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[WSS Unable to connect][No Debug logs]
by Bilal Abbasi 10 Feb '16

10 Feb '16

Hi all, I just want to enable the wss on freeswitch, so i have configuration in place, wss listening on port 7443, wss.pem is added in the certs directory, but i am unable to connect using https://tryit.jssip.net/,(i can get registered using wss://tryit.jssip.net:10443 but not through wss://MYSERVERIP:7443) So my question is how can i check the debug logs that whats actually happening on the switch, i can see the hits coming on server using tcpdump. Any help will be highly appreciated. Thanks Abbasi

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Outbound async ESL, linger, pickup
by royj＠yandex.ru 09 Feb '16

09 Feb '16

Hi, all Is anybody used pickup ' https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+pickup ' in outbound async ESL, linger? When trying pickup for example after this: sendmsg execute-app-name: bridge execute-app-arg: sofia/gateway/gateway_name/number,pickup/100 call-command: execute with: sendmsg execute-app-name: pickup execute-app-arg: 100 call-command: execute I got +OK and immediately CHANNEL_EXECUTE_COMPLETE, CHANNEL_HANGUP_COMPLETE for channel that calls pickup. It is not a script issues hangup command. But if that pickup to call from dialplan like <action application="pickup" data="100"/> and bridge from ESL then all works as expected. Can somebody point what might be wrong here? Or clarify the picture.

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FreeSWITCH abandons calls
by Elvis 09 Feb '16

09 Feb '16

I can see form my logs that calls are abandoned with reason [CS_NEW] [WRONG_CALL_STATE]. Please see logs below: [WARNING] switch_core_state_machine.c:572 2b48d9b0-ce3d-11e5-a1b4-8f0e26eefc3e sofia/internal/61415158474(a)212.61.145.185 Abandoned [NOTICE] switch_core_state_machine.c:575 Hangup sofia/internal/61415158474(a)212.61.145.185 [CS_NEW] [WRONG_CALL_STATE] [NOTICE] switch_core_session.c:1642 Session 8 (sofia/internal/61415158474(a)212.61.145.185) Ended [NOTICE] switch_core_session.c:1646 Close Channel sofia/internal/61415158474(a)212.61.145.185 [CS_DESTROY] Can someone please help out? Regards Elvis -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-abandons-calls-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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video conference not working
by Achintha 09 Feb '16

09 Feb '16

hi all, We are checking SIP video conferencing on FreeSwitch through mod_conference but the video is not getting enabled. Video works fine on a normal user to user call and codecs are matching. But for the conference room we create, the softphone displays an error message saying “No matching video codecs found”. We are using FreeSwitch version 1.6 kindly advice me to solve this problem Thanking you Achintha

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Capturing leg B hangup from bridge
by ik 09 Feb '16

09 Feb '16

Hello, I'm trying to use hangup api hook for leg b, while leg a is still connected and transferred to do other tasks. The hook only triggers when leg a disconnected, not when leg b. I found this question on Google, but no answer that works. So how can I execute API (curl in this case) when leg b is disconnected? Thank you, Ido

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Verto vs. SIP.js
by Thorsten Göllner 09 Feb '16

09 Feb '16

Hi, I want to setup a Click-2-Call-Button for our website. Is there any significant difference between Verto-Mod and libraries such as SIP.js? I do not really understand the advantage of Verto (if there is any). Thanks in advance, Thorsten

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Enterprise/sequential originate different stop codes per gateway
by Mateusz Bartczak 08 Feb '16

08 Feb '16

Hi all I'm looking for a proper way to achieve functionality of different stop codes per every called gateway Example call flow: call provider 1, stop on code 404 if provider 1 fails call provider 2, stop on code 404,486 if provider 2 fails call provider 3, stop on code 480 and so on I'm aware of variable continue_on_fail but I had no luck in specifying this per gateway, it seems to work only for whole originate string, not individual gateways Could you share some examples how to do this? Best Regards

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Class 5 and softphone app supporting ZRTP
by Maciej Bylica 08 Feb '16

08 Feb '16

Hi All, I am looking for a class 5 platform (basic VAS) and softphone (IOS, Android) both supporting ZRTP protocol to achieve the highest voice security. C.5 and UA should be delivered from the same supplier (like sipwise for instance) Could anybody recommend me any solution here? Thanks in advanced

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BLF Subscriptions sometimes don't send an initial Notify
by Stephen Thwaites 08 Feb '16

08 Feb '16

Hello, I have setup presence and in most cases it is working as expected. i.e. Subscription is sent to FS, FS returns Accepted then immediately FS sends the notify to the phone, thereafter all the Notify events for ringing, pickup and hangup. Great. However in some cases a subscribe to an extension does subscribe,FS sends the accepted response but a Notify is not sent out at that point. However if I call the extension the Notify works perfectly. Any ideas of what could cause the initial Notify not to be sent after the Acceptance 202? Any help would be appreciated. Regards, Steve. p.s I sent this to the list a few days ago but it didn't seem to come through on the list. Some detail info below: FS is configured as Multi-Tennant <param name="presence-proto-lookup" value="false"/> <param name="send-presence-on-register" value="true"/> <param name="manage-presence" value="true"/> <param name="presence-probe-on-register" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="dbname" value="freeswitch"/> <param name="presence-hosts" value="_DISABLE_"/> # Multi-Tennant <param name="presence-privacy" value="false"/> SIP Trace for Subscription FS Receives this from the phone: SUBSCRIBE sip:203 at xxx.mydomain.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:25060;branch=z9hG4bK991003231;rport From: <sip:201 at xxx.mydomain.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;tag=946578510 To: <sip:203 at xxx.mydomain.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;tag=nvqzSrr57RZg Call-ID: 968288340-25060-7 at BJC.BGI.B.CE <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> CSeq: 20050 SUBSCRIBE Contact: <sip:201 at 192.168.1.24 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:25060> X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2140 1.0.5.29 Expires: 480 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 FS Sends this back to the phone: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.1.24:25060 ;branch=z9hG4bK991003231;rport=59364;received=x.x.x.x From: <sip:201 at xxx.mydomain.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;tag=946578510 To: <sip:203 at xxx.mydomain.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;tag=nvqzSrr57RZg Call-ID: 968288340-25060-7 at BJC.BGI.B.CE <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> CSeq: 20050 SUBSCRIBE Contact: <sip:203 at y.y.y.y <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>:5060> Expires: 480 User-Agent: Callstera VOIP PBX v1.20 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, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=480 Content-Length: 0

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ASTPP Team launched Mobile SIP Dialer
by ASTPP Opensource VOIP Billing 08 Feb '16

08 Feb '16

Hi Everyone, ASTPP Team glad to announce first ever Mobile SIP Dialer which is integrated with ASTPP. *ASTPP Dialer* is a brand new Mobile SIP Dialer launched by ASTPP Team. Its a complete SIP Softphone which allows you to register your SIP account on any SIP server, specially ASTPP server. *Features:* - Easy to use & User friendly graphical interface - Call Transfer, Call Hold - Integration with Mobile Phonebook - Fast registration and call connectivity - Works with any SIP server - Displays real time Account balance (ASTPP only) - Displays Call history - Connect through 3G, 4G, GPRS and Wi-Fi - G729, GSM, iLBC, Speex, G711, G722, AMR Codec Support *Download:* ASTPP Dialer is available freely in *Google play store*: https://play.google.com/store/apps/details?id=com.inextrix.astppdialer Please do not forget to share your review in play store. *Want your own branded ASTPP Dialer?* We do also offer complete integration with your ASTPP server, and *rebrand / customize it with your company name/logo* in both *Android and iOS*. To get more details on it please *Click Here <http://www.astppbilling.org/astpp-mobile-dialer-transformation> *and send your requirements on sales(a)inextrix.com -- Regards, ASTPP Team iNextrix Technologies Pvt. Ltd.

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