18QBEBkcVJBzU a=ssrc:289067359 msid:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O v0 a=ssrc:289067359 mslabel:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O a=ssrc:289067359 label:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8Ov0 a=ice-ufrag:VkpGpYNFztvQB068 a=ice-pwd:mf7j9tFZLoAH62BS a=candidate:7819168960 1 udp 659136 198.22.64.248 23786 typ host generation 0 a=candidate:7819168960 2 udp 659134 198.22.64.248 23787 typ host generation 0 *---------------------------------------------------------* -- Thank You: Karl Mack Title: Software Engineer Address: Vail Systems 570 Lake Cook Road Deerfield, IL 60015 Telephone: 1-800-360-8245 Email: kmack(a)vailsys.com

2 4

FreeSWITCH Community Conference Call Today
by Michael Collins 10 Jul '13

10 Jul '13

Hello all, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_10 We are going to be discussing several topics: * ClueCon 2013 <http://www.cluecon.com> - register now and triple your chances to win * New FreeSWITCH wiki site under development * Community discussion of some mailing list questions, like the broker's call<http://lists.freeswitch.org/pipermail/freeswitch-users/2013-July/097491.html>feature Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

park and blf
by fs＠voice2net.ca 10 Jul '13

10 Jul '13

I have blf's working fine with extensions but I cannot get them to work with parking. Below is the setup I am using to park calls. Any ideas what I need to change to get this working?? Darcy Primrose <extension name="parking_slots"> <condition field="destination_number" expression="^(700\d{1})$" break="on-false"> <action inline="true" application="set" data="slot_count=${fifo(count $1@${domain})}"/> <action inline="true" application="set" data="slot_count=${slot_count:-9:2}"/> </condition> <condition field="${slot_count}" expression="^\:0$" break="always"> <action application="unset" data="fifo_chime_list"/> <action application="set" data="fifo_chime_freq=0"/> <action application="fifo" data="${destination_number}@${domain} in undef local_stream://moh"/> <anti-action application="fifo" data="${destination_number}@${domain} out nowait"/> </condition> </extension>

3 5

Problems with G722 sample rate
by Wesley Akio 10 Jul '13

10 Jul '13

Hi all, I'm having some trouble with G722 and AudioCodes HD310 in latest 1.2-stable... The phone sends the SDP: v=0 o=1000 946688278 946688278 IN IP4 192.168.0.108 s=- c=IN IP4 192.168.0.108 t=0 0 m=audio 4000 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 But Freeswitch is prepared to accept only G722@8000 as can be seen in: [DEBUG] sofia_glue.c:5181 Audio Codec Compare [G722:9:16000:20:64000]/[G722:9:8000:10:64000] For now I've changed the source to accept 16000 but according to Anthony, as per the RFC, 8000 is the correct value. I've seen some conversations between Nandy Dagondon ans Steven Ayre but I'm not sure if it is the same issue, nonetheless I could not find a solution. has anyone faced a similar problem and found a viable solution? Best, Wesley Akio TuntsCorp.com

4 4

Using say in play_and_get_digits
by Vishal Kakkar 10 Jul '13

10 Jul '13

Hi All, I have an IVR like below.. Problem is it repeat complete input recieved from user before asking for confirmation. i.e. user can not just press 1 to proceed while system is repeating recieved inputs. <action application="play_and_get_digits" data="20 21 50 15000 # $${IVR_PROMPTS}/AppVoiceFolder/EnterRechargeDetails.wav $${IVR_PROMPTS}/InvalidInput.wav my_digit ^((\d{10})\*(\d{6})\*(1|3|12))$"/> <action application="playback" data="$${IVR_PROMPTS}/AppVoiceFolder/RepeatEnteredDetails.wav"/> <action application="say" data="hi name_spelled iterated NEUTER ${my_digit}"/> <action application="play_and_get_digits" data="1 1 2 5000 # file_string://$${IVR_PROMPTS}/AppVoiceFolder/Press1ToConfirm.wav!file_string://$${IVR_PROMPTS}/AppVoiceFolder/Press2ToReEnter.wav $${IVR_PROMPTS}/InvalidInput.wav my_choice ^[1-2]"/> Something i tried to achieve this is- <action application="play_and_get_digits" data="20 21 50 15000 # $${IVR_PROMPTS}/AppVoiceFolder/EnterRechargeDetails.wav $${IVR_PROMPTS}/InvalidInput.wav my_digit ^((\d{10})\*(\d{6})\*(1|3|12))$"/> <action application="playback" data="$${IVR_PROMPTS}/AppVoiceFolder/RepeatEnteredDetails.wav"/> <action application="play_and_get_digits" data="1 1 2 5000 # say:'hi name_spelled iterated NEUTER ${my_digit}' file_string://$${IVR_PROMPTS}/AppVoiceFolder/Press1ToConfirm.wav!file_string://$${IVR_PROMPTS}/AppVoiceFolder/Press2ToReEnter.wav $${IVR_PROMPTS}/InvalidInput.wav my_choice ^[1-2]"/> But issue is that it reads "hi name_spelled iterated NEUTER 912345678*10001*12" as string. Please help how can i fix it. Please also note that i have local language sound files kept in "hi" folder so that digits pronounced by system use the same and this is working when i use first method. But when i include say in "play_and_get_digits" it picks up english again. Thanks, -Vishal

1 0

Cause: ALLOTTED_TIMEOUT
by Lloyd Aloysius 10 Jul '13

10 Jul '13

Hi All When originate fail . The cli shows mod_dptools.c:3123 Originate Failed. Cause: ALLOTTED_TIMEOUT Which channel variables gives this cause value. I try several no luck bridge_hangup_cause last_bridge_hangup_cause hangup_cause_q850 Thanks LLoyd * *

2 2

outgoing INVITE without SDP in proxy_media mode
by 최현국 10 Jul '13

10 Jul '13

Hello freeswitch-users, I've encountered the following problem. Outgoing INVITE message from FS has no SDP in proxy_media mode. (in default mode(hard proxy?), everything is working fine.) test enviorment is the following: SIP client (CSipSample) ---> FS ---> other SIP server I've configured the following configuration at FS for proxy_media mode. [sip_profile] <param name="inbound-late-negotiation" value="true"/> [dialplan] <extension name="public_extensions"> <condition field="destination_number" expression="(.+)$"> <action application="set" data="proxy_media=true"/> <action application="export" data="codec_string=${ep_codec_string}"/> <action application="bridge" data="sofia/trunk/$1(a)yyy.yyy.yyy.yyy"/> </condition> </extension> [freeswitch version] ./freeswitch -version FreeSWITCH version: 1.5.1b+git~20130624T105435Z~275f2cf092 this is INVITE message from SIP client. INVITE sip:2002@test.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.19:41089;rport;branch=z9hG4bKPjs.6qQMadZveqMBGhgPt4VOA20eJmYm4- Max-Forwards: 70 From: "2004" <sip:2004@test.com>;tag=TfVNZT3HS7yRsl8UhZfcv4zYjaHHgFfq To: <sip:2002@test.com> Contact: "2004" <sip:2004@192.168.2.19:41089> Call-ID: DJBhgeByGZJ7WJLzwLNtPliL1mc5vM8A CSeq: 32048 INVITE Route: <sip:xxx.xxx.xxx.xxx:5080;transport=udp;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_SC-02C-10/r2272 Content-Type: application/sdp Content-Length: 340 v=0 o=- 3582370009 3582370009 IN IP4 192.168.2.19 s=pjmedia c=IN IP4 192.168.2.19 t=0 0 m=audio 4000 RTP/AVP 99 0 8 101 c=IN IP4 192.168.2.19 a=rtcp:4001 IN IP4 192.168.2.19 a=sendrecv a=rtpmap:99 SILK/24000 a=fmtp:99 useinbandfec=0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 this is the outgoing INVITE message from FS. the message has no SDP. INVITE sip:2002@yyy.yyy.yyy.yyy SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5090;rport;branch=z9hG4bK1Djm23v8c17vF Route: <sip:cmf.nhn.jp> Max-Forwards: 69 From: "2004" <sip:2004@xxx.xxx.xxx.xxx>;tag=1K848K75U26Ne To: <sip:2002@yyy.yyy.yyy.yyy> Call-ID: 3b9cae19-6349-1231-80a1-3640b59986cf CSeq: 46354541 INVITE Contact: <sip:mod_sofia@xxx.xxx.xxx.xxx:5090> User-Agent: FreeSWITCH-mod_sofia/1.5.1b+git~20130624T105435Z~275f2cf092 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Length: 0 X-FS-Support: update_display,send_info P-Asserted-Identity: "2004" <sip:2004@xxx.xxx.xxx.xxx> Am is missing some configuration for proxy_media? Thank you in advance. Regards, hkchoi

1 0

Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.
by Ashley B 10 Jul '13

10 Jul '13

Gerald Were you able to resolve this issue? I am experiencing the exact same problem. Rgds Ashley. > > > ---------- Forwarded message ---------- > From: Gerald Weber <gerald.weber(a)besharp.at> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 27 Jun 2013 16:12:35 +0000 > Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 > BETA. > > I've played a little more and tried the demo at www.webrtc.org, which > works to chrome should be fine.**** > > ** ** > > I called another ext (snom) instead of funny prompts in the switch:**** > > Audio from the headset mic -> snom is ok but not the other way round. The > headset dont get any audio...**** > > ** ** > > Fsctl debug is here: **** > > http://pastebin.freeswitch.org/21120**** > > ** ** > > Pcap trace done on the fs server with pcapsipdump is here (hope this is > correct):**** > > > https://dl.dropboxusercontent.com/u/21597352/20130627-175955-2005-2002-7363… > **** > > ** ** > > ** ** > > *Von:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *Im Auftrag von *Anthony > Minessale > *Gesendet:* Donnerstag, 27. Juni 2013 17:27 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new > 1.4 BETA.**** > > ** ** > > try fsctl debug_level 10 and get a pcap for more info.**** > > ** ** > > It looks like before it was hairpinning then figuring out the local addr > and after you added localnet.auto it was right from the start, either way > it should have worked. **** > > ** ** > > On Thu, Jun 27, 2013 at 9:51 AM, Gerald Weber <gerald.weber(a)besharp.at> > wrote:**** > > Nope, windows firewall is disabled, also iptables on FreeSWITCH Testserver > is disabled. > Every other client is working (phonerlite, bria, xlite,etc) > > -----Ursprüngliche Nachricht----- > Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] Im Auftrag von Kristian > Kielhofner > Gesendet: Donnerstag, 27. Juni 2013 16:34**** > > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 > BETA. > > The traces you provided look fine... Do you have a firewall running on > your test PC, by chance? > > On Thu, Jun 27, 2013 at 10:07 AM, Gerald Weber <gerald.weber(a)besharp.at> > wrote: > > Ps: i also included the output from jssip in the pastebin link > > > > > > > > Von: freeswitch-users-bounces(a)lists.freeswitch.org > > [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von > > Anthony Minessale > > > > > > Gesendet: Donnerstag, 27. Juni 2013 15:16 > > An: FreeSWITCH Users Help > > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new > > 1.4 BETA. > > > > > > > > try adding the sofia profile param > > > > > > > > apply-candidate-acl=localnet.auto > > > > > > > > > > > > > > > > On Thu, Jun 27, 2013 at 4:45 AM, Gerald Weber > > <gerald.weber(a)besharp.at> > > wrote: > > > > I got rid of sip_secure_media and tried again, here is the trace: > > > > http://pastebin.freeswitch.org/21116 > > > > > > > > SIP User 2020 on Chrome calls Ext 20000 (the funny prompts thing) > > > > I'm no expert, but i guess there is something wrong with this**** > > > stun/run/ice thing.**** > > > > > > > > > Thanks, > > > > gw > > > > > > > > Von: freeswitch-users-bounces(a)lists.freeswitch.org > > [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von > > Anthony Minessale > > Gesendet: Mittwoch, 26. Juni 2013 17:24 > > > > > > An: FreeSWITCH Users Help > > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new > > 1.4 BETA. > > > > > > > > You don't have to set anything in the dialplan. > > > > Get a debug console trace and post it to pastebin. > > > > > > > > > > > > On Wed, Jun 26, 2013 at 9:41 AM, Gerald Weber > > <gerald.weber(a)besharp.at> > > wrote: > > > > Hi, > > > > > > > > how did you get audio working ? > > > > I've also tried jssip and latest chrome and sip works, but there is no > > audio. Codecs like opus, ilbc and isac are installed. > > > > In the dialplan i set sip_secure_media=true but the speaker stays silent. > > I've tried ws and wss bindings but no luck. > > > > > > > > Thanks for any suggestions, > > > > gw > > > > > > > > Von: freeswitch-users-bounces(a)lists.freeswitch.org > > [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von > > João Mesquita > > Gesendet: Mittwoch, 26. Juni 2013 15:28 > > An: FreeSWITCH Users Help > > Betreff: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new > > 1.4 BETA. > > > > > > > > Whooray!!!! I've installed the new beta yesterday and tried it out > > with > > Chrome+JsSIP and it works fantastic!!! > > > > > > > > One question tho, is it possible/desirable to have a sofia profile to > > work only with webrtc and not with udp or tcp bindings? > > > > > > > > I haven't tried and I guess I should've before asking dumb questions > but.... > > > > > > João Mesquita**** > > > FreeSWITCHT Solutions**** > > > > > > > > > On Tue, Jun 25, 2013 at 7:42 PM, Anthony Minessale > > <anthony.minessale(a)gmail.com> wrote: > > > > Now also working in latest FireFox 22 > > > > > > > > sipml5 works with both chrome and FF > > > > > > > > https://webrtc.freeswitch.org/sipml5/ > > > > > > > > > > > > On Tue, Jun 25, 2013 at 2:04 AM, Michael Jerris <mike(a)jerris.com> wrote: > > > > Media is not handled by websockets. Web-sockets are a new transport > > for sip signaling. If an endpoint registers from a websocket > > transport, when we send it a call, we automatically send it an sdp > > appropriate for webrtc. If you want to send a call with an sdp for > > webrtc that isn't for a registration over websockets, just add > > media_webrtc=true var to the origination vars. If we receive an sdp > from webrtc, we will automatically handle it. > > > > Mike > > > > > > On Jun 25, 2013, at 12:37 AM, Jeff Leung <jleung(a)v10networks.ca> wrote: > > > >> Media wise, how would that be handled with web sockets? > >> > >>> -----Original Message----- > >>> From: freeswitch-users-bounces(a)lists.freeswitch.org > >>> [mailto:freeswitch- users-bounces(a)lists.freeswitch.org] On Behalf Of > >>> Dan Barua > >>> Sent: Monday, June 24, 2013 3:10 PM > >>> To: freeswitch-users(a)lists.freeswitch.org > >>> Subject: Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to > >>> new > >>> 1.4 BETA. > >>> > >>> I asked the same question on Google+, Michael Jerris helpfully replied: > >>> > >>> you just need to setup FreeSWITCH as normal, and add 2 configuration > >>> params for the binding addresses for the websockets and secure > >>> websockets. the params are ws-binding and wss-binding format is > >>> ip:port or just :port if your using the same ip as sip-ip > >>> > >>> > >>> I haven't got round to trying it myself yet. > >>> > >>> -Dan > >>> > >>> > >>> > >>> > >>> -- > >>> View this message in context: http://freeswitch- > >>> users.2379917.n2.nabble.com/FreeSWITCH-adds-WebRTC-support-to-new- > >>> 1-4-BETA-tp7592138p7592141.html > >>> Sent from the freeswitch-users mailing list archive at Nabble.com. > >>> > >>> __________________________________________________________ > >>> _______________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting(a)freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >>> http://www.cudatel.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users(a)lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u > >>> sers > >>> http://www.freeswitch.org > >> > >> > >> _____________________________________________________________________** > ** > > >> ____ Professional FreeSWITCH Consulting Services:**** > > >> consulting(a)freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >> http://www.cudatel.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >> ers > >> http://www.freeswitch.org > > > > > > ______________________________________________________________________** > ** > > > ___ Professional FreeSWITCH Consulting Services:**** > > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs**** > > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > googletalk:conf+888@conference.freeswitch.org > > pstn:+19193869900 > > > >**** > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services:**** > > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > > > > > > ______________________________________________________________________** > ** > > > ___ Professional FreeSWITCH Consulting Services:**** > > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs**** > > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > googletalk:conf+888@conference.freeswitch.org > > pstn:+19193869900 > > > >**** > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services:**** > > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs**** > > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > googletalk:conf+888@conference.freeswitch.org > > pstn:+19193869900 > > > >**** > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services:**** > > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs**** > > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 **** > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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FS as SBC
by Nikolay Kondratyev 09 Jul '13

09 Jul '13

Hi All, I'm new to freeswitch and I'm investigating if i can use FS as SBC. I see that one can use FS as SBC for incoming and outgoing calls. But I was not able to find out if FS can be used for registration. I mean the subscriber from internet should send Register to FS, and FS should send Register to internal sip server. Is this possible? Another question is: what will happen with "contact parameters" in Invite and Register messages? Will FS pass unknown parameters through? I'm interested in particular in the following Contact field: Contact: <sip:+79980000004@10.169.10.64:5070;transport=udp>;+g.3gpp.icsi-ref="urn%3Au rn-7%3A3gpp-service.ims.icsi.mmtel,urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel .gsma.videocall";audio;video;q=0.8;+g.oma.sip-im;+g.oma.sip-im.large-message ;expires=3600 If FS will get Invite or Register from Internet with such Contact, is it possible to preserve parameters (namely +g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel,urn%3Aurn-7%3A3g pp-service.ims.icsi.mmtel.gsma.videocall";audio;video;q=0.8;+g.oma.sip-im;+g .oma.sip-im.large-message;expires=3600) on the internal call leg? Thanks in advance, Nikolay.

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