search for: t38_udptl

Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : "Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP provider account as well as the fax account. " But above, you can read "[general] t38pt_udptl = yes " Has this parameter name changed between 1.4 to 1.6 from t38_udptl to t38pt_udptl ? A...

...--- 3800 & 3801 When I place a call from 3800 to 3700 or the other way around , asterisk seg faults on both machines at roughly the same time. All connections are done using PJSIP. The crash occurs when the ringing extension is answered. If I set (directmedia=no) OR (directmedia=yes & t38_udptl=yes) on the trunk then the call completes fine. All phones and servers are on the same LAN with no firewalls active. The trunk between AST-A and AST-B is configured like this in pjsip.conf and is identical on both machines: [transport-lan] type=transport protocol=udp bind=0.0.0.0 tos=af31 [pb...

...=====TRANSPORT========================= [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_patt...

Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk <=> B A: INVITE + Audio SDP => Asteris...

...media feature 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

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

...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. 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...

...w = !all,ulaw direct_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>

...very disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes ;qualify=yes [caller] type=friend secret=123456 host=dynamic callerid="caller <12129887777>" context=out nat=yes dtmfmode=rfc2833 canreinvite=yes insecure=no disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm t38_udptl=yes qualify=yes I have registered [caller] on xlite at client system and dialing following context in local system that will dial [abc] [out] exten=> _X.,1,Dial(SIP/${EXTEN}@abc,30,1) exten=> _X.,n,Hangup as you can see above *highlighted that context of abc is payasyougo.*problem is tha...

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)

...04]: udptl.c: 852 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>

...faxing, and the endpoint responds with a declined media stream a crash will then occur in Asterisk. Modules Affected res_pjsip_t38.c Resolution If T.38 faxing is not required then setting the “t38_udptl” configuration option on the endpoint to “no” disables this functionality. This option defaults to “no” so you have to have explicitly set it “yes” to potentially be affected by this issue....

...faxing and has a port of 0 and no c line in the SDP, a crash will occur. Modules Affected res_pjsip_t38.c Resolution If T.38 faxing is not needed, then the “t38_udptl” configuration option in pjsip.conf can be set to “no” to disable the functionality. This option automatically defaults to “no” and would have to be manually turned on to experience this crash....

Greetings, I have a fresh asterisk installation. When I install I get all of the config files. What is the best way to get a 'stripped' down system with just the bare config files I would need to do a sip connection? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/4e1042b1/attachment.htm

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

...... packets) 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_retr...

...k (11.18.0 t0gw) 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, -- >...

...v4] type=transport 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 1...

...this issue | > | | is 'moderate'. | > | | | > | | Users who are using the default configuration with | > | | 't38_udptl' set to 'no' or an equivalent value are not | > | | susceptible to this vulnerability. Users who have set | > | | this configuration item to 'yes' or an equivalent value | > | | but are not actually using T.38 support c...