October 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Opensips integration questions
by Schneur Rosenberg 12 Oct '16

12 Oct '16

Hi, I'm pretty new to Freeswitch, I built a complete solution using OpenSIPS/Asterisk and I'm in the process of replacing Asterisk with Freeswitch. My current system is built that OpenSIPS act as a registrar and as a load balancer, the rest is done on a cluster of Asterisk machines, which as I said before is being rewritten to Freeswitch, in Freeswitch I'm using xml_curl for my dialplan and call routing, I'm using Intralanman project to handle the sip users. With Asterisk I had OpensSIPS as a trunk and I know that everything coming from that trunk, is either a DID or a authenticated outgoing call, Asterisk and OpenSIPS shared the user table, Asterisk recognized the sip user name from the FROM field and used all the variables etc from its own sip user table as if the user has registered directly to itself, I had some issues with blind transfers but I was able to do some manipulations and DB queries to handle it fine. I tried doing the same with Freeswitch, I let the ACL allow the ip from Freeswitch, but Freeswitch would not use the variables from the user like Asterisk did, for example accountcode, user_context etc, I was able to partially fix that by having my web server do a db query and then using the "set" application set all the variables, but I'm not sure if its the proper solution, does anyone have a better solution? will setting my OpenSIPS as a gateway instead of merely allowing it in with ACL work better? right now I'm using the application "execute_extension" to send it to the proper context that I've pulled from the DB, I dont like the hack and if there is a better solution I would like to know. One more major problem that I have is NAT related, when I place a outgoing call for example from a phone which is behind a NAT (all of our customers phones are behind NAT) the call connects and audio works 2 ways, but when the called party hangs up, the carrier sends the BYE to Freeswitch, freeswitch then sends it to my OpenSIPS which is supposed to send it to my clients phone, but Freeswich writes the RURI with the natted address, the RURI is BYE sip:101@192.168.0.11:5062 SIP/2.0. and my OpenSIPS does not know what to do with it and ignores it, my Asterisk replaced the RURI with the received address in the VIA, so same RURI would be BYE sip:101@64.69.47.175:5062 SIP/2.0. which worked fine, same problem happens with all subsequent requests if user does not hang up, I need a solution for this major issue. thank you S. Rosenberg

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Freeswitch is not receiving audio unless and until send RTP to other end
by Naveen Tamanam 12 Oct '16

12 Oct '16

Hi, Recently I have found the following scenario, Freeswitch is not receiving audio unless and until we send RTP(audio) to other end. I have taken both recording and pcap for the call which is not having outgoing audio. Surprisingly, there is a incoming audio in pcap but not in recording. There is no NAT on machine. Is this any kind of known bug ? -- Thanks & Regards, Naveen Tamanam

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DTMF mode
by Yu Boot 12 Oct '16

12 Oct '16

FS receives DTMF tones from A-leg in RFC2833 mode, but generates SIP INFO to B-leg instead of RFC2833. How to fix this?

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mod_lcr with mssql backend
by Guillermo Ruiz Camauer 11 Oct '16

11 Oct '16

Does anyone have mod_lcr running against a MSSQL database through ODBC? What modifications did you have to make? -- Guillermo Ruiz Camauer

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Using SIPS and not SIP TLS
by Emrah 11 Oct '16

11 Oct '16

Hi there, I am in the middle of an argument with a softphone vendor over how to handle SIP and TLS. They support TLS with SIP and TLS with SIPS. If I pick TLS with SIPS, the softphone can lookup my SRV records and register on my FS box. However, it comes through with a transport=tcp as a contact parameter. If I use TLS with SIP, the softphone will ignore my _SIPS._TCP records but come through with a transport=tls. TLS with SIP and TLS with SIPS are almost identical with SIPS ensuring, in theory, that all signaling hops will use TLS. This, however, requires that FreeSWITCH uses a SIPS uri scheme to contact my endpoint and encapsulate in TLS regardless of the transport parameter. What do you suggest? Here is my ticket with them: https://www.acrobits.net/hesk/ticket.php?track=YQL-TRS-J3M4 Best, Emrah

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Re: [Freeswitch-users] Playback on Congestion
by Stanislav Sinyagin 11 Oct '16

11 Oct '16

Then you need some scripting, like Lua or an external ESL agent. On 10 Oct 2016 18:49, "Richard Gleason" <rgleason(a)voipnetwork.net> wrote: But it does not allow to specify rejection codes, for example I want to have two different recordings for SIP 404 and SIP 480. On Mon, Oct 10, 2016 at 3:12 AM, Stanislav Sinyagin <ssinyagin(a)gmail.com> wrote: > You can see an example of using continue_on_fail here: > http://wiki.sangoma.com/NSC-SIP-Trunking-Load-Balancing-And-Failover > > In your case, you play back a message instead of the second bridge. > > > > On Mon, Oct 10, 2016 at 5:15 AM, Richard Gleason > <rgleason(a)voipnetwork.net> wrote: > > Hello Guys, > > > > Need some help with a dial plan that I'm working on. > > > > The goal is to play a pre-recorded message to the originator on > congestion. > > > > For example, if GW replies back to FreeSWITCH with code SIP 404, I want > to > > trigger a pre-recorded message and play it back to the originator. > > > > Dial plan uses option bridge. > > > > If it is not difuclt can some one provide an example of dial plan for > this > > action. > > > > Thanks. > > > > -- > > Best Regards, > > Richard Gleason > > > > ____________________________________________________________ > _____________ > > 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 > -- Best Regards, Richard Gleason _________________________________________________________________________ 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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Playback on Congestion
by Richard Gleason 10 Oct '16

10 Oct '16

Hello Guys, Need some help with a dial plan that I'm working on. The goal is to play a pre-recorded message to the originator on congestion. For example, if GW replies back to FreeSWITCH with code SIP 404, I want to trigger a pre-recorded message and play it back to the originator. Dial plan uses option bridge. If it is not difuclt can some one provide an example of dial plan for this action. Thanks. -- Best Regards, Richard Gleason

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Meaning of flags and caps in XML CDRs
by Ciprian Dosoftei 10 Oct '16

10 Oct '16

Hello! I was wondering what is meaning (how to interpret) of the following CDR XML nodes: cdr > channel_data > flags cdr > channel_data > caps https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_cdr Thank you -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender.

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How to detect if the call is being transferred?
by Anis Bedhiafi 10 Oct '16

10 Oct '16

Hello I am working on application using Freeswitch and SIP.js in which I need to detect if the call is being transferred. The Sip.js application is actually an outbound dialer in which I load a list of phone numbers to call. The application starts by calling people. It is interesting to detect voicemail boxes. For that I am using mod_avmd. My problem is when the other end transfers the call, the sip.js client should know that the current call is being transferred. Is there a way that the original caller (sip.js client) knows that the call is being transferred? I was looking at all the channel variables and I found that the RDNIS variable and the transfer_time variable are filled up when there is a transfer. Is there a way to listen to the channel variables during the call and detect any change? Or is there a better way I can do it? Thank you for any suggestion. Anis Bedhiafi

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mod_python and new Session from freeswitch.Session
by Андрей Асташов 10 Oct '16

10 Oct '16

Hi Guys. >From Dialplan run scripts in Python, from which I perform freeswitch.session("......"). That is, create a new session. And if I do not get an answer for a long time from the gateway, the execution of other scripts in Python is delayed until such time as the session will not be installed. Please tell me how you can create a new session, and do not block completely mod_python. I understand that once had to be done through ESL, but now everything has already been written by the script from the dialplan. And it works, if it does not create a new session. freeswitch@Test-Environment> # # Execute python script from fs_cli(in dialer mode): # fs_cli --retry --batchmode --interrupt --execute="python python_test_script" # 2016-10-07 17:55:29.970991 [NOTICE] mod_python.c:212 Invoking py module: python_test_script 2016-10-07 17:55:30.119989 [NOTICE] switch_cpp.cpp:1360 -->(489d5441-701d-4a98-80ca-3389b00fa98a) Create a new session through freeswitch.Session("sofia/gateway/Goip_One/+71234567890@192.168.100.28:5060 ") 2016-10-07 17:55:30.119989 [NOTICE] switch_channel.c:1104 New Channel sofia/external-g-eth0.nostun.noauth/%2B71234567890@192.168.100.28:5060 [489d5441-701d-4a98-80ca-3389b00fa98a] freeswitch@Test-Environment> # # New incoming call # 2016-10-07 17:55:33.920114 [NOTICE] switch_channel.c:1104 New Channel sofia/external-g-eth0.stun.auth/384851(a)1.2.3.4 [e11d39b8-0479-4d4b-bd8b-28634f0a970a] 2016-10-07 17:55:33.920114 [INFO] mod_dialplan_xml.c:637 Processing 384851 <384851>->318386 in context hive-context # # Execute second python script from dialplan: # 2016-10-07 17:55:33.920114 [NOTICE] mod_python.c:212 Invoking py module: python_test_script freeswitch@Test-Environment> # # 12 seconds of silence during which any python script will not be executed and will be expected to complete # freeswitch@Test-Environment> 2016-10-07 17:55:45.140231 [NOTICE] sofia.c:8078 Hangup sofia/external-g-eth0.nostun.noauth/%2B71234567890@192.168.100.28:5060 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER] 2016-10-07 17:55:45.160086 [NOTICE] switch_cpp.cpp:1360 -->(489d5441-701d-4a98-80ca-3389b00fa98a) After Create a new session through freeswitch.Session 2016-10-07 17:55:45.160086 [NOTICE] switch_core_session.c:1665 Session 15 (sofia/external-g-eth0.nostun.noauth/%2B71234567890@192.168.100.28:5060) Ended 2016-10-07 17:55:45.160086 [NOTICE] switch_core_session.c:1669 Close Channel sofia/external-g-eth0.nostun.noauth/% 2B71234567890@192.168.100.28:5060 [CS_DESTROY] # # Continue executing second python script(from 2016-10-07 17:55:33.920114) after any answer from first python new freeswitch.Session # 2016-10-07 17:55:45.160086 [ALERT] switch_cpp.cpp:1360 (mtid:3656): Traceback 280920161717(e11d39b8-0479-4d4b-bd8b-28634f0a970a): --> 12 <--; now: 1475841345; python_start_time:1475841333;

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