search for: t38_udptl_ec

...========================= [simpletrans] type=transport protocol=udp bind=0.0.0.0:5060 ;===============TRUNK============================== [trunk-provider] type=endpoint transport=simpletrans context=in_provider direct_media=no disallow=all allow=alaw allow=g729 aors=trunk-provider t38_udptl=yes t38_udptl_ec=redundancy t38_udptl_maxdatagram=400 [trunk-provider] type=aor contact=sip:X.X.X.X:5060 [trunk-provider] type=identify endpoint=trunk-provider match=X.X.X.X match=X.X.X.X [trunk-patton] type=aor max_contacts=5 [trunk-patton] type=endpoint transport=simpletrans context=in_patton direct_media=n...

...s 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

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

...eiving 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. 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...

...rect_media = no from_domain = us-west-wa.sip.flowroute.com tos_audio = ef tos_video = af41 ; Note: "force_avp = yes" fixes issues with calls coming from MagicJack with no audio after a few seconds. force_avp = yes auth = flowroute outbound_auth = flowroute aors = flowroute t38_udptl = yes t38_udptl_ec = fec [anonymous] type=endpoint context = anonymous allow = !all,ulaw -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200601/f775eb62/attachment.html>

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)

...2 calculate_far_max_ifp: UDPTL (no tag): Can not calculate far_max_ifp before far_max_datagram has been set After the warning begins exchanging packets UDPTL, but all with sequence number 0 or 256. The fax is not properly transmitted or received. I configured the trunk as follow: t38_udptl = yes t38_udptl_ec = redundancy t38_udptl_maxdatagram = 400 How can I fix the problem? Thank You Marco -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150204/2b939377/attachment.html>

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:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a ?crit : > I think the "488 Not acceptable here" is occurring because the channel > connecting through is not T.38 capable, that will be the IAX channel > from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more

...It should be a typical scenario, 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 expiratio...

...connected to the PSTN via ISDN; the call is to my test fax machine, connected 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 = (...

...rt protocol=tcp bind=10.75.22.8:5060 I've then configured an endpoint to use it: [outgoing] type = endpoint context = default dtmf_mode = none disallow = all allow = all rtp_symmetric = yes force_rport = yes rewrite_contact = yes direct_media = no language = en aors = outgoing t38_udptl = yes t38_udptl_ec = none ice_support=yes transport=transport-tcp-ipv6 But this seems to result in SDP payloads with ICE not offering any IPv4 candidates: v=0 o=- 508048280 508048280 IN IP6 2001:1234:5678:abcd::2 s=Asterisk c=IN IP6 2001:1234:5678:abcd::2 t=0 0 m=audio 14384 RTP/AVP 4 0 8 3 111 112 5 10 118 7 18 11...

...amed_pickup_group= record_on_feature=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...