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Re: [Freeswitch-users] one way speech path after attempting voice path redirecting in new invite
by Bejan, Alex 15 Feb '17

15 Feb '17

Hello, This is my first posting into this list although I monitored activities in the past. Freeswitch behaves great in our environment - thank you all who contribute! We started recently observing some one way speech path issues in the following scenario and I would like to ask for suggestions, please: We have incoming calls from provider to FS that further bridge to different Standard SIP PBX. Call is established to SIP device connected on the PBX. The PBX immediately invites back attempting to redirect rtp path to start recording the call. FS starts sending RTP to the new destination. There are 2 different very similar PBX environments connected to the same FS: - In one the PBX starts sending RTP even before the connection is ACKed. Recording succeeds. - In the second one, PBX starts sending RTP almost one second later. It seems FS reverts back to the original speech path, while the other devices remain unaware, resulting in one way speech path. The rtp-timeout-sec and rtp-hold-timeout-sec are on the default values  <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> Are there any other parameters that could have such an effect? Your suggestions on further troubleshooting or how to make this work even when RTP is not being received will be greatly appreciated! Thank you very much! Regards

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Do I need a STUN-Server?
by Thorsten Göllner 15 Feb '17

15 Feb '17

Hi, I read some docs about STUN/TURN-Servers and I understand the concept behind it (nearly). Do I need a STUN-Server when a SIP-Client (behind NAT) must connect to a Freeswitch-Server (public ip)? Or is a STUN-Server only needed if a SIP/P2P-Connection must be established? Thank in advance, Thorsten

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Voicemail Forward in MultiDomain setup
by Jai Rangi 15 Feb '17

15 Feb '17

We are using Freeswitch for a multidomain setup and experiencing problem in forwarding voice mail message to another extension in the same domain. This is what I see in the logs 2017-01-31 21:23:37.116117 [WARNING] mod_voicemail.c:3268 Cannot locate domain company.phonesystemdomain.com <http://j2h.la.cebodtelecom.com> 30a4bcaa-cd5a-4be8-88bf-94e888f98a16 2017-01-31 21:23:37.116117 [ERR] mod_voicemail.c:1768 Failed to forward message to 130 30a4bcaa-cd5a-4be8-88bf-94e888f98a16 2017-01-31 21:23:37.116117 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 30a4bcaa-cd5a-4be8-88bf-94e888f98a16 2017-01-31 21:23:37.136121 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-that_was_an_invalid_ext.wav] (en:en) Anyone else experiencing this, any fix? Not sure if this is a bug. I have experienced this in 1.4, 1.6.6, 1.6.9 and latest stable 1.6.13. Thank you, Jai

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Use inband dtmf only
by Nikolay Zaytsev 15 Feb '17

15 Feb '17

Hello, I have a problem with an inbound provider: sometimes dtmfs do not reach the freeswitch. Provider told me that occurred because I am using G711 with rfc2833, and I should use only G711. Inbound call from the carrier to FreeSwitch: SDP in the INVITE carrier --> FreeSwitch v=0 o=Sansay-VSXi 188 1 IN IP4 x.x.x.x s=Session Controller p=+X XXXXXXXX c=IN IP4 x.x.x.x t=0 0 m=audio 17774 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 SDP in the 200 OK message from FreeSwitch-->carrier: v=0 o=FreeSWITCH 1486794789 1486794790 IN IP4 x.x.x.x s=FreeSWITCH c=IN IP4 x.x.x.x t=0 0 m=audio 30088 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Carrier wants me to get rid of those session attributes in my 200 OK message: a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 and media to be like this m=audio 30088 RTP/AVP 0. Could you please advise me how to do it? I have already tried to set up dtmf-type=none and liberal-dtmf=false in SIP profiles, dtmf_type=none in Dialplan. In addition I tried start_dtmf before application answer. Also in the vars.xml I specified to use PCMU and PCMA only for both global_codec_prefs and outbound_codec_prefs. My dialplan instruction for inbound call: <extension name="inbound_call" continue="false"> <condition field="destination_number" expression="^79" > <action application="answer"/> <action application="sleep" data="1000"/> <action application="socket" data="127.0.0.1:8800 async full"/> </condition> </extension> Im using several FreeSwitch boxes, one of them is 1.6.2 version another one is 1.6.14, but this problem occurres on both. Thank you for your time and best regards, Nikolay Zaytsev

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Strip and use SIP headers in freeswitch dialplan configuration
by vivek＠advaitamtech.com 15 Feb '17

15 Feb '17

Hi All, Am trying to fetch "Proxy-Authorization" header present in the SIP INVITE request through dialplan xml. Since there is no Built-in variable to fetch this header, is there a way to find if this header is present in the SIP INVITE request?. I browsed to find any possible solution but wasnt successful. Thanks, Vivek

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spandsp_start_fax_detect sometimes miss fax detection
by Yehavi Bourvine 15 Feb '17

15 Feb '17

Hello, I am using the above application to detect an incoming fax. My dialplan has the following code: <action application="set" data="execute_on_answer=spandsp_start_fax_detect transfer 'fax_${dest}' XML ${context}' 10 cng"/> When using the same fax machine to send, sometimes it detects the fax, sometimes not. This mainly happens when the call lands into voicemail; when answering the call the success rate is much higher (almost 100%). Is there any way to "tune" this detection? Or shall I use something different than "execute_on_answer" for voicemail? Thanks! __Yehavi:

1 0

incoming call going to the wrong context
by David Villasmil 14 Feb '17

14 Feb '17

Hello guys, So i have a gw in external profile, this gw's context is "default" which doesn't exist, still when a call comes in, fs sends it to public context (which is the context on external.xml), why is that? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Freeswitch sends 2 SDP answers which causes webrtc client to drop the call
by Georgi Nikolaev 14 Feb '17

14 Feb '17

Hi Folks, I am experiencing an issue where remote SDP can not be set due to receiving 2 SDP answers. The setup is webrtc (sipjs)->Freeswitch->SIP Termination, where a call is originated by the sipjs.ua calling a public number. As soon as the remote party answers the call, the call drops due to not being able to gather the remote media path. Worth mentioning is that only some calls fail where the majority are successful. The difference between the successful and the unsuccessful ones is in the SDP messages. The good one receive ‘answer' for UDP/TLS/RTP/SAVPF followed by ‘offer’ for RTP/SAVPF. The one that fail are 2 SDP answers. I have discussed the topic on the webrtc group: https://groups.google.com/forum/?utm_medium=email&utm_source=footer#!msg/di… Also raised this in the sipjs forum: https://groups.google.com/forum/?utm_medium=email&utm_source=footer#!topic/… On both it has been suggested that this is either a setting in Freeswitch or its incorrectly sent SDP messages. Freeswitch version: 1.6.14 SIP.js: 0.7.5 Browser: Chrome - latest version I would appreciate any help on this as its causing me some random call drops affecting the service. Here are the logs as well: FS log: http://pastebin.com/UQW6AdcQ Chrome log: http://pastebin.com/Crk69MrS webrtc-internals-log: http://pastebin.com/BVr7v4EB Any ideas/suggestions/comments are welcome. Thank you in advance. Best Regards, Georgi Nikolaev

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Playing the loudest participant in a conference and muting the others.
by Oivvio Polite 14 Feb '17

14 Feb '17

Is there a way to only hear the currently loudest speaker in a conference? I know about "energy-level" that let's me suppress silent channels that don't reach a certain level. But that's not quite what I want. Rather I want the output of each channel to be dependent on the current level of the other channels. Oivvio

2 1

FreeSwitch originate api + early_media timeout
by Vyronas Tsingaras 14 Feb '17

14 Feb '17

Hi, Is there a way to use bridge_answer_timeout when using the originate API? For example something like: api originate {origination_uuid=XXXX,ignore_early_media=true,answer_ timeout=5}[]sofia/gateway/ to timeout 5 seconds AFTER going into Callstate EARLY. Kind Regards, * Vyronas Tsingaras * Network & Systems Administrator vtsingaras(a)amdtelecom.net www.amdtelecom.net

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