September 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

No ringtone to caller when third party call control (3PCC) is enabled
by Jaan Kaja 12 Sep '18

12 Sep '18

Hi, I have configured "enable-3pcc=proxy" in the SIP profile, and have an external application that controls answer-call, make-call, etc. The configuration means that incoming calls w/o SDP are accepted, which works fine. The problem is that when I make an outgoing call, ringtone is not played to the caller, although the "ringback" variable is set for the caller's leg. I have checked the code in mod_sofia, and in sofia.c, there is a piece of code: if (sofia_test_flag(tech_pvt, TFLAG_SKIP_EARLY)) { sofia_clear_flag_locked(tech_pvt, TFLAG_SKIP_EARLY); goto done; } which disables reaching the ring-ready state. TFLAG_SKIP_EARLY is set when TFLAG_3PCC is true. If I comment out the goto statement, the A-party gets ringtone. N.B.: This is just a minimal quick fix, not a proposed solution, which works because I'm not doing an enterprise originate. Question: Is there a reason for this behavior? Documentation shows that early media must be disabled when an enterprise originate is done, but why is playing ringtone disabled in this case? Should I submit a bug report to Jira? Best regards, Jaan

2 1

Problem with NAT & bypass media configuration
by Anacelia Sarlo 12 Sep '18

12 Sep '18

Hi, I have the following environment : FS (ethernet interface with public address -190.64.204.60) ---- FW (provider) ------------- INTERNET ---- Router --NAT/FW----LAN Softphone (type A) INTERNET ---- Mobile network ---- Softpone (type B) Freeswitch v1.6.20, on Centos 7, behind a firewall, with sip ports tcp/udp 5560 already opened, and it is not natting. I have two type of SIP devices: Type A) connected to a LAN behind a router/firewall, natted and with upnp enabled Type B) connected through the mobile network, also natted. Calls (audio / video) between two A endpoints works!, but calls between A and B endpoints are established, but there is no audio / video, and after 30 seconds it hungup. I'am using FS for SIP signaling, and bypassing media. I tried many softphones (Linphone, antisip, and more) without luck. Also enable stun at both softphones, but I have the same result. FS is not using nat functionality (no UPNP/PMP). I have just one sip profile (external) with port 5560. I'am attaching the configuration. I suppose it's something related to RTP and NAT but I tried many configurations, and can't resolve it. Thanks a lot for your help!. Regards, Anacelia vars.xml <include> <X-PRE-PROCESS cmd="set" data="default_password=otrapassword"/> <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="domain=190.64.204.60"/> <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/> <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/> <X-PRE-PROCESS cmd="set" data="use_profile=external"/> <X-PRE-PROCESS cmd="set" data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/> <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,H264,VP8"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,H264,VP8"/> <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/> <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/> <X-PRE-PROCESS cmd="set" data="bind_server_ip=127.0.0.1"/> <X-PRE-PROCESS cmd="set" data="unroll_loops=true"/> <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/> <X-PRE-PROCESS cmd="set" data="call_debug=false"/> <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/> <X-PRE-PROCESS cmd="set" data="default_areacode=918"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="presence_privacy=false"/> <X-PRE-PROCESS cmd="set" data="au-ring=%(400,200,383,417);%(400,2000,383,417)"/> <X-PRE-PROCESS cmd="set" data="be-ring=%(1000,3000,425)"/> <X-PRE-PROCESS cmd="set" data="ca-ring=%(2000,4000,440,480)"/> <X-PRE-PROCESS cmd="set" data="cn-ring=%(1000,4000,450)"/> <X-PRE-PROCESS cmd="set" data="cy-ring=%(1500,3000,425)"/> <X-PRE-PROCESS cmd="set" data="cz-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="de-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="dk-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="dz-ring=%(1500,3500,425)"/> <X-PRE-PROCESS cmd="set" data="eg-ring=%(2000,1000,475,375)"/> <X-PRE-PROCESS cmd="set" data="es-ring=%(1500,3000,425)"/> <X-PRE-PROCESS cmd="set" data="fi-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440)"/> <X-PRE-PROCESS cmd="set" data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/> <X-PRE-PROCESS cmd="set" data="hu-ring=%(1250,3750,425)"/> <X-PRE-PROCESS cmd="set" data="il-ring=%(1000,3000,400)"/> <X-PRE-PROCESS cmd="set" data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/> <X-PRE-PROCESS cmd="set" data="jp-ring=%(1000,2000,420,380)"/> <X-PRE-PROCESS cmd="set" data="ko-ring=%(1000,2000,440,480)"/> <X-PRE-PROCESS cmd="set" data="pk-ring=%(1000,2000,400)"/> <X-PRE-PROCESS cmd="set" data="pl-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="ro-ring=%(1850,4150,475,425)"/> <X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425)"/> <X-PRE-PROCESS cmd="set" data="sa-ring=%(1200,4600,425)"/> <X-PRE-PROCESS cmd="set" data="tr-ring=%(2000,4000,450)"/> <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/> <X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440,480)"/> <X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/> <X-PRE-PROCESS cmd="set" data="beep=%(1000,0,640)"/> <X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> <X-PRE-PROCESS cmd="set" data="df_us_ssn=(?!219099999|078051120)(?!666|000|9\d{2})\d{3}(?!00)\d{2}(?!0{4})\d{4}"/> <X-PRE-PROCESS cmd="set" data="df_luhn=?:4[0-9]{12}(?:[0-9]{3})?|5[1-5][0-9]{14}|3[47][0-9]{13}|3(?:0[0-5]|[68][0-9])[0-9]{11}|6(?:011|5[0-9]{2})[0-9]{12}|(?:2131|1800|35\d{3})\d{11}"/> <X-PRE-PROCESS cmd="set" data="digits_dialed_filter=(($${df_luhn})|($${df_us_ssn}))"/> <X-PRE-PROCESS cmd="set" data="default_provider=example.com"/> <X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/> <X-PRE-PROCESS cmd="set" data="default_provider_password=password"/> <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/> <X-PRE-PROCESS cmd="set" data="default_provider_register=false"/> <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/> <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/> <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5560"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=1mb"/> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=1mb"/> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> <X-PRE-PROCESS cmd="set" data="rtp_liberal_dtmf=true"/> <X-PRE-PROCESS cmd="set" data="video_mute_png=$${images_dir}/default-mute.png"/> <X-PRE-PROCESS cmd="set" data="video_no_avatar_png=$${images_dir}/default-avatar.png"/> </include> sip_profiles/external.xml <profile name="external"> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="yes"/> <param name="sip-capture" value="no"/> <param name="sip-port" value="5560"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="dtmf-duration" value="2000"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="auth-calls" value="false"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-bypass-media" value="true"/> <param name="rtp-ip" value="190.64.204.60"/> <param name="sip-ip" value="190.64.204.60"/> <param name="ext-rtp-ip" value="190.64.204.60"/> <param name="ext-sip-ip" value="190.64.204.60"/> <param name="rtp-autofix-timing" value="false"/> </settings> </profile> dialplan/default.xml <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="extension-portero"> <condition field="destination_number" expression="^(portero)$"> <action application="set" data="inherit_codec=true"/> <action application="export" data="nolocal:absolute_codec_string=${ep_codec_string}"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/$1"/> <action application="answer"/> <action application="sleep" data="1000"/> </condition> </extension> <extension name="Local_Extension_anacelia"> <condition field="destination_number" expression="^(anacelia)$"> <action application="set" data="inherit_codec=true"/> <action application="export" data="absolute_codec_string=PCMU,PCMA,H264"/> <action application="set" data="bypass_media=true"/>  <action application="bridge" data="user/$1"/> <action application="answer"/> <action application="sleep" data="1000"/> </condition> </extension> </context> </include> --------------------- Sip trace (Softphone B call Softphone A) ------------ Type A softphone= portero(a)190.64.204.60 Type B softphone= anacelia(a)190.64.204.60 freeswitch@Vps-230738> freeswitch@Vps-230738> freeswitch@Vps-230738> freeswitch@Vps-230738> freeswitch@Vps-230738> recv 539 bytes from udp/[167.61.200.85]:5560 at 17:31:20.410606: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport;branch=z9hG4bKPjMq4r69tHAXghmW42wdZsTOt9QV0IO2-0 Route: <sip:190.64.204.60:5560;lr> Max-Forwards: 70 From: "Portero" <sip:portero@190.64.204.60 >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA To: "Portero" <sip:portero@190.64.204.60> Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47401 REGISTER User-Agent: Beward SIP Contact: <sip:portero@167.61.200.85:5560;ob> Expires: 300 Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 644 bytes to udp/[167.61.200.85]:5560 at 17:31:20.411630: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport=5560;branch=z9hG4bKPjMq4r69tHAXghmW42wdZsTOt9QV0IO2-0 From: "Portero" <sip:portero@190.64.204.60 >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA To: "Portero" <sip:portero@190.64.204.60>;tag=ae7Sv6D29B5DH Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47401 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces WWW-Authenticate: Digest realm="190.64.204.60", nonce="a4650f68-b601-11e8-9841-cdce49bc5d42", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 806 bytes from udp/[167.61.200.85]:5560 at 17:31:20.423845: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport;branch=z9hG4bKPjjyqnVssY4nrxRQeLok.W2V8yJGaw2k37 Route: <sip:190.64.204.60:5560;lr> Max-Forwards: 70 From: "Portero" <sip:portero@190.64.204.60 >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA To: "Portero" <sip:portero@190.64.204.60> Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47402 REGISTER User-Agent: Beward SIP Contact: <sip:portero@167.61.200.85:5560;ob> Expires: 300 Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Authorization: Digest username="portero", realm="190.64.204.60", nonce="a4650f68-b601-11e8-9841-cdce49bc5d42", uri="sip:190.64.204.60:5560", response="f5a8e2e4021c06ed681121f5a0deaf1d", algorithm=MD5, cnonce="VDG.OZQvmqtAqPh2ZLaUpgyBhLhG5szL", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 607 bytes to udp/[167.61.200.85]:5560 at 17:31:20.426835: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport=5560;branch=z9hG4bKPjjyqnVssY4nrxRQeLok.W2V8yJGaw2k37 From: "Portero" <sip:portero@190.64.204.60 >;tag=2zorZWBP.dMgngW5f47cKTAmysnKYjLA To: "Portero" <sip:portero@190.64.204.60>;tag=BQ0jy1y56mU0c Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47402 REGISTER Contact: <sip:portero@167.61.200.85:5560;ob>;expires=30 Date: Tue, 11 Sep 2018 20:31:20 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 771 bytes to udp/[167.61.200.85]:5560 at 17:31:20.488658: ------------------------------------------------------------------------ NOTIFY sip:portero@167.61.200.85:5560;ob SIP/2.0 Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bKZcm1vQ4N7ajyN Max-Forwards: 70 From: <sip:portero@190.64.204.60>;tag=c0SB0vF93XHKr To: <sip:portero@190.64.204.60> Call-ID: 7bcf2a15-30a4-1237-4894-00505601018a CSeq: 128012876 NOTIFY Contact: <sip:mod_sofia@190.64.204.60:5560> User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Event: message-summary Allow-Events: talk, hold, conference, refer Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 68 Messages-Waiting: no Message-Account: sip:portero@190.64.204.60 ------------------------------------------------------------------------ recv 319 bytes from udp/[167.61.200.85]:5560 at 17:31:20.832843: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bKZcm1vQ4N7ajyN Call-ID: 7bcf2a15-30a4-1237-4894-00505601018a From: <sip:portero@190.64.204.60>;tag=c0SB0vF93XHKr To: <sip:portero@190.64.204.60>;tag=z9hG4bKZcm1vQ4N7ajyN CSeq: 128012876 NOTIFY Content-Length: 0 ------------------------------------------------------------------------ recv 1346 bytes from udp/[186.51.236.123]:38474 at 17:31:21.388899: ------------------------------------------------------------------------ INVITE sip:portero@190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.tPc8Rktcw;rport From: <sip:anacelia@190.64.204.60>;tag=v0lM8TzLu To: sip:portero@190.64.204.60 CSeq: 20 INVITE Call-ID: thpKqQo3Kf Max-Forwards: 70 Supported: replaces, outbound, gruu Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 496 Contact: <sip:anacelia@186.51.236.123:38474 ;app-id=929724111839;pn-type=firebase;pn-tok=cofXHCv3YgA:APA91bFPHHgimCnRO8esdZvDgLWpitTLPMf1d7vn50EWLZEIeM95CBRbf3DHZrgLNv5dePml__0AYgd7jcE56-5dw-rWVOtEqiVKtBojE8c9kezaTvFFNohHrs1zdq8FAqgUxFBxaiEP;pn-silent=1;transport=udp>;+sip.instance="<urn:uuid:0707dcae-0314-40d0-90e2-79e6063cf202>";+org.linphone.specs=groupchat User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) v=0 o=anacelia 1261 1974 IN IP4 10.45.210.97 s=Talk c=IN IP4 10.45.210.97 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:97 speex/16000 a=fmtp:97 vbr=on a=rtpmap:98 speex/8000 a=fmtp:98 vbr=on a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/48000 a=rtpmap:99 telephone-event/16000 a=rtpmap:100 telephone-event/8000 a=rtcp-fb:* ccm tmmbr ------------------------------------------------------------------------ send 305 bytes to udp/[186.51.236.123]:38474 at 17:31:21.389159: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 From: <sip:anacelia@190.64.204.60>;tag=v0lM8TzLu To: sip:portero@190.64.204.60 Call-ID: thpKqQo3Kf CSeq: 20 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:21.377624 [NOTICE] switch_channel.c:1104 New Channel sofia/external/anacelia(a)190.64.204.60 [a4fa5a3c-b601-11e8-9842-cdce49bc5d42] 2018-09-11 17:31:21.377624 [INFO] mod_dialplan_xml.c:637 Processing anacelia <anacelia>->portero in context default 2018-09-11 17:31:21.377624 [NOTICE] switch_channel.c:1104 New Channel sofia/external/portero@167.61.200.85:5560 [a4fab0e0-b601-11e8-984c-cdce49bc5d42] send 1292 bytes to udp/[167.61.200.85]:5560 at 17:31:21.392377: ------------------------------------------------------------------------ INVITE sip:portero@167.61.200.85:5560;ob SIP/2.0 Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK0NDtyjNS4K8gH Max-Forwards: 69 From: "anacelia" <sip:anacelia@190.64.204.60>;tag=ejcX3jHgyFyrF To: <sip:portero@167.61.200.85:5560;ob> Call-ID: 7c590ee4-30a4-1237-4894-00505601018a CSeq: 128012876 INVITE Contact: <sip:mod_sofia@190.64.204.60:5560> User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 496 X-FS-Support: update_display,send_info Remote-Party-ID: "anacelia" <sip:anacelia@190.64.204.60 >;party=calling;screen=yes;privacy=off v=0 o=anacelia 1261 1974 IN IP4 10.45.210.97 s=Talk c=IN IP4 10.45.210.97 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7076 RTP/AVP 96 97 98 0 8 18 101 99 100 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:97 speex/16000 a=fmtp:97 vbr=on a=rtpmap:98 speex/8000 a=fmtp:98 vbr=on a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/48000 a=rtpmap:99 telephone-event/16000 a=rtpmap:100 telephone-event/8000 a=rtcp-fb:* ccm tmmbr ------------------------------------------------------------------------ recv 313 bytes from udp/[167.61.200.85]:5560 at 17:31:21.447821: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH Call-ID: 7c590ee4-30a4-1237-4894-00505601018a From: "anacelia" <sip:anacelia@190.64.204.60>;tag=ejcX3jHgyFyrF To: <sip:portero@167.61.200.85;ob> CSeq: 128012876 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 453 bytes from udp/[167.61.200.85]:5560 at 17:31:21.466797: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH Call-ID: 7c590ee4-30a4-1237-4894-00505601018a From: "anacelia" <sip:anacelia@190.64.204.60>;tag=ejcX3jHgyFyrF To: <sip:portero@167.61.200.85;ob>;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R CSeq: 128012876 INVITE Contact: <sip:portero@167.61.200.85:5560;ob> Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:21.457617 [NOTICE] sofia.c:7192 Ring-Ready sofia/external/ portero@167.61.200.85:5560! 2018-09-11 17:31:21.457617 [NOTICE] mod_sofia.c:2273 Ring-Ready sofia/external/anacelia(a)190.64.204.60! send 675 bytes to udp/[186.51.236.123]:38474 at 17:31:21.473235: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 From: <sip:anacelia@190.64.204.60>;tag=v0lM8TzLu To: <sip:portero@190.64.204.60>;tag=D9j41Q0c1675K Call-ID: thpKqQo3Kf CSeq: 20 INVITE Contact: <sip:portero@190.64.204.60:5560;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "Outbound Call" <sip:portero@190.64.204.60 >;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2018-09-11 17:31:21.457617 [NOTICE] switch_ivr_originate.c:525 Ring Ready sofia/external/anacelia(a)190.64.204.60! recv 817 bytes from udp/[167.61.200.85]:5560 at 17:31:22.018913: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bK0NDtyjNS4K8gH Call-ID: 7c590ee4-30a4-1237-4894-00505601018a From: "anacelia" <sip:anacelia@190.64.204.60>;tag=ejcX3jHgyFyrF To: <sip:portero@167.61.200.85;ob>;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R CSeq: 128012876 INVITE Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Contact: <sip:portero@167.61.200.85:5560;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 287 v=0 o=- 3745675951 3745675952 IN IP4 192.168.1.139 s=Door station call b=AS:84 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 0 100 c=IN IP4 192.168.1.139 b=TIAS:64000 a=rtcp:4003 IN IP4 192.168.1.139 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 ------------------------------------------------------------------------ send 413 bytes to udp/[167.61.200.85]:5560 at 17:31:22.019799: ------------------------------------------------------------------------ ACK sip:portero@167.61.200.85:5560;ob SIP/2.0 Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK1y6j0D6v1vy3c Max-Forwards: 70 From: "anacelia" <sip:anacelia@190.64.204.60>;tag=ejcX3jHgyFyrF To: <sip:portero@167.61.200.85:5560 ;ob>;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R Call-ID: 7c590ee4-30a4-1237-4894-00505601018a CSeq: 128012876 ACK Contact: <sip:mod_sofia@190.64.204.60:5560> Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:22.017623 [NOTICE] sofia.c:8180 Channel [sofia/external/ portero@167.61.200.85:5560] has been answered 2018-09-11 17:31:22.017623 [NOTICE] switch_ivr.c:779 Channel [sofia/external/anacelia(a)190.64.204.60] has been answered send 983 bytes to udp/[186.51.236.123]:38474 at 17:31:22.035733: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.tPc8Rktcw;rport=38474;received=186.51.236.123 From: <sip:anacelia@190.64.204.60>;tag=v0lM8TzLu To: <sip:portero@190.64.204.60>;tag=D9j41Q0c1675K Call-ID: thpKqQo3Kf CSeq: 20 INVITE Contact: <sip:portero@190.64.204.60:5560;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 275 Remote-Party-ID: "Outbound Call" <sip:portero@190.64.204.60 >;party=calling;privacy=off;screen=no v=0 o=- 3745675951 3745675952 IN IP4 192.168.1.139 s=Door station call b=AS:84 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 0 100 c=IN IP4 192.168.1.139 b=TIAS:64000 a=rtpmap:0 PCMU/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=rtcp:4003 IN IP4 192.168.1.139 ------------------------------------------------------------------------ recv 335 bytes from udp/[186.51.236.123]:38474 at 17:31:22.426439: ------------------------------------------------------------------------ ACK sip:portero@190.64.204.60:5560;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.45.210.97:39245;rport;branch=z9hG4bK.bkN7s~~XL From: <sip:anacelia@190.64.204.60>;tag=v0lM8TzLu To: <sip:portero@190.64.204.60>;tag=D9j41Q0c1675K CSeq: 20 ACK Call-ID: thpKqQo3Kf Max-Forwards: 70 User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) ------------------------------------------------------------------------ recv 539 bytes from udp/[167.61.200.85]:5560 at 17:31:45.458386: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport;branch=z9hG4bKPjUwdgBl5lbYRd0Bd4NuXDuEbeyCmfnmry Route: <sip:190.64.204.60:5560;lr> Max-Forwards: 70 From: "Portero" <sip:portero@190.64.204.60 >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 To: "Portero" <sip:portero@190.64.204.60> Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47403 REGISTER User-Agent: Beward SIP Contact: <sip:portero@167.61.200.85:5560;ob> Expires: 300 Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 644 bytes to udp/[167.61.200.85]:5560 at 17:31:45.459222: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport=5560;branch=z9hG4bKPjUwdgBl5lbYRd0Bd4NuXDuEbeyCmfnmry From: "Portero" <sip:portero@190.64.204.60 >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 To: "Portero" <sip:portero@190.64.204.60>;tag=FU5N5D2KUrmBB Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47403 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces WWW-Authenticate: Digest realm="190.64.204.60", nonce="b3530bec-b601-11e8-9854-cdce49bc5d42", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 806 bytes from udp/[167.61.200.85]:5560 at 17:31:45.477423: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport;branch=z9hG4bKPjilUCDP-iNtMqCSZS.-2dsZ4-bG809p9w Route: <sip:190.64.204.60:5560;lr> Max-Forwards: 70 From: "Portero" <sip:portero@190.64.204.60 >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 To: "Portero" <sip:portero@190.64.204.60> Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47404 REGISTER User-Agent: Beward SIP Contact: <sip:portero@167.61.200.85:5560;ob> Expires: 300 Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS Authorization: Digest username="portero", realm="190.64.204.60", nonce="b3530bec-b601-11e8-9854-cdce49bc5d42", uri="sip:190.64.204.60:5560", response="01447ae196414735cbe9fa0da2048d18", algorithm=MD5, cnonce="VDG.OZQvmqtAqPh2ZLaUpgyBhLhG5szL", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 607 bytes to udp/[167.61.200.85]:5560 at 17:31:45.479598: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 167.61.200.85:5560 ;rport=5560;branch=z9hG4bKPjilUCDP-iNtMqCSZS.-2dsZ4-bG809p9w From: "Portero" <sip:portero@190.64.204.60 >;tag=iH-xQxEm2nGbTCZdSh8Z2HzbPZ0qq8Y6 To: "Portero" <sip:portero@190.64.204.60>;tag=g4ye78jQr1ayp Call-ID: sx0F7bTRa.8f.MUlhASdyRD-LIrIvxsE CSeq: 47404 REGISTER Contact: <sip:portero@167.61.200.85:5560;ob>;expires=30 Date: Tue, 11 Sep 2018 20:31:45 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 335 bytes from udp/[186.51.236.123]:38474 at 17:31:53.066351: ------------------------------------------------------------------------ BYE sip:portero@190.64.204.60:5560;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.MOC0o5vx8;rport From: <sip:anacelia@190.64.204.60>;tag=v0lM8TzLu To: <sip:portero@190.64.204.60>;tag=D9j41Q0c1675K CSeq: 21 BYE Call-ID: thpKqQo3Kf Max-Forwards: 70 User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) ------------------------------------------------------------------------ 2018-09-11 17:31:53.057623 [NOTICE] sofia.c:1012 Hangup sofia/external/ anacelia(a)190.64.204.60 [CS_HIBERNATE] [NORMAL_CLEARING] send 442 bytes to udp/[186.51.236.123]:38474 at 17:31:53.067386: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.MOC0o5vx8;rport=38474;received=186.51.236.123 From: <sip:anacelia@190.64.204.60>;tag=v0lM8TzLu To: <sip:portero@190.64.204.60>;tag=D9j41Q0c1675K Call-ID: thpKqQo3Kf CSeq: 21 BYE User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:53.057623 [NOTICE] switch_ivr_bridge.c:1302 Hangup sofia/external/portero@167.61.200.85:5560 [CS_HIBERNATE] [NORMAL_CLEARING] send 586 bytes to udp/[167.61.200.85]:5560 at 17:31:53.068575: ------------------------------------------------------------------------ BYE sip:portero@167.61.200.85:5560;ob SIP/2.0 Via: SIP/2.0/UDP 190.64.204.60:5560;rport;branch=z9hG4bK27ZB28p0y5mpr Max-Forwards: 70 From: "anacelia" <sip:anacelia@190.64.204.60>;tag=ejcX3jHgyFyrF To: <sip:portero@167.61.200.85:5560 ;ob>;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R Call-ID: 7c590ee4-30a4-1237-4894-00505601018a CSeq: 128012877 BYE User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1683 Session 3 (sofia/external/anacelia(a)190.64.204.60) Ended 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/anacelia(a)190.64.204.60 [CS_DESTROY] 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1683 Session 4 (sofia/external/portero@167.61.200.85:5560) Ended 2018-09-11 17:31:53.057623 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/portero@167.61.200.85:5560 [CS_DESTROY] recv 343 bytes from udp/[167.61.200.85]:5560 at 17:31:53.083598: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 190.64.204.60:5560 ;rport=5560;received=190.64.204.60;branch=z9hG4bK27ZB28p0y5mpr Call-ID: 7c590ee4-30a4-1237-4894-00505601018a From: "anacelia" <sip:anacelia@190.64.204.60>;tag=ejcX3jHgyFyrF To: <sip:portero@167.61.200.85;ob>;tag=4KVi0Vn0xR3iytu9Tjf9Mm4VmUT9BG.R CSeq: 128012877 BYE Content-Length: 0 ------------------------------------------------------------------------ recv 1062 bytes from udp/[186.51.236.123]:38474 at 17:31:53.537993: ------------------------------------------------------------------------ REGISTER sip:190.64.204.60:5560 SIP/2.0 Via: SIP/2.0/UDP 10.45.210.97:39245;branch=z9hG4bK.FMbxN6bPS;rport From: <sip:anacelia@190.64.204.60>;tag=MBKS11QUc To: sip:anacelia@190.64.204.60 CSeq: 24 REGISTER Call-ID: DFvn85sO3E Max-Forwards: 70 Supported: replaces, outbound, gruu Accept: application/sdp Accept: text/plain Accept: application/vnd.gsma.rcs-ft-http+xml Contact: <sip:anacelia@10.45.210.97:39245 ;app-id=929724111839;pn-type=firebase;pn-tok=cofXHCv3YgA:APA91bFPHHgimCnRO8esdZvDgLWpitTLPMf1d7vn50EWLZEIeM95CBRbf3DHZrgLNv5dePml__0AYgd7jcE56-5dw-rWVOtEqiVKtBojE8c9kezaTvFFNohHrs1zdq8FAqgUxFBxaiEP;pn-silent=1;transport=udp>;+sip.instance="<urn:uuid:0707dcae-0314-40d0-90e2-79e6063cf202>";+org.linphone.specs=groupchat Expires: 3600 User-Agent: LinphoneAndroid/4.0.1 (belle-sip/1.6.3) Authorization: Digest realm="190.64.204.60", nonce="6fce67e0-b601-11e8-982b-cdce49bc5d42", algorithm=MD5, username="anacelia", uri="sip:190.64.204.60:5560", response="077e0b6c433066db828f9ec829f8d558", cnonce="~Oh2sj~-KZHHDh9I", nc=00000004, qop=auth ------------------------------------------------------------------------ send 765 bytes to udp/[186.51.236.123]:38474 at 17:31:53.540658: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.45.210.97:39245 ;branch=z9hG4bK.FMbxN6bPS;rport=38474;received=186.51.236.123 From: <sip:anacelia@190.64.204.60>;tag=MBKS11QUc To: <sip:anacelia@190.64.204.60>;tag=HDr7833tNa1gj Call-ID: DFvn85sO3E CSeq: 24 REGISTER Contact: <sip:anacelia@186.51.236.123:38474 ;app-id=929724111839;pn-type=firebase;pn-tok=cofXHCv3YgA:APA91bFPHHgimCnRO8esdZvDgLWpitTLPMf1d7vn50EWLZEIeM95CBRbf3DHZrgLNv5dePml__0AYgd7jcE56-5dw-rWVOtEqiVKtBojE8c9kezaTvFFNohHrs1zdq8FAqgUxFBxaiEP;pn-silent=1;transport=udp>;expires=30 Date: Tue, 11 Sep 2018 20:31:53 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.20~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------

3 2

Freeswitch 1.8.1 build without core pgsql support
by sukitha jayasinghe 12 Sep '18

12 Sep '18

Dear All, I installed freeswitch 1.8.1 in Debian 9 using apt commands. It seems like packages has build without pgsql support. When I run the application with pgsql configurations it gives bellow error. switch_core_sqldb.c:468 Failure! PGSQL NOT AVAILABLE! is there any special reason to remove this from prebuild packages. Is there anyway to enable pgsql support without rebuild. BR, Sukitha.

1 0

how to push data frames directly to FS from an endpoint
by fabio 12 Sep '18

12 Sep '18

Hi All I'm developing a simple FS endpoint. At the moment I have registered a 'read_frame' function to return some data to the FS core. For what I understand this function is called by the FS core. It works as a charm. Anyway I would like to push the data frames directly to the FS core without registering a 'read_frame' function as soon as the data are available from the lower layer. Is there any FS API for this purpose? I didn't find any example about this approach. I remember that this was possible in Asterisk. Where can I found any informations about the FS internals? Any help will be greatly appreciated. Best regards Fabio -- Fabio Antonini /Software Engineer (Ph.D)/ f.antonini(a)tiesse.com <mailto:f.antonini@tiesse.com> *Tel* +39.0863.455830 *Mob* +39.393.9261941 *Fax* +39.0863.455830 Via Corradini 80 67051 Avezzano (AQ) Logo Tiesse dal 1998 al 2018, vent'anni di Innovazione Made in Italy. Clicca per visitare il sito Tiesse <http://www.tiesse.com/> *Tiesse S.p.A.* - www.tiesse.com <http://www.tiesse.com/> Via Asti 4, 10015 Ivrea (TO) Pagina Tiesse su Linkedin, clicca e visitaci <https://linkedin.com/company/tiesse-spa> *Disclaimer:* il contenuto di questa email è riservato e non vincolante per Tiesse S.p.A.. Se lo avesse ricevuto per errore, la preghiamo di segnalarlo immediatamente al mittente, di non utilizzare e divulgare il contenuto e di distruggere ogni copia in suo possesso. Tiesse S.p.A. declina ogni responsabilità da qualsiasi conseguenza derivante da utilizzi non autorizzati, contraffazioni o manomissioni di email recanti riferimenti all'azienda. *Rispetta l'ambiente. Non stampare questa mail se non è necessario.*

2 1

DTLS renegotiation
by Jon Schøpzinsky 10 Sep '18

10 Sep '18

Hi List, We are currently investigating a problem, where our verto based softphone client re-connects, renegotiates ICE and then fails, as DTLS is never renegotiated. We registered a ticket, FS-11268, but there has been no response yet. Anybody got an idea of where we should look for a fix? Kind Regards Jon Schøpzinsky

1 0

How to use originate command call to number and then bridge to call center module
by Do Nguyen Ha 07 Sep '18

07 Sep '18

Hi i read on the freeswitch website about a syntax of the originate command there are a lot of examples show how to use it, but there is no example to make a new call(callA) and then bridge a call(callA) to call center module(callcenter application) example to make call and bridge a call to another number originate user/1000 &bridge(user/100) please help

3 8

Phone spontaneously going into Hold when a call is established?
by Nathan Neulinger 07 Sep '18

07 Sep '18

I had seen this symptom a few times randomly, and just got a couple of reports from users where a call to an outside number was going immediately to 'On Hold' when the person answered. Is there anything in the SIP/RTP stream that could be expected to trigger that sort of behavior? I've seen it from a polycom, and I believe one of the user reports today was with a yealink. User report was with a recently upgraded 1.6 -> 1.8 deployment. I believe the previous times I've seen it would have been on git builds of 1.8, but not certain. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

2 4

1.8.1 zip missing ./configure
by Mario 07 Sep '18

07 Sep '18

Just saw 1.8.1 show up in files.freeswitch.org <http://files.freeswitch.org/>. I tried to build from .zip but ./configure is missing, not sure if this is a bug or design change. INSTALL file still says use ./configure. Mario G

2 1

Re: [Freeswitch-users] mod_tts_commandline hangs in call flow
by Lee Vidor 07 Sep '18

07 Sep '18

It seems enabling the following command resolves my problem: <param name="threaded-system-exec" value="true"/> Can anyone explain in detail what this variable does? Thanks! On Mon, 3 Sep 2018 at 20:30, Lee Vidor via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Lee Vidor <leevidor89(a)googlemail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Fri, 31 Aug 2018 13:50:27 +0100 > Subject: mod_tts_commandline hangs in call flow > Hi > > I'm currently using mod_tts_commandline with a python script to AWS text > to speech API. The flow works fine during a single call. > > Event socket API -> Freeswitch dialplan -> mod_tts_commandline -> Python > script -> AWS API -> Voice file -> Freeswitch say > > However when there a number of concurrent calls (even just 10 to 15) the > request to execute the python script hangs until all the calls are at the > same requesting state. As if mod_tts_commandline is waiting for something > to happen. > > Dial plan: > > <extension name="tts_google"> > <condition field="destination_number" expression="^11111$"> > <action application="set" data="call_timeout=60" /> > <action application="answer"/> > <action application="detect_silence" data="100 30 25000"/> > <action application="speak" > data="tts_commandline|${lang}|${text}" /> > <action application="hangup"/> > </condition> > </extension> > > tts_commandline.conf: > > <param name="command" value="/usr/bin/python > /usr/local/freeswitch/scripts/synthesize_text.py --language=${voice} > --filename=${file} --text=${text}"/> > </settings> > > Execution of python script in freeswitch logs: > > 2018-08-30 11:19:57.641892 [ERR] switch_ivr_play_say.c:2385 > switch_ivr_detect_silence: SILENCE DETECTED > 2018-08-30 11:19:57.641892 [DEBUG] switch_core_codec.c:248 > sofia/external/33123456789 Restore previous codec PCMU:0. > EXECUTE sofia/external/33123456789 speak(tts_commandline|en-US|Hi, This is > a test message) > 2018-08-30 11:19:57.641892 [DEBUG] switch_ivr_play_say.c:3025 OPEN TTS > tts_commandline > 2018-08-30 11:19:57.641892 [DEBUG] switch_ivr_play_say.c:3035 Raw Codec > Activated > 2018-08-30 11:19:57.641892 [DEBUG] mod_tts_commandline.c:160 Executing: > /usr/bin/python /usr/local/freeswitch/scripts/synthesize_text.py > --language='en-US' > --filename='/tmp/92ec5996-0dc8-4293-b5ba-465430811fed.tmp.wav' --text='Hi, > This is a test message' > > "Simultaneous calls requesting mod_tts_commandline" > > Log from python script: (notice it's called with a 15 second delay) ---> *What > is causing this delay?* > > 2018-08-30 11:20:12,488 DEBUG Called script > 2018-08-30 11:20:12,490 DEBUG Arguments > Namespace(filename='/tmp/92ec5996-0dc8-4293-b5ba-465430811fed.tmp.wav', > language='en-US', ssml=None, text='Hi, This is a test message') > 2018-08-30 11:20:12,490 DEBUG synthesize_text requested > > The python script executes fine and returns to freeswitch: > > 2018-08-30 11:20:14.541919 [DEBUG] switch_core_file.c:342 File > /tmp/c61d2034-c1b1-4f42-932f-350484553b5a.tmp.wav sample rate 24000 doesn't > match requested rate 8000 > 2018-08-30 11:20:14.541919 [DEBUG] switch_ivr_play_say.c:2729 Speaking > text: Hi, This is a test message > > Any help or further debugging steps would be appreciated. > > Thanks > > Lee > > > > ---------- Forwarded message ---------- > From: Lee Vidor via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 03 Sep 2018 12:30:09 -0700 (PDT) > Subject: [Freeswitch-users] mod_tts_commandline hangs in call flow > _________________________________________________________________________ > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com

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Show ring-group name on phone
by René Weiss 07 Sep '18

07 Sep '18

Hi What is the best/right way to show the name of the called ring-group on the ringing phone? At the moment I have configured 2 ring groups (let's call them "main number" and "hotline") which bridge incoming calls like this: <action application="bridge" data="user/101@$${domain_name}, user/102@$${domain_name}, user/103@$${domain_name} "/> The problem is now, that on the display of the ringing phones I can only see that there's a call from <caller-number> to the number of the individual phone. What I would like to see is: call from <caller-number> to <ring-group number> (or <ring-group-name>) (We use Yealink phones (mostly T46s) if that helps) Thanks.

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