- FreeSWITCH-users - FreeSWITCH Mailing Lists

Mod_Callcenter callers
by Richard Genthner 04 Oct '13

04 Oct '13

Is there a way in the dial plan for mod call center make it listen for a key press and break out and go to voicemail like if max-wait-time is hit ? -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com

3 3

Friday Free For All!
by Ken Rice 04 Oct '13

04 Oct '13

Hey Guys, Get in here! Friday Free For All is go!! SIP:888@conference.freeswitch.org or use the https:/webrtc.freeswitch.org webrtc Demo to call 888! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

1 0

Error Creating SIP UA for profile: X
by Henrik Aagaard Sørensen 04 Oct '13

04 Oct '13

I've had a server running only FreeSWITCH for about 6 months now. I have had it turned off for some while. Now, when trying to start FreeSWTICH, I get the errors: Error Creating SIP UA for profile: X where X differs between internal, external and others. When starting my CentOS 6 server I run: service iptables stop service ip6tables stop just to make sure that they aren't blocking anything. I have nothing else running on CentOS. I've tried to log everything from console, warning etc. but there's nothing indicating what the problem is. Can someone help me out?

7 12

freeswitch openfire
by Groupe SOGO 04 Oct '13

04 Oct '13

Dear Friends, I am working on freeswitch.Everything is going well. I want to integrate openfire 3.7.1 to freeswitch for video conference part. In my search, I find there is asterisk plugin for openfire. But I don't know if freeswitch plugin for openfire exists. Has anyone tried it? Where can I find this plugin? Thanks in advance for your help. Labolinux K. Amouzou

2 1

SRTP instructions on wiki wrong/outdated
by Luis Azedo 04 Oct '13

04 Oct '13

Hi, i have followed the instructions on wiki to setup srtp encryption and everything failed until i looked up in code and found that wiki refers to "sip_secure_media" but code uses "rtp_secure_media". cheers

3 2

get BYE event in javascripts
by Khue Nguyen Minh 04 Oct '13

04 Oct '13

Hi all, I have a simple javascripts with simple scenario. In this scenario, user can listen audio and send dtmf to play next or previous audio file. When user send dtmf to play other audio file, I will store play time of this audio to a text file (time.txt), but, if user hangup this call, the play time of last file cannot store in time.txt. How I can capture event user hangup to store play time into time.txt before javascripts session destroyed? Brs, Khue.

2 1

Problem with NAT configuration. Trying to understand autonat.
by Federico Castro 04 Oct '13

04 Oct '13

Hi everybody, please someone could help me to understand what "autonat:" makes when is used to modify ext-rtp-ip and ext-sip-ip in profiles. I faced the problem I'm going to explain and I solved it using "autonat" but I couldn't understand why it fixed my problem: *Problem:* Calls from SIP phones are dropped after 32 seconds *Scenario (not real IPs):* FS IP: 172.23.9.4 IP phone "A": 172.23.9.5 IP phone "B": 172.23.9.10 FS is behind NAT to call some VoIP providers. Public IP: 190.190.190.190 *What happens:* IP phone "A" calls IP phone "B", no NAT is necesary. All devices are in the same LAN, FS and both IP phones. I copy below profile configuration and 200 OK message that freeswitch sends to IP phone "A" when call is answered. Contact header is wrong, it is using Public IP instead of local IP. It produces that ACK from IP phone "A" never reaches FS and call is dropped by timeout. In profile configuration file: <param name="ext-rtp-ip" value="190.190.190.190"/> <param name="ext-sip-ip" value="190.190.190.190"/> Then, I changed to: <param name="ext-rtp-ip" value="autonat:190.190.190.190"/> <param name="ext-sip-ip" value="autonat:190.190.190.190"/> And it solved my problem. *My Questions:* Why when I dont use autonat "contact header" contains public IP if all devices are in the local network? And why autonat change this behaviour? Thanks for your help. Federico Castro *Profile configuration:* http://pastebin.com/X4DBg3yv *SIP 200 OK:* * * U 2013/10/03 16:22:35.536073 172.23.9.4:5060 -> 172.23.9.5:59453 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.23.9.4:59453 ;branch=z9hG4bK-d8754z-c07981061a655f7f-1---d8754z-;rport=59453. From: "somebody" <sip:2320@172.23.9.4:5060>;tag=87348974. To: <sip:2340@172.23.9.4:5060>;tag=1BQZX56QaNgZQ. Call-ID: M2RhY2EzNDA0YmQ5Y2E5YmJkYWIxYzI5ZTM0ZjkxZWE.. CSeq: 2 INVITE. Contact: <sip:2340@190.190.190.190:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.2.11+git~20130703T205557Z~60adf50f86. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 269. Remote-Party-ID: "2340" <sip:2340@172.23.9.4 >;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1380800075 1380800076 IN IP4 172.23.9.4. s=FreeSWITCH. c=IN IP4 172.23.9.4. t=0 0. m=audio 28080 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. m=video 0 RTP/AVP 19.

2 1

mod_voicemail and non_numeric id
by Lloyd Aloysius 03 Oct '13

03 Oct '13

Hi All I use non numeric id at user configuration(<user id="mike" >). All my dial plans in lua. *Problem :* when a user logged into his voicemail and try forward a message to another user's voicemail, mod_voicemail do not understand the extension number and always say invalid. Is there any way map extension number to user id , so that mod_voicemail can get the non_numeric id and forward the voicemail? Thanks Lloyd

2 3

Update SDP of Leg-B
by mehroz 03 Oct '13

03 Oct '13

Hi All, I am facing audio/video and video/audio switching issue under proxy media mode. It looks as of SDP does not update itself on the B-leg. OR if it does, it doesn't transmit video RTP after a switch from audio only. With the default media mode, this is achievable but , it b-Leg doesn't see any call update and hence pauses the video . 1) is there a way to explicitly change the SDP to be transmitted ? (proxy mode) 2) Any trigger or event at the B-leg to update the call , one A has done so (default mode) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Update-SDP-of-Leg-B-tp7595445… Sent from the freeswitch-users mailing list archive at Nabble.com.

3 4

sofia's SLA confusing caller_profiles
by Vik Killa 03 Oct '13

03 Oct '13

im may be wrong about this but it is confusing.... when an outbound call (B-leg) is "stolen" via SLA (sofia's intercept, meaning that A-leg pushes hold and a new leg, C-leg, is created when another phone hit's the line key to bridge to the B-leg) the B-leg's caller_profile contains an originator_caller_profile of the new C-leg's caller_profile and the C-leg's caller_profile contains an originatee_caller_profile of B-leg's caller_profile this seems counter-intuitive and backwards compared to how it functions in all other scenarios shouldn't the original B-leg have an originatee_caller_profile and the new C-leg have an originator_caller_profile? since the C-leg is the NEW leg, it should have an originator, no? and since the B-leg already exists, it should have an originatee? Thanks

1 0