July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

1 0

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.

1 0

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.

2 1

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.

2 1

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 :)

3 2

Execute dialplan after valet_park
by SamyGo 17 Jul '15

17 Jul '15

Hi All, I need some help in finding a way to execute some dialplan code after the parked user Hangups from the parking lot. I've a scenario where Parked party decides to hangup while listening to MOH while parked. The Dialplan just Hangsup right away and I have no control. I've tried using: *session_in_hangup_hook*, and *api_hangup_hook=myscript.lua *but they get executed for the parker just after they hear the Parked slot number. Then I've tried *nolocal:export* these two variables for the other leg but nothing happens. Kindly suggest what other option are there. Best Regards. Sammy

2 5

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 17 Jul '15

17 Jul '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0

verto.conf.xml parameter for public access
by Nandy Dagondon 17 Jul '15

17 Jul '15

I succesfully installed mod_verto in my intranet. FS is behind NAT. Next step, I'm setting up where remote web clients can login. I setup my router DMZ to FS. This parameter is commented out "ext-rtp-ip" in the Wiki. Is this parameter necessary for accepting remote clients?

3 8

how to record SIP+RTP sessions
by Maciej Bylica 17 Jul '15

17 Jul '15

Hi, I need some help in finding tool to sniff SIP+RTP packets and separate each session. Following the wiki page https://wiki.freeswitch.org/wiki/Packet_Capture i've installed pcapsipdump 0.2 rel. Unfortunately it utilizes just one core to maximum and all .pcap files are quality affected. As i assume this is the limitation of this app and there is nothing i can do to overcome this. If possitive, could you please point me to some similiar free app. Thanks Maciej.

6 7

Issues with simultaneous call hangup
by Niraj Roy 17 Jul '15

17 Jul '15

Hello Gurus, We are using FreeSWITCH Version 1.4.18~64bit. We observed when both caller and caller hangs up the call simultaneously exec_after_bridge_app function does not execute for one of the legs. Here is the segment of executing the xml file. <action application="set" data="exec_after_bridge_app=transfer"/> <action application="set" data="exec_after_bridge_arg=endcall-handler XML CALLENDING"/> <action application="set_zombie_exec"/> <action application="set" data="continue_on_fail=true"/> <action inline="true" application="set" data="hangup_after_bridge=false"/> For normal or sequential call clearing everything works fine, but somehow this creates a race condition and state mismatch in the freeswitch which I don't know. Can anybody shed some light on this? Thanks,

3 4