July 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Mobile WebRTC session
by Luca Pradovera 06 Jul '17

06 Jul '17

Hello, I am trying to diagnose the reason why we only have one-way audio (browser to device, not the reverse) in a WebRTC call using Verto, which does not really matter in this particular case as signaling works great. This is where the issue starts appearing: https://gist.github.com/lpradovera/607de959022d06adf4ff6f8fcfc97510#file-gi… Based on the fact that payload 96 might be SRTP, I am postulating this is is a TLS problem and not a codec issue, but I am really grasping at straws. Does anyone have a pointer, please? Thanks! Luca

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mod_callcenter - Unloading queue with abandoned members
by Shaun Stokes 06 Jul '17

06 Jul '17

When using the command 'callcenter_config queue unload queuename@context' this works great except when there are abandoned members in the queue, in this situation we get the following error repeatedly: [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member The only solution that seems to work is re-add the queue and wait for the abandoned members to disappear as defined by 'discard-abandoned-after'. Is this normal behaviour in which case what's the command to clear abandoned members from the queue before it's unloaded, or is this a bug? Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes(a)itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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Any plan to use DTLSv1.2
by Varghese Paul 06 Jul '17

06 Jul '17

Hi all, Right now we are hard coded for using DTLSv1.0 which use TLS1.1. In the latest master I can see we are using DTLS1.0 and DTLS1.2 method if the SSL version is 1.1.0. Do we have any plan to add support for DTLSv1.2 ( TLS1.2) ?. Regards Varghese Paul

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from Asterisk sip.conf to FS
by Valter Nogueira 06 Jul '17

06 Jul '17

I have an app that runs in Asterisk and I am trying port it to FS. Below, I pasted a small sip.conf that sums up all my needs. 1. We register out in a VOIP_PROVIDER 2. An external asterisk register in our asterisk as CLIENT01 3. Calls from CLIENT01 route to VOIP_PROVIDER 4. Yealink's, with ACL restrictions, also uses VOIP_PROVIDER to external calls 5. VOIP PROVIDER and Yealinks use one NIC (eth0) 6. CLIENT01 uses another NIC (eth1) In Asterisk all configs are keep together in sip.conf. However, in FS it seems I should spread out things in different configs parts. So: Should Yealink's accounts go into /usr/local/freeswitch/conf/directory/default? Should I put VOIP_PROVIDER in sofia's external profile? Should I create an additional external profile to CLIENT01? Where should I define CLIENT01 account to allow it register in? In /usr/local/freeswitch/conf/directory/default? Thanks for any help. Valter === SIP.CONF === [general] disallow=all context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls alwaysauthreject=yes allowguest=no ;voip provider register => 185999:1010@xxx.xxx.xxx.xxx:5060 [VOIP_PROVIDER] type=peer context=FDTRONCO_94 host=xxx.xxx.xxx.xxx port=5060 username=185999 authuser=185999 authname=185999 secret=XXXXXXXXXXXXXX fromdomain=xxx.xxx.xxx.xxx fromuser=185999 insecure=port,invite canreinvite=no nat=no disallow=all allow=ulaw:30 dtmfmode=rfc2833 ignoreregexpire=yes language=pt_BR call-limit=9999 ;EXTERNAL ASTERISK ACCOUNT - CALLS GOES THRU VOIP_PROVIDER [CLIENT01] type=peer context=FDTRONCO_60 host=dynamic port=5060 username=CLIENT01 authuser=CLIENT01 authname=CLIENT01 secret=XXXXXXX insecure=port,invite canreinvite=no nat=yes disallow=all allow=g729 dtmfmode=rfc2833 ignoreregexpire=yes language=pt_BR call-limit=440 ;yealinks ip phones [1000] type=friend secret=XXXX host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.0.0/16 username=1000 context=FASTDIALER_RAMAIS callerid=testefone <1000> requirecalltoken=no nat=no canreinvite=no qualify=yes disallow=all allow=alaw call-limit=2 dtmfmode=RFC2833 language=pt_BR Callgroup=1 pickupgroup=1 [2012] type=friend secret=XXXX host=dynamic deny=0.0.0.0/0.0.0.0 permit=192.168.0.0/255.255.0.0 username=2012 context=FASTDIALER_RAMAIS callerid=Telefone sem fio <2012> requirecalltoken=no nat=no canreinvite=no qualify=no disallow=all allow=alaw call-limit=2 dtmfmode=RFC2833 language=pt_BR Callgroup=1 pickupgroup=1

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Verto calls
by Agustí Ubalde 06 Jul '17

06 Jul '17

Hi all, I am trying to call from Verto extension to another Verto extension. *Both are successfully registered* (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): *EXECUTE verto.rtc/1001 bridge()* Regards, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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Conditional call forward
by Ashwin Rath 06 Jul '17

06 Jul '17

Hi I have an extension which has call forward setup BUT i would like the call forward to work only when dialed from a certain number and not from another numbers. Can this be achieved ? -- Ashwin Kumar Rath

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Re: [Freeswitch-users] Dialplan: Outbound route: destination_number: regex question
by Nate 06 Jul '17

06 Jul '17

Hi Roy, many thanks for responding. I am simply trying to finishing configuring the server to make/receive calls from the PSTN. My reference to the regex is from trying to configure an outbound route. Can anyone offer any advice on a regex that will work for me to reach any of the following phone numbers: International: +64 22 333 4444 <+64%2022%20333%204444> non-local: 022 333 4444 local: 333 4444 >From the number assigned to my account? +64 9 xxx yyyy Many many thanks! Nate. On Sat, Jul 1, 2017 at 9:17 AM, <royj(a)yandex.ru> wrote: > Not quite clear. What are you trying to achieve? > > 30.06.2017, 11:11, "Nate" <nathan.port(a)gmail.com>: > > Good day/evening everyone, > > Apologies for not being able to figure this out on my own. I've been > searching and trying for several days to get inbound/outbound working but > have yet to see success. > > At this stage I need help determining a proper regex expression for > handling New Zealand phone numbers. > > For instance, there are three different ways of expressing numbers here in > NZ: > > International: +64 22 333 4444 <+64%2022%20333%204444> > non-local: 022 333 4444 > local: 333 4444 > > A couple questions related to the SIP proxy: > > If the SIP proxy is located in Hong Kong, but the phone number is a New > Zealand number, does the location of the proxy have any impact on the > number of characters in the string for inbound/outbound calls? > > Again, apologies for the rudimentary nature of these questions, but having > nearly exhausted all other options (docs, searches, IRC), I am now spending > a large amount of time guessing and trial and error without any progress. > > Many thanks for any feedback at all. > > Nate > , > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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VOIP-GSM Gateway configuration Error
by Deepika Yadav 06 Jul '17

06 Jul '17

Hi, I have been trying for a while the configuration of a VOIP-GSM gateway that I have bought but the registration fails every time. Error at Freeswitch console 2017-06-27 16:27:10.191913 [WARNING] sofia_reg.c:2906 Can't find user [*testadmin(a)192.168.2.71 <testadmin(a)192.168.2.71>*] from *192.168.2.69* You must define a domain called '192.168.2.71' in your directory and add a user with the id="*testadmin*" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2017-06-27 16:27:14.071916 [ERR] sofia_reg.c:2447 MySIP Failed Registration with status Request Timeout [408]. failure #1 2017-06-27 16:27:14.111913 [WARNING] sofia_reg.c:505 MySIP Failed Registration [408], setting retry to 30 seconds. Here *'192.168.2.71*' is the local IP of the Freeswitch server and *192.168.2.69* that of the gateway attached to the LAN port of the Freeswitch server. I have put SIP UserID in the gateway admin panel as "*testadmin*" alongwith its passoword. my external SIP profile is as follows: <include> <gateway name="MySIP"> <param name="username" value="testadmin"/> <param name="realm" value="192.168.2.69"/> <param name="from-user" value="testadmin"/> <param name="from-domain" value="192.168.2.69"/> <param name="password" value="****"/> <param name="extension" value="5888"/> <param name="proxy" value="192.168.2.69"/> <param name="expire-seconds" value="120"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> </gateway> I have tried number of permutations and combinations but the gateway is not getting registered. Regards Deepika https://www.iiitd.edu.in/~deepikay/

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Dialplan: destination_number: regex question
by Nate 06 Jul '17

06 Jul '17

Good day/evening everyone, Apologies for not being able to figure this out on my own. I've been searching and trying for several days to get inbound/outbound working but have yet to see success. At this stage I need help determining a proper regex expression for handling New Zealand phone numbers. For instance, there are three different ways of expressing numbers here in NZ: International: +64 22 333 4444 non-local: 022 333 4444 local: 333 4444 A couple questions related to the SIP proxy: If the SIP proxy is located in Hong Kong, but the phone number is a New Zealand number, does the location of the proxy have any impact on the number of characters in the string for inbound/outbound calls? Again, apologies for the rudimentary nature of these questions, but having nearly exhausted all other options (docs, searches, IRC), I am now spending a large amount of time guessing and trial and error without any progress. Many thanks for any feedback at all. Nate

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Simple script for ringtone
by Alex Lake 05 Jul '17

05 Jul '17

I want to set up a number that just rings and never answers. Unfortunately my freeswitch is very rusty! Can someone help me out with what must be a 2 or 3 line script? ;-) Alex

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