February 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dest./port unreachable for RTCP and first few RTP packages
by René Weiss 10 Feb '17

10 Feb '17

Hello, While looking at some traces in Wireshark I noticed some packages which generated an "Destination unreachable (Port unreachable)" ICMP answer from my server. These where: - All RTCP packages - At least in one case the first few RTP packages in a new call (6 packages over around 0.12s, after that RTP worked) So my questions are: - Should I be worried about these? - What's the deal with RTCP packages? Can/should I simply ignore them or is there an option in freeswitch that I should set to handle them? Thank you, René

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T38 488 behavior on 1.6.14
by andymurray439＠yahoo.com 10 Feb '17

10 Feb '17

Good evening, I noticed Bryan's thread on "BEHAVIOR CHANGE (YES PAY ATTENTION TO THIS IFYOU USE FAXING)" but still having issues. I'm running FreeSWITCH Version 1.6.14+git~20170130T165359Z~e9cb966d18~64bit (git e9cb966 2017-01-30 16:53:59Z 64bit) When a G711 call comes in and then the reinvite occurs from the calling side for T38; FS sends back a "488 Not Acceptable Here" I have set this in vars.xml per the suggestion: <X-PRE-PROCESS cmd="set" data="fax_enable_t38=true"/> Below is the pastebin of the siptrace: http://pastebin.com/hjnhEqym Below is the dialplan for it: <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/> <action application="hangup"/> Am I missing anything else to just make this work? Thanks! andy....

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OpenSSL + Windows build
by Gregor Nanger 09 Feb '17

09 Feb '17

I am using FS with Windows. Due to bug or compatibility in OpenSSL, I get error audio Handshake failure 1 when I try to call from verto client in Chrome. Solution is to get newer OpenSSL dlls (ssleay32.dll and libeay.dll) and replace ones that come with FS installation. Since both dlls are created during Windows build, does anyone knows how to change FS windows build to download newest source files of OpenSLL and create newest version of OpenSSL dll. Best regards, Gregor

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strange behavior of ignore_early_media
by Brandon Johnson 09 Feb '17

09 Feb '17

My freeswitch application is using early media to send a video preview from the caller to the callee before answering. This works, if I am not forking the call to multiple users. If I fork the call, only the first endpoint to receive the early media gets to answer, and the other calls are cancelled (this is expected, according to the documentation). If I set ignore_early_media to true in the dialplan, then the forking issue goes away but the early media stops working. I did some packet captures with ignore_early_media turned on and off, and the signaling doesn’t change, so I’m sure it’s not a problem on the endpoints side of things. Freeswitch just doesn’t send the RTP when it is set to true. Is there some trick I am missing, or a workaround I can try? Here’s my dialplan, it’s very straightforward: <extension name="C4 All Group"> <condition field="destination_number" expression="^All$"> <action application="set" data="ignore_early_media=true"/> <action application="bridge" data="user/000FFF0B001A,user/000FFF803637"/> </condition> </extension>

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Fwd: Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
by Michael Nielsen 09 Feb '17

09 Feb '17

I'm trying to create a voicemail, but are getting error due to: EXECUTE sofia/internal/4412345678(a)sip.nexmo.com bridge({absolute_codec_string='PCMU'}user/4487654321(a)my-domain.com) 2017-02-09 14:44:33.887109 [DEBUG] switch_ivr_originate.c:2138 Parsing global variables 2017-02-09 14:44:33.887109 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f047c0262f0 Connected. 2017-02-09 14:44:33.887109 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f047c0262f0 released. 2017-02-09 14:44:33.887109 [DEBUG] switch_ivr_originate.c:2138 Parsing global variables 2017-02-09 14:44:33.887109 [NOTICE] switch_ivr_originate.c:2841 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2017-02-09 14:44:33.887109 [DEBUG] switch_ivr_originate.c:3829 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2017-02-09 14:44:33.887109 [NOTICE] switch_ivr_originate.c:2841 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2017-02-09 14:44:33.887109 [DEBUG] switch_ivr_originate.c:3829 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2017-02-09 14:44:33.887109 [INFO] mod_dptools.c:3409 Originate Failed. Cause: USER_NOT_REGISTERED My dial plan is as follows: <condition field="${user_exists}" expression="^true$" break="never"> <action application="set" data="call_timeout=20"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="{absolute_codec_string=' PCMU'}user/${dialed_number}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_number}"/> </condition> So I guess the user should not be registered, as it is for voicemail.

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Unable to show avatar/mute image in video confererence
by Muthupandi Govindharaj 09 Feb '17

09 Feb '17

I am using conference in FreeSWITCH. The moderator should be able to join with video+audio. Non-moderators who join the conference should have avatar/image(their video should not be shown in the conference).video-mode is video-mode is mux and video-layout-name is 1x1. I tried the following. But I tried the following. I am getting a blank screen. default-avatar.png is not shown in the video. I have confirmed that the image is available in the location. <action application="answer"/> <action application="set" data="video_mute_png=${images_dir}/default-avatar.png"/> <action application="conference" data="myconf@confname++flags{non_moderator|vmute}"/> Please help me. Regards, Muthu

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update callee_id_name on pickup (using intercept)
by Piccinin Francesco 09 Feb '17

09 Feb '17

Hi all, I'm experiencing the following issue on pickup groups. We would like to update callee_id_name on pickup groups (using intercept). Please consider the following example: Extensions A and B are in the same pickup groups. Extension C is calling extension B. Extension A perform a pickup using the code *8 (please find below the instersting part of our dialplan.xml). On extension A display we see extension B callee_id_name and callee_id_number. We expect to see extension C number and name. Which channel variable to we need to update? Find below an export of our dialplan.xml Thanks. regards  <condition field="destination_number" expression="^\*8$" break="on-true"> <action application="answer"/> <action application="set" data="intercept_unbridged_only=true"/> <action application="intercept" data="${db(select/$${hostname}_last_dial_${domain_name}/${pickup_group})}"/> <action application="lua" data="functions/clean_limit_pickup.lua ${pickup_group} ${domain_name}" inline="true"/> <action application="sleep" data="2000"/> </condition> Francesco Piccinin Innovative Services Telecommunications Insiel S.p.A. via San Francesco d'Assisi, 43 34133 - Trieste (TS) Italia tel +39 040 3737752 mob +39 335 1329440 francesco.piccinin(a)insiel.it<mailto:francesco.piccinin@insiel.it> www.insiel.it<http://www.insiel.it> rispetta l'ambiente: se non ti è necessario non stampare questa mail AVVISO DI RISERVATEZZA Informazioni riservate possono essere contenute nel messaggio o nei suoi allegati. Se non siete i destinatari indicati nel messaggio, o responsabili per la sua consegna alla persona, o se avete ricevuto il messaggio per errore, siete pregati di non trascriverlo, copiarlo o inviarlo ad alcuno. In tal caso vi invitiamo a cancellare il messaggio ed i suoi allegati. Grazie. CONFIDENTIALITY NOTICE Confidential information may be contained in this message or in its attachments. If you are not the addressee indicated in this message, or responsible for message delivering to that person, or if you have received this message in error, you may not transcribe, copy or deliver this message to anyone. In that case, you should delete this message and its attachments. Thank you.

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freeswitch responding to other port for register request and fail to register
by devang nathwani 09 Feb '17

09 Feb '17

Hello, Freeswitch is getting registration request from ip:random_port and freeswitch responding to ip:5060 and that is why As shown in image 91.18.19.90:50319 is sending register request to freeswitch server 38.201.12.30 but freeswitch is responding to 91.18.19.90:5060 and device is not able to register, what is wrong here? [image: Inline image 1]

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3-party call where 2 people should not hear each other
by Oivvio Polite 09 Feb '17

09 Feb '17

I'm looking at a problem for a potential client. They want to do three party PSTN calls where all audio is NOT routed to all parties. Two parties (A and B) are in the same room but are using separate handsets rather then a speaker phone. The third party (C) is at a remote location. C should hear audio from A and B A and B should both hear audio from C but not from one another. Since they are both in the same room hearing the audio from one another in the handset as well is just distracting. My client has previously experimented with Twilio and figured out that their API does provide a solution.

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Problems with privacy settings on invite
by Christoph Russow 09 Feb '17

09 Feb '17

Hi All, i recently got informed by a collegue that it seems im not setting privacy stuff correct on our outbound calls. After investigating the mentioned call i see the following invite on the inbound leg (i xxxxxed out the last 5 digits of every phone number): INVITE sip:4980058xxxxx@88.198.223.120:5060 SIP/2.0 Via: SIP/2.0/UDP 217.110.38.229:5060;branch=z9hG4bK0dB7a5469c2220f98ee From: "Anonymous" <sip:Anonymous@Anonymous.invalid>;tag=gK0d356f75 To: <sip:4980058xxxxx@88.198.223.120> Call-ID: 1863167222_96798667(a)217.110.38.229 CSeq: 21390 INVITE Max-Forwards: 26 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: "Anonymous" <sip:Anonymous@217.110.38.229:5060> P-Asserted-Identity: <sip:499132xxxxx@217.110.38.229:5060> Privacy: id Supported: timer,100rel Session-Expires: 1800 Min-SE: 90 Content-Length: 262 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 29264 8032 IN IP4 217.110.38.229 s=SIP Media Capabilities c=IN IP4 217.110.38.228 t=0 0 m=audio 19174 RTP/AVP 18 8 100 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=sendrecv a=ptime:20 and on the Outbound Leg i see freeswitch sending the following invite to our outbound carrier: INVITE sip:49151708xxxxx@80.84.31.98 SIP/2.0 Via: SIP/2.0/UDP 88.198.223.120;rport;branch=z9hG4bKFyctt2ZcKvBaD Max-Forwards: 25 From: "Anonymous" <sip:499132xxxxx@88.198.223.120>;tag=eDe24j5Hp5cvm To: <sip:49151708xxxxx@80.84.31.98> Call-ID: 736df04f-64b0-1235-ad92-b8ac6f7db265 CSeq: 102726678 INVITE Contact: <sip:mod_sofia@88.198.223.120:5060> User-Agent: Emtex Voice Switch v1.9 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Privacy: id Content-Type: application/sdp Content-Disposition: session Content-Length: 251 X-FS-Support: update_display,send_info P-Asserted-Identity: "Anonymous" <sip:499132xxxxx@88.198.223.120> v=0 o=FreeSWITCH 1486105285 1486105286 IN IP4 88.198.223.120 s=FreeSWITCH c=IN IP4 88.198.223.120 t=0 0 m=audio 20200 RTP/AVP 8 0 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 my system is starting the outbound leg with the following command: bridge {origination_uuid=0043c244-7f46-4484-a4a1-fa8863f22321,sip_cid_type=pid,origination_privacy=hide_name:hide_number:screen}sofia/default/49151708xxxxx@80.84.31.98 The questions now are: Is there any way to prevent freeswitch from pulling the informations from the P-Asserted-Identity header into the From: header in the invite? Am i doing something wrong while setting the privacy options (any option that cancel each other out)? What is the preferred way of getting a fully anonymous invite (caller number nowhere mentioned) but correctly set P-Asserted-Identity with the caller number (required by german carriers by law) (like the inbound leg)? If you need any additional informations from me just ask. Best regards Christoph Russow -- ________________________________________________________________ EMTEX GmbH Christoph Russow Software Engineer Bischof-Otto-Weg 9 D-91086 Aurachtal Geschäftsführer: Markus Enzinger Tel. +49 9132 7490 0 Sitz der Gesellschaft: 91086 Aurachtal Fax. +49 9132 7490 900 Amtsgericht Fürth: HRB6804 ________________________________________________________________

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