search for: t38_udptl_nat

...type=endpoint disallow=all allow=g722 allow=ulaw transport=transport-lan context=phone-level3 aors=pbxbeta send_rpid=no send_pai=yes trust_id_inbound=yes trust_id_outbound=yes direct_media=yes direct_media_glare_mitigation=outgoing ;direct_media_method=update tos_audio=46 tos_video=34 t38_udptl=no t38_udptl_nat=no [pbxbeta] type=aor contact=sip:{remote IP address}:5060 [pbxbeta] type=identify endpoint=pbxbeta match={remote IP address} The phones have the following set in pjsip.conf (snippet): type=endpoint disallow=all allow=g722 allow=ulaw transport=transport-lan send_rpid=no send_pai=yes direct_medi...

Hello! I'm still trying to get a working t.38 configuration w/ pjsip. I'm now able to send t.38 faxes to my own extension: hylafax -> t38modem -> extension -> extension -> t38modem -> hylafax. The fax is sent by t38modem. The receiving part of t38modem accepts the call, sends ReInvite for t.38 and things are working as expected. Now, let's do the nearly same

...#39;t know why it isn't sent to the ISP. The extension is a normal pjsip extension with these additional options: t38_udptl : true t38_udptl_ec : redundancy t38_udptl_ipv6 : false t38_udptl_maxdatagram : 400 t38_udptl_nat : no (or yes - doesn't matter) The trunk looks exactly the same: t38_udptl : true t38_udptl_ec : redundancy t38_udptl_ipv6 : false t38_udptl_maxdatagram : 400 t38_udptl_nat...

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)

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

...tag sip.fax in the contact header. Did I miss some configuration? That's my setup: Hylafax sends fax to t38modem and t38modem is connected via SIP to asterisk as extension. The extension is bound to an outbound route, which uses the t.38 capable ISP. pjsip.endpoint.conf: [ISP] t38_udptl=yes t38_udptl_nat=no # there is no nat necessary t38_udptl_ec=fec [t38endpoint] t38_udptl=yes t38_udptl_ec=fec Any idea? Thanks, Michael

...o, but it does not work... Is there any way to make it working ? -------------- next part -------------- [0.0.0.0-udp] type=transport protocol=udp bind=0.0.0.0:5060 [endpoint0](!) type=endpoint transport=0.0.0.0-udp disallow=all allow=alaw allow=ulaw t38_udptl=no t38_udptl_ec=none fax_detect=no t38_udptl_nat=no dtmf_mode=auto direct_media=yes from_domain=172.16.25.23 timers_sess_expires=1800 tone_zone=ru language=ru rewrite_contact=yes rtp_symmetric=yes force_rport=yes [registration0](!) type=registration transport=0.0.0.0-udp retry_interval=60 max_retries=10 expiration=3600 auth_rejection_permanent=y...

...nnected to the PSTN. chan_pjsip is used on Asterisk-13, and chan_sip is used on Asterisk-11. This how endpoint t0gw (Asterisk-11) is configured on tiare (Asterisk-13 ): tiare*CLI> pjsip show endpoint t0gw ... t38_udptl : true t38_udptl_ec : fec t38_udptl_ipv6 : false t38_udptl_maxdatagram : 400 t38_udptl_nat : false ... Could someone explain why I'm getting "Not acceptable" below? -- Accepting AUTHENTICATED call from 127.0.0.1:4570: -- > requested format = slin, -- > requested prefs = (), -- > actual format = slin, -- > host pre...

...e=automixmon dtls_private_key= named_call_group= t38_udptl_maxdatagram=0 media_encryption_optimistic=false aors=DEADDEADBEEF rpid_immediate=false outbound_proxy= identify_by=username inband_progress=false rtp_symmetric=false transport=transport-udp rtp_keepalive=0 t38_udptl_ec=none fax_detect=false t38_udptl_nat=false allow_transfer=true tos_video=0 srtp_tag_32=false timers_min_se=90 call_group= sub_min_expiry=0 100rel=yes direct_media=true rtp_timeout_hold=0 g726_non_standard=false dtmf_mode=rfc4733 voicemail_extension= rtp_timeout=0 dtls_cert_file= media_encryption=no media_use_received_transport=false d...