February 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep!
by Aqs Younas 19 Feb '16

19 Feb '16

Hi. I am able to compile mod_avmd without any error. But when i try to load it i see this error in console. freeswitch@internal> load mod_avmd +OK Reloading XML -ERR [module load file routine returned an error] 2016-02-19 05:27:31.644461 [INFO] mod_enum.c:880 ENUM Reloaded 2016-02-19 05:27:31.644461 [ERR] mod_avmd.c:226 Couldn't register subclass avmd::beep! 2016-02-19 05:27:31.644461 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_avmd.so **Module load routine returned an error** These are the lines I see in mod_avmd.c but unable to understand. if (switch_event_reserve_subclass(AVMD_EVENT_BEEP) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't register subclass %s!\n", AVMD_EVENT_BEEP); return SWITCH_STATUS_TERM; } Any pointer is much appreciated. Best Regards. This email has been sent from a virus-free computer protected by Avast. www.avast.com <https://www.avast.com/sig-email> <#DDB4FAA8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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Firefox vs Chrome in Verto Communicator (self-signed certificates)
by Gonzalo Gasca Meza 19 Feb '16

19 Feb '16

I compiled FS 1.6 from source and using Verto Communicator with *Firefox* shows: "Waiting for server reconnection." Firefox 44.0.2 - MacOS system 10.10.4. - Windows 7 -*Works fine with Chrome* (I tested Chrome successfully to confirm there is no network issues). *-Same error in 2 different networks.* It works only after I enter in Browser manually: https://<IP>:8082 and accept the certificate I'm using self-signed certificates. Is this Browser related or Application related? Looks like is Browser related, need confirmation. *Console logs:* http://pastebin.com/XSJA9dZf *References:* https://bugzilla.mozilla.org/show_bug.cgi?id=594502 https://freeswitch.org/jira/browse/FS-8805 [image: Inline image 1]

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Persistent caller id on the download links ...
by Fabio Mafra 18 Feb '16

18 Feb '16

I am a new user FreeSWITCH. I found a flaw in the system right after installation to make a call handover between 3 SIP extensions. Example: Extension 1 Extension 2 calls. 2 extension meets and press the "Hold" button. Extension 2 press the "transfer" button, and dials the number of the extension 3. Extension 2 and Extension 3 communicate. Extension 2 press the "transfer" button again to complete the transfer of extension 1 to extension 3. Pressing the "transfer" the extension 2 comes out link. Now Extension 1 and Extension 3 are communicating connection with the extension 2. But in Extension 1 and Extension Display 3 appear as if they were communicating with Extension 2 (no longer part of the link). After transfer to the Station 1 does not have any indication be communicating with the Station 3? After the transfer of the branch line 3 does not have any indication be communicating with Extension 1? How can I fix this problem? My FreeSWITCH is installed on a Debian machine 8-Jessie at: / usr / local / freeswitch. I use the internal settings of extensions 1000.xml to 1019.xml. Thank you for your attention right now!

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uuid_simplify or uuid_deflect in mod_callCenter
by Varsha Agarwal 18 Feb '16

18 Feb '16

Hi, Can I use uuid_simplify or uuid_deflect in CallCenter config to route agent calls? I want to send REFER with Replace for Agent transfers as our agents are registered to a different PBX and FreeSwitch is just a gateway. REFER will remove FreeSwitch from the call path. If yes, can I get an example, I tried using it in CallCenter config but I get error cannot locate uuid_simplify or cannot locate channel uuid_deflect. Thanks, Varsha

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Oneway audio issues in freeswitch
by mohammed shafeeque 17 Feb '16

17 Feb '16

Hello All We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server. Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine) We have tried multiple softphones and the result is same. Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch. Any help or hint will be much appreciated. Thank you,

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Can FS offer both internal & external IP address?
by Ahmed Habiba 17 Feb '16

17 Feb '16

It is all about how you want things to run, FS allow to have multiple profile each of them can work on different ip considering that all users can call each other even they are registered on different IPs here is below and example: profile1: Name—> Internal IP —> 192.168.1.1 SIP Port —> 5060 Sip_profile.xml: <param name="ext-rtp-ip" value=“192.168.1.1"/> <param name="ext-sip-ip" value="192.168.1.1”/> Profile2: Name —> internal_nat IP —> 192.168.1.1 SIP Port —> 5065 —>(Port Forwarded on Router to —> RealIP:5060) Sip_profile.xml <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> However of the above configuration still both internal and external will be able to call each other based on the below dial string , the “*/“ assure that users from different profile can call each other conf/directory/default.xml : <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/> > On Feb 18, 2016, at 12:17 AM, freeswitch-users-request(a)lists.freeswitch.org wrote: > > From: Dan Edwards <DEdwards(a)vertical.com <mailto:DEdwards@vertical.com>> > Subject: [Freeswitch-users] Can FS offer both internal & external IP address? > Date: February 18, 2016 at 12:16:47 AM GMT+3 > To: "freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>" <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > > > I'm running FS behind Nginx, so I explicitly removed the local IP address from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside users work fine in that scenario. > > Internal users also work, as long as we turn on NAT reflection in our router. I'm looking for a way to not require that router change. > > If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as candidates in the initial INVITE, the browser could then use ICE to determine which IP address was correct. > > Is there a way to offer both and allow ICE on the client side to figure out which one to use? > > Thanks, > Dan >

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Can FS offer both internal & external IP address?
by Dan Edwards 17 Feb '16

17 Feb '16

I'm running FS behind Nginx, so I explicitly removed the local IP address from localnet.auto. This forces FS to always offer 'ext-rtp-ip'. Outside users work fine in that scenario. Internal users also work, as long as we turn on NAT reflection in our router. I'm looking for a way to not require that router change. If I can get FS to offer up both it's internal IP and 'ext-rtp-ip' as candidates in the initial INVITE, the browser could then use ICE to determine which IP address was correct. Is there a way to offer both and allow ICE on the client side to figure out which one to use? Thanks, Dan

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early media after bridge
by royj＠yandex.ru 17 Feb '16

17 Feb '16

Hi, all Is there any chance to use the early media from the b-leg as ringback to the a-leg when a-leg is answered for play_and_get_digits for example?

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Disable PRACK messages
by Antonio Silva 17 Feb '16

17 Feb '16

Hi, Is it possible to disable or not to send prack messages? I've set in the sip profile the option enable-100rel=false, but FS still sends the prack message... My problem is that after receive the 180 from the remote side, FS sends the prack message and the remote side reply with SIP/2.0 481 Call/Transaction Does Not Exist, This make FS to cancel the call. I know that this is an issue on the remote side... but i want to avoid this message if possible. Full sip trace in https://pastebin.freeswitch.org/24562 Thanks, António

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hiredis: realm must be defined
by royj＠yandex.ru 17 Feb '16

17 Feb '16

Hi, all Can anybody clarify what mean 'hiredis: realm must be defined'? Where is the misconfiguration? I am trying to use hiredis backend for limits and call api like 'api limit_usage hiredis myrealm id', got api/response '-1' and FreeSWITCH says in log 'hiredis: realm must be defined'. Module mod_hiredis loaded and exists. For hash backend works fine.

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