- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch doesnt transcode
by Rajil Saraswat 15 Feb '16

15 Feb '16

Hello, I have a remote sip phone (Linksys SPA3102) which only supports PCMU. When I call to this remote sip phone i get a 406 error that opus is not supported as shown by the sip trace below. However, if I force the codec to absolute like this {absolute_codec_string='PCMU,PCMA'}sofia/internal/303(a)192.168.1.5 the call works fine. Is there anyway I can make FreeSWITCH to automatically transcode without forcing the codec string in the dial plan? The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and outbound_codec_prefs=PCMU,PCMA,GSM ---------------------------siptrace-------------------------------- recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499: ------------------------------------------------------------------------ SIP/2.0 406 Not Acceptable Via: SIP/2.0/UDP 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q CSeq: 87372504 INVITE Content-Length: 0 ------------------------------------------------------------------------ send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591: ------------------------------------------------------------------------ ACK sip:303@192.168.1.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej Max-Forwards: 68 From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124 CSeq: 87372504 ACK Content-Length: 0 ------------------------------------------------------------------------ 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel sofia/internal/303(a)192.168.1.5 entering state [terminated][406] 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup sofia/internal/303(a)192.168.1.5 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] Thanks Rajil

11 20

bind_digit_action in a conference
by Matt Broad 15 Feb '16

15 Feb '16

Hi, is it possible to bind a digit action and set the target as another member of a conference? I would like to be able to press a digit and have another member execute an extension where they enter some digits before returning back to the conference. thanks Matt

2 2

FS Bridged call, no RTP until DTMF pressed
by Adnan Ahmed 15 Feb '16

15 Feb '16

Hi, I have a peculiar situation in which I'm hoping someone can help me out with. I have a Dahdi trunk coming into Asterisk (*), which then sends the call directly to freeswitch (FS), FS will then bridge this incoming call to a SIP device. The problem i'm having is that when FS bridges the call there is no media (or RTP packets) sent back to asterisk until I press a dtmf key from the caller side. The reason that * is there is due to the fact that mod_freeTDM for FS wasn't able to configure the trunk parameters required to control the T1 (E&M with Feature Group B MF), with chan_dahdi in * i was able to set that up with signalling=featb. The dialplan in asterisk is as follows, [from-pstn] > exten => _X.,1,NoOp(Incoming DID matches as ${EXTEN}) > exten => _X.,n,Answer() > exten => _X.,n,Set(CALLERID(all)="0000000000"<0000000000>) > exten => _X.,n,Dial(SIP/freeswitch/1819${EXTEN:0:7},90,M(send-dtmf-1)r) > exten => _X.,n,Hangup() > > [macro-send-dtmf-1] > exten => s,1,SendDTMF(1) I tried sending a DTMF from astersk, and FS recognizes the DTMF, but still no RTP until the key is physically pressed on the caller side. The asterisk dialplan is very simple, answer the incoming dahdi call and send it to FS via SIP. Once the DTMF is pressed, the audio is complete and no issues anymore, so its not a routing, or firewall issue. Both asterisk and FS run on the same machine (* on port 5065, and FS on 5060). Looking at the tcpdump traces, there really is no RTP from FS until after the DTMF is pressed, but the RTP from asterisk is always there. I have the output of "sofia global siptrace on" at the following pastebin: https://pastebin.freeswitch.org/24552 In that SIP trace you will see the call as follows, Incoming call from * bridge to SIP device Failure to connect to SIP device Forward call to voicemail bridge to voicemail connects to voicemail system hangup I can press the DTMF at any point once the first bridge is dialed and will start hearing the audio from that point onwards ... in this case i pressed the DTMF key 1 (you see it being recognized in the FS sip trace log). It makes no difference if I wait to press the DTMF till the second bridge or after the second bridge connects. I have even tried it with a sip device that answers on the first bridge session, and its the same scenario: no audio until dtmf is pressed, again making no difference if its pressed right away or 10 seconds after the call is connected and the other party can hear me but i don't hear them until i press the dtmf. Thanks, Adnan.

1 1

Call recording audio too fast
by Rutu Patel 13 Feb '16

13 Feb '16

Hi All, I am using record_session application to record the calls but the audio speed has too fast in recording files. Below is the code I am using for call recording: <action application="set" data="RECORD_STEREO=true"/> <action application="set" data="record_sample_rate=8000"/> <action application="export" data="execute_on_answer=record_session $${base_dir}/recordings/${uuid}.wav"> Can anyone please help me to solve the issue with fast audio speed in recording files ? Also is it possible to decrease the size of .wav audio file ? -- Thanks, Rutu Patel <http://www.inextrix.com>

2 3

Get actual duration of the call after hangup
by Alexandru Covalschi 13 Feb '16

13 Feb '16

Hello folks! I have a problem - I need to get actual duration of the call - I mean from ANSWER to HANGUP, without the time user waited being parked on the bridge. The thing is I need to write that info into a custom table which belongs to a custom database where I have leg_a uuid. How can I achieve that in case of: 1. Outoing origination via bridge() 2. Connecting two customers which are already inside a freeswitch via bridge() I've read about mod_cdr_csv but it afaik doesn't allow to connect to a custom db. Would mod_odbc_cdr allow me to do that? And if yes - can you tell me "chan-var-name"? And don't be afraid - such slow method is not for billing :) Just to provide customer more detailed calls history. Thanks! -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 1

Freeswitch in high available clustered environment
by Varsha Agarwal 12 Feb '16

12 Feb '16

Hi All, Is there a good documentation on how to setup Freeswitch in a clustered environment with a redundant node as well? I am looking through Wiki but there no one good article I found that has it all. Thanks, Varsha

3 3

FUND THE PARTY! @ FreeSWITCH Annual Core Devs Meeting
by Giovanni Maruzzelli 12 Feb '16

12 Feb '16

How much we all made in 2015 thanks to FreeSWITCH? How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions? Shovel your HARD EARNED MONEY to our HARD PARTYING and HARD CODING CORE DEVS ! BUY THEM LUNCH AND REFRESHMENTS ! HERE! ==> https://www.gofundme.com/freeswitch If not now, when? Because! -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

2 1

fs_path adding route header
by Aqs Younas 12 Feb '16

12 Feb '16

Hi, I am using fs_path to proxy Invite to specified proxy. But I see fs_path adding route header in Invite. As B2BUA freeswitch must be generating new call instead of adding route header. Is this intended behavior? Or something wrong with my configuration Best Regards.

1 1

How to add both rpid and pid in freeswitch.
by Aqs Younas 12 Feb '16

12 Feb '16

Hi, I am bridging Invite containing both RPID and PID to some destination. Like <action application="bridge" data="{sip_cid_type=rpid,origination_caller_id_name=_undef_}sofia/external/${sip_req_uri};${sip_req_params};fs_path=sip:${Dest}"/> Currently, I see freeswitch can add only one header PID or RPID but not both with sip_cid_type variable. How can i Add both? Remote-Party-ID: <sip:14248271614@192.168.10.25>;privacy=off;screen=yes P-Asserted-Identity: <sip:14248271614@192.168.10.25;user=phone> Any suggestion is much appreciate. Best Regards.

2 2

Over the half way mark!!! :D
by Blake Priddy 12 Feb '16

12 Feb '16

They are over the half way mark! Thanks y'all. Keep it coming!! :) https://www.gofundme.com/freeswitch

1 0