August 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Ubuntu and FreeSWITCH
by Saumar Hajjar 20 Aug '15

20 Aug '15

Hi, Those statements have been made in this list: "Whoever convinced you to use ubuntu must have been mad at you." "Ubuntu is really for desktops if you ask me. I would never put anything production on Ubuntu. Way to much bloat." "There are multiple known issues with Ubuntu 12.04. We reccomend Debian 8 Jessie." I have a system in production with a headless (of course) Ubuntu 14.04 with FS compiled from source (1.4.19). It seems to be running fine. Nothing fancy and it's not heavy loaded (yet). Is there any JIRA ticket that confirms that Ubuntu is to blame for some bad behavior? Are there any "known issues" with Ubuntu that I need to be aware of, documented elsewhere? If Ubuntu is not suited to run FS, I'll be glad to switch distros. But please, provide some evidence, otherwise some of you seem to be forcing consensus and trowing Ubuntu under the bus. Cheers, Saumar

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valet park
by Darcy Primrose 20 Aug '15

20 Aug '15

When we use valet park, and set announce_slot=true, it plays to the wrong party. The valet park works and you can retrieve it. If we park using the meta-ap, then it plays in the correct direction. Dial plan to park from a key <extension name="valet_park"> <condition field="destination_number" expression="^(park\+)?(\d{2})$" > <action application="bridge_export" data="hold_music=holdmusic/xxxx.xxxx.ca/moh.wav"/> <action application="set" data="valet_parking_timeout=300" /> <action application="set" data="valet_parking_orbit_exten=0" /> <action application="valet_park" data="park(a)xxxx.xxxx.ca $2" /> </condition> </extension> Using Meta AP <action application="bind_meta_app" data="0 ab s valet_park::park@voice2net.voice2net.ca auto in 61 70"/> Any guidance would be appreciated. Darcy Primrose Voice2Net Corp.

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Since we are talking about fax
by Blake Priddy 20 Aug '15

20 Aug '15

I have several fax extensions at the school district. How do you tell the extension to use t.38 The option is not available on the spa 2102. I appreciate any help I can get :) Thanks ya'll! -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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183 automatically marks origination as success?
by Avi Marcus 20 Aug '15

20 Aug '15

I have several carriers, and I use: <action application="set" data="continue_on_fail=true"/> ... so I was a little surprised to see that a call that ended with NORMAL_TEMPORARY_FAILURE didn't route advance. I dug into the PCAPs of this and comparable, and the only difference I noticed was that the ones that worked were invite -> 100 and then 503 NORMAL_TEMPORARY_FAILURE and the one that didn't work was invite -> 100, 183, and then 503. I have a lua script that told me: session:getVariable("originate_disposition") = SUCCESS session:getVariable("bridge_hangup_cause") = NORMAL_TEMPORARY_FAILURE so... I'm guessing that a 183 marks the originate as a success which is why continue_on_fail didn't work, since it wasn't a fail. Is this expected? Is the carrier "doing something wrong" here? They did helpfully send: "SIP/2.0 503 Service Unavailable - Route Advance" More importantly: how can I fix it? Thanks, -Avi Marcus BestFone

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Is there a way to acquire the eavesdrop mode of an active eavesdrop session?
by Steven Szeto 20 Aug '15

20 Aug '15

Suppose you created a silent monitor eavesdrop session by issuing the following commands via fs_cli: * originate sofia/internal/5401(a)10.47.41.109 &bridge(sofia/internal/5901(a)10.47.41.109<mailto:sofia/internal/5901@10.47.41.109>) * originate sofia/internal/3004(a)10.47.32.159 'eavesdrop: 0b2c9536-2f97-400a-b4da-dfb1d496855f ' inline The supervisor (3004 in the above example) then presses DTMF 3 to escalate the silent monitor eavesdrop session to a barge-in conference eavesdrop session. Is there a command in Freeswitch that would tell us that the supervisor is now in a barge-in conference? Note that the "show calls" command does not provide any indication of the current mode of the eavesdrop session 68d2beec-f853-4a67-99de-2e27e194adf8 outbound 2015-08-20 09:38:28 1440077908 sofia/internal/3004(a)10.47.32.159 CS_EXECUTE Steve Szeto 5380ip 3004 10.47.32.159 eavesdrop:0b2c9536-2f97-400a-b4da-dfb1d496855f ACTIVE 0000000000 SEND 68d2beec-f853-4a67-99de-2e27e194adf8 SSZETO-PC Outbound Call 5401 ________________________________ 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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ASTPP Version 2.1 is Released
by astpp 20 Aug '15

20 Aug '15

Hello Everyone, We are glad announcing release of version 2.1 of ASTPP with new features, improvements and bug fixes. Here is the complete list of enhancements and fixes: http://www.astpp.org/version2-1-enhancements *ASTPP is the most popular Open Source VoIP Billing Solution for Freeswitch.* It supports prepaid and postpaid billing along with call rating and credit control features. The list does not stop here, it also provides many other features such as calling cards, Least Cost Routing (LCR), DID management, Resellers management and many more. To view *full feature list*: http://www.astpp.org/astpp-features *Check out demo here:* http://www.astpp.org/online-demo *Give it a try:* /Quick Installation steps for v2.1:/ # wget --no-check-certificate https://goo.gl/ETmAJl -O install.sh # chmod +x install.sh # ./install.sh If you already have v2.0 installed, No worries!!! Upgrade is also available from v2.0 to v2.1. /Upgrade steps from v2.0 to v2.1:/ # wget --no-check-certificate https://goo.gl/RIuIej -O install.sh # chmod +x install.sh # ./install.sh In case if you find any issues, then please feel free to report in our bug tracker below: http://bugs.astpp.org/ -- Regards, ASTPP Team iNextrix Technologies Pvt. Ltd. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ASTPP-Version-2-1-is-Released… Sent from the freeswitch-users mailing list archive at Nabble.com.

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ASTPP Version 2.1 is Released
by ASTPP Opensource VOIP Billing 20 Aug '15

20 Aug '15

Hello Everyone, We are glad announcing release of version 2.1 of ASTPP with new features, improvements and bug fixes. Here is the complete list of enhancements and fixes: http://www.astpp.org/version2-1-enhancements *ASTPP is the most popular Open Source VoIP Billing Solution for Freeswitch.* It supports prepaid and postpaid billing along with call rating and credit control features. The list does not stop here, it also provides many other features such as calling cards, Least Cost Routing (LCR), DID management, Resellers management and many more. To view *full feature list*: http://www.astpp.org/astpp-features *Check out demo here:* http://www.astpp.org/online-demo *Give it a try:* *Quick Installation steps for v2.1:* # wget --no-check-certificate https://goo.gl/ETmAJl -O install.sh # chmod +x install.sh # ./install.sh If you already have v2.0 installed, No worries!!! Upgrade is also available from v2.0 to v2.1. *Upgrade steps from v2.0 to v2.1:* # wget --no-check-certificate https://goo.gl/RIuIej -O install.sh # chmod +x install.sh # ./install.sh In case if you find any issues, then please feel free to report in our bug tracker below: http://bugs.astpp.org/ -- Regards, ASTPP Team iNextrix Technologies Pvt. Ltd.

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vdeaf in mod_conference
by Richard Screene 20 Aug '15

20 Aug '15

Hello, In mod_conference is there any way for the API to suppress sending of the video to a member (essentially like a deaf command but for video)? I’ve looked through the documentation and I can’t see anything that exactly matches. Would it be possible to define an empty conference layout and assign it to the deaf’d user with vid-watching-canvas? Are there any other solutions? Regards, Richard

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no billsec variable after sending fax
by Dmitry Saratsky 20 Aug '15

20 Aug '15

Hi, It is no billsec variable session variable after txfax and disconnect. In 10% of faxes I do send – the var is present, in rest 90% cases the var is absent((( Usually I do send fax with following settings: return_ring_ready=false ignore_early_media=true sip_cid_type=rpid origination_privacy=hide_name origination_caller_id_number=17772818674 sip_contact_user=17772818674 origination_caller_id_name=_undef_ autohangup=false absolute_codec_string='PCMU,PCMA,G729' fax_use-ecm=false fax_enable_t38=true fax_enable_t38_request=true fax_disable_v17=true fax_ident='blah’ fax_header='blah 2’ Version 1.5.15b git 3d64402 2015-03-21 19:55:21Z 64bit Any ideas? Thanks, Dmitry.

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OT SIP Debugging ideas
by ik 20 Aug '15

20 Aug '15

Hello, I have a bit of off topic question. I have a PBX with soft-phones and everything seems to go well. The PBX have several gateways, and all seems to work. But when an extension is connected using a VPN (different IP range then the PBX), an incoming call is having one side RTP, but only for some gateways, that is, there are gateways that you are able to hear the call. When the VPN based extension dial to a gateway, you always have RTP. I used tcpdump, and was able to see when there is an RTP arriving back from the extension, and when there are none, but I could not figure out any reason for it. IT does not act as NAT issue, and there is no firewall. My question is, how to debug this mess, what should I be looking for, because I'm out of ideas. Thank you for any help Ido

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