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Freeswitch+iptel
by Murugan Pandian 20 Jul '15

20 Jul '15

Hi, I need to use FS only media server and WS-SIP signalling server Media-server i disabled auth-call in internal.xml so i can make call from iptel account WS-SIP But when i try to use WS-SIP i can't able to register using iptel account getting 408 error(chrome browser) If i am doing wrong here pls help me

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Re: [Freeswitch-users] Is there a way to temporarily disable DTMF controls during an eavesdrop (or control an eavesdrop session with non-DTMF stimulus)?
by Steven Szeto 20 Jul '15

20 Jul '15

Michael Jerris asks: does it work if you do the set before the eavesdrop? That set is not running until after the eavesdrop ends. Answer: Yes, set:eavesdrop_enable_dtmf=false, does work if it is invoked before the eavesdrop. However there are two issues: * The "queue_dtmf" command is now ignored, which means the eavesdrop session starts as silent monitor instead of a whisper/coach session * The attempt to set eavesdrop_enable_dtmf=true does not seem to reopen the DTMF listening capabilities of the eavesdrop session. DTMF keystrokes 0,1,2,3 have no effect. The solution for me would be this: * Before the eavesdrop session is started, , set:eavesdrop_enable_dtmf=false * Introduce the ability to change the eavesdrop mode via the command line o E.g. eavesdrop_change_mode <uuid> [0,1,2,3] o Where [0,1,2,3] are the following eavesdrop modes respectively: silent monitor, coach the a-leg, coach the b-leg, barge-in conference From: Steven Szeto Sent: Friday, July 17, 2015 4:01 PM To: 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> Subject: Is there a way to temporarily disable DTMF controls during an eavesdrop (or control an eavesdrop session with non-DTMF stimulus)? I can successfully create a whisper/coach eavesdrop session with the following commands issued from fs_cli: originate sofia/internal/5401(a)10.47.41.109<mailto:sofia/internal/5401@10.47.41.109> &bridge(sofia/internal/5901(a)10.47.41.109<mailto:sofia/internal/5901@10.47.41.109>) whisper/coach: originate sofia/internal/5902(a)10.47.41.109<mailto:sofia/internal/5902@10.47.41.109> 'queue_dtmf:w2@500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline I would like to temporarily disable DTMF so that the supervisor can not change the eavesdrop mode via DTMF, but this does not work: originate sofia/internal/5902(a)10.47.41.109<mailto:sofia/internal/5902@10.47.41.109> 'queue_dtmf:w2@500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861, set:eavesdrop_enable_dtmf=false' inline Moreover, I have a GUI app that presents the supervisor with the ability to change the eavesdrop mode. When he presses the appropriate icon in the GUI, I would then like to reenable DTMF and change the eavesdrop mode on the session: Uuid_setvar a28739d0-00f0-4a59-8c82-7a5a74ab6861 eavesdrop_enable_dtmf true; Uuid_recv_dtmf a28739d0-00f0-4a59-8c82-7a5a74ab6861 w3@500 Is there another way to change the mode of an eavesdrop session? ________________________________ NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

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Serving Verto HTML from a different server
by Chad Phillips 20 Jul '15

20 Jul '15

I’ve gotten fairly comfortable with the setup offered in the Confluence wiki for the new Verto video features. I’m curious to know if I can expect any gotchas if I serve the Verto web stuff from a different server than the FreeSWITCH instance is running on. >From what I can tell, as long as I make sure to use the same server certs on both servers, it *should* work. Is there anything I’m not taking into consideration? Any better way to approach the problem besides certificate sharing across servers? Thanks, Chad

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I am using outbound to ivrd. My Call gets answered however why do I not get events with $conn->events("plain", "ALL");
by jaflong jaflong 20 Jul '15

20 Jul '15

<include> <extension name="demo"> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="ivr_path=/usr/local/freeswitch/scripts/ivr_test.php"/> <action application="socket" data="127.0.0.1:8004" async full/> </condition> </extension> </include> ivr_test.php -------------- #!/usr/bin/php <?php require_once('ESL.php'); $conn = new ESLconnection(0); $music = "/usr/local/freeswitch/sounds/demo.wav"; $conn->execute("answer"); $conn->execute("playback", $music); while ($conn->connected() == 1) { $conn->events("plain", "ALL"); $e = $conn->recvEvent(); if ($e) { $name = $e->getHeader('event-name'); //echo "Got event $name\n"; $out = fopen('php://stderr', 'w+'); fwrite($out, "Got event $name\n"); } else { //echo "Event empty\n"; $out = fopen('php://stderr', 'w+'); fwrite($out, "Event empty\n"); } } I am using outbound to ivrd. My Call gets answered however why do I not get events with $conn->events("plain", "ALL");

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shared line or other suggestion
by Jason Holden 20 Jul '15

20 Jul '15

I am currently trying to have the ability to place a call on hold and be able to pick the call up from a displayed list on a group of Polycom phones. I am currently using shared line but when the main phone places a call on hold the shared lines to not show the held call with the remote user CID but the primary phonesCID information. Does anyone have a recommendation that'll work better. I want to be able to see the remote CID of the caller who is on hold.

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Voice call failed from softphone (X-lite) to Hard phone ( Grand stream GXP280 )
by Shisheer Teli 20 Jul '15

20 Jul '15

Hi Team, I am using softphone: x lite and hard phone: Grand stream GXP280 with FreeSWITCH. Both phones registered with FreeSWITCH. When I called from hard phone to the softphone, I am able to receive a call. But if I called from softphone to hard phone it gives me an error (failed to established call) -- Regards, Shisheer T

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ESL PHP error
by Admin 19 Jul '15

19 Jul '15

Thank you for responding to my question. I am using the standard PHP not the embedded one. However, i edited the php.ini file and set "enable_dl" parameter to 'on' but still getting the same error. I am starting to think that this issue is related to PHP version 5.6 and may be a reswig is needed as it was mentioned in one the archived posts.

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ESL PHP error
by Admin 18 Jul '15

18 Jul '15

I am getting the following error when trying to test an inbound connection to FS using ESL with PHP. --------------------------------------------------------------- > php test.php PHP Warning: dl(): Dynamically loaded extensions aren't enabled in /usr/share/php/ESL.php on line 22 PHP Fatal error: Call to undefined function new_ESLconnection() in /usr/share/php/ESL.php on line 155 ---------------------------------------------------------------- The installation (make phpmod-install) went just fine with no errors. It copied ESL.so to '/usr/lib/php5/20131226' and created esl.ini under "/etc/php5/apache2/conf.d/esl.ini" I had ESL php working last year when I was testing a php application. But then I was running Debian 7.1 and PHP 5.5. Now i am using the following: FreeSWITCH Version 1.4.20+git~20150703T164215Z~b95362f965~64bit PHP 5.6.9-0+deb8u1 (cli) Debian 8.1 64-bit Any help is very much appreciated. Thanks.

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Is there a way to temporarily disable DTMF controls during an eavesdrop (or control an eavesdrop session with non-DTMF stimulus)?
by Steven Szeto 17 Jul '15

17 Jul '15

I can successfully create a whisper/coach eavesdrop session with the following commands issued from fs_cli: originate sofia/internal/5401(a)10.47.41.109 &bridge(sofia/internal/5901(a)10.47.41.109) whisper/coach: originate sofia/internal/5902(a)10.47.41.109 'queue_dtmf:w2@500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline I would like to temporarily disable DTMF so that the supervisor can not change the eavesdrop mode via DTMF, but this does not work: originate sofia/internal/5902(a)10.47.41.109 'queue_dtmf:w2@500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861, set:eavesdrop_enable_dtmf=false' inline Moreover, I have a GUI app that presents the supervisor with the ability to change the eavesdrop mode. When he presses the appropriate icon in the GUI, I would then like to reenable DTMF and change the eavesdrop mode on the session: Uuid_setvar a28739d0-00f0-4a59-8c82-7a5a74ab6861 eavesdrop_enable_dtmf true; Uuid_recv_dtmf a28739d0-00f0-4a59-8c82-7a5a74ab6861 w3@500 Is there another way to change the mode of an eavesdrop session? ________________________________ NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

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Re- Connecting into British Telecom (BTs) SIP sandbox test platform
by Andrew Keil 17 Jul '15

17 Jul '15

To FreeSWITCH users, Just a quick question that I hope someone can help me with. I am sure I can solve this myself when BT sort out their firewall, however while I wait 1-2 days I thought I would post this question here. Currently BT have provided me with two SIP IPs and MEDIA IPs for their Sandbox: BT SBC1 IP: Signalling xxx.xxx.xxx.26 (Media xxx.xxx.xxx.25) BT SBC2 IP: Signalling xxx.xxx.xxx.23 (Media xxx.xxx.xxx.22) Plus a SIP Trunk Number Range: 0551xxxx100 - 0551xxxx109 Model PSTN numbers: 0207xxxx713 and 0207xxxx714 First local CentOS 6.6 FreeSWITCH server (version 1.4.20) has two NICs (eth0: 192.168.2.10 {with public IP address mapped to this IP (xxx.xxx.xxx.67)} ; eth1: 192.168.2.11 (used internally only)} Second local CentOS 6.6 FreeSWITCH server (version 1.4.20) has two NICs (eth0: 192.168.2.12 {with public IP address mapped to this IP (xxx.xxx.xxx.68)} ; eth1: 192.168.2.13 (used internally only)} They have stated to me that they do not require and registration or authentication. Unfortunately when I looked at all the samples on the FreeSWITCH Confluence they all seem to mention registering and sip proxy etc.... Can someone provide a simple .xml file(s) example for the above (even if it is to only one SBC) to fast-track my setup. Any other hints or tips (or links to freeswitch.org Docs) I would appreciate. One more thing to note, I am connecting FreeSWITCH via a local Firewall to BT's SBC's listed above, there is no SBC on my side since currently it is not needed based on my client's current requirements. Hopefully this makes everything a little simpler, however I do understand that this configuration maybe less flexible (especially when commissioning more FreeSWITCH servers). Thanks in advance, Andrew Keil Visytel Pty Ltd PS. Since I am currently undertaking BT's testing on behalf of my client in the UK, I will provide back to the FreeSWITCH community the completed BT Interoperability SIP test cases with the associated FreeSWITCH dialplans used and any comments. Hopefully this will help any other company to pass BT's SIP testing requirements in a fast manner :)

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