On 06/05/2017 at 05:00 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: >> On 06/05/2017 at 11:30 AM, Joshua Colp wrote: >>> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >>>> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>>>> Just a guess (without knowing about your network), but are the two ends >>>>> points on public networks and visible to one another? If not the reinvite >>>>> may be passing an internal (nat'ed) address to the other and the >>>>> connection >>>>> will fail...just a though >>>> >>>> t38modem -tt -o /var/log/t38modem.log --no-h323 -u 91 --sip-listen >>>> udp\$127.0.0.1:6060 --ptty +/dev/ttyT380,+/dev/ttyT381 --route >>>> 'modem:.*=sip:<dn>@127.0.0.1:5061' --route 'sip:.*=modem:<dn>' >>>> --sip-register 91@127.0.0.1:5061,password >>>> >>>> I tried it with a global IP (instead of 127.0.0.1) - same behavior. >>>> >>>> The point is, that the receiving part, which initiates the t.38 switch, >>>> doesn't sent the switch to the ISP. It is blocked / ignored by asterisk >>>> at all - don't know why it isn't sent to the ISP. >>> >>> I'd suggest providing the console output and SIP traffic (pjsip set >>> logger on) so we can see exactly what is going on. >>> >> >> I attached the debug output I already created before. >> >> Interesting part starts around line 2740. >> >> >> 91 -> local pjsip fax-extension >> >> 127.0.0.1:5061 -> asterisk server local connect for fax-extension (-> >> not encrypted even if it is port 5061!) >> >> external fax number at easybell (195.185.37.60), which is called and >> which is answered here: 11111222222 > > And the pjsip.conf endpoint entry for easybellPJSIP_FAX? >
[easybellPJSIP_FAX] type=endpoint transport=0.0.0.0-udp context=from-trunk disallow=all allow=alaw,ulaw aors=easybellPJSIP_FAX language=de outbound_auth=easybellPJSIP_FAX from_domain=sip.easybell.de from_user=+4911112222222 t38_udptl=yes t38_udptl_ec=redundancy fax_detect=no t38_udptl_nat=no send_rpid=yes send_pai=yes dtmf_mode=rfc4733 tos_audio=0xb8 direct_media=yes rewrite_contact=no force_rport=yes ParameterName : ParameterValue ========================================================= 100rel : yes accountcode : acl : aggregate_mwi : true allow : (alaw|ulaw) allow_overlap : true allow_subscribe : true allow_transfer : true aors : easybellPJSIP_FAX asymmetric_rtp_codec : false auth : bind_rtp_to_media_address : false call_group : callerid : <unknown> callerid_privacy : allowed_not_screened callerid_tag : connected_line_method : invite contact_acl : context : from-trunk cos_audio : 0 cos_video : 0 device_state_busy_at : 0 direct_media : true direct_media_glare_mitigation : none direct_media_method : invite disable_direct_media_on_nat : false dtls_ca_file : dtls_ca_path : dtls_cert_file : dtls_cipher : dtls_fingerprint : SHA-256 dtls_private_key : dtls_rekey : 0 dtls_setup : active dtls_verify : No dtmf_mode : rfc4733 fax_detect : false fax_detect_timeout : 0 force_avp : false force_rport : true from_domain : sip.easybell.de from_user : +4911112222222 g726_non_standard : false ice_support : false identify_by : username inband_progress : false language : de mailboxes : media_address : media_encryption : no media_encryption_optimistic : false media_use_received_transport : false message_context : moh_suggest : default mwi_from_user : mwi_subscribe_replaces_unsolicited : false named_call_group : named_pickup_group : one_touch_recording : false outbound_auth : easybellPJSIP_FAX outbound_proxy : pickup_group : record_off_feature : automixmon record_on_feature : automixmon rewrite_contact : false rpid_immediate : false rtcp_mux : false rtp_engine : asterisk rtp_ipv6 : false rtp_keepalive : 0 rtp_symmetric : false rtp_timeout : 0 rtp_timeout_hold : 0 sdp_owner : - sdp_session : Asterisk send_diversion : true send_pai : true send_rpid : true set_var : srtp_tag_32 : false sub_min_expiry : 0 subscribe_context : t38_udptl : true t38_udptl_ec : redundancy t38_udptl_ipv6 : false t38_udptl_maxdatagram : 0 t38_udptl_nat : false timers : yes timers_min_se : 90 timers_sess_expires : 1800 tone_zone : tos_audio : 184 tos_video : 0 transport : 0.0.0.0-udp trust_id_inbound : false trust_id_outbound : false use_avpf : false use_ptime : false user_eq_phone : false voicemail_extension : Thanks, Michael -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users