search for: udptl

Dear all, Anyone get the lastest spandsp with udptl.c and tpkt.c compile in Fedora 3? tpkt.c: In function `accept_thread': tpkt.c:140: error: `TCP_NODELAY' undeclared (first use in this function) tpkt.c:140: error: (Each undeclared identifier is reported only once tpkt.c:140: error: for each function it appears in.) tpkt.c:144: error...

Hello guys. When I am trying to send fax through T38 to linksys SPA (properly configured etc. - I have tried it with other systems), I'm getting error and fax is not delivered. I'm getting this errors in asterisk.log: WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c: UDPTL creation failed WARNING[687] udptl.c: No UDPTL ports remaining then, couple lines down: WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image 16400 udptl t38 WARNING[3514] chan_sip.c: Failing due to no acceptable offer found...

Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-----[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:...

On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: <snip> > > t38modem and asterisk are using > > m=image 35622 udptl t38 > ^^^^^ > > Provider uses > > m=image 35622 UDPTL t38 > ^^^^^ > > Could this be a problem? If I'm sending internal only, it's always > lowercase. Looking at the tests we have we only use 'udptl' as the transport. With...

Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation. What can I do to prevent this from happening? TIA,...

Hello, We're trying to receive faxes on the Asterisk server, but for the time being T.38 negotiation fails. The SDP that the Asterisk reINVITE sends contains these lines: ---------------------- m=image 4968 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy ---------------------- The MaxDatagram and MaxBitRate are definitely not what they should b...

...see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl = yes,fec,maxdatagram=400 faxdetect = t38 And in udptl.conf, I just uncommented 'use_even_ports = yes ;' and rest of it set as default. Here is the error I'm usually seeing in Asterisk side; [Jan 15 14:13:28] NOTICE[20514] udptl.c: UDPTL (SIP/10.3.22.6-00000ad6): Transmission error...

...t" <447>") in new stack -- Executing [170 at internal:4] Dial("DAHDI/1-1", "SIP/office-home-sip/170") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 4 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Called office-home-sip/170 On the office asterisk: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Executing [170 at default:1] Mac...

...97673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38 on 0197673581, changing t38pt_udptl=no to t38pt_udptl=yes,fec and try to send from 0197673581 to 0851711201, it is not working, switches to T38 sendimg a lot of UDPTL packages but it looks like (at least for me) that addresses are "wrong". UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPT...

...the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963 at gafachi1a for application SendFAX(/srv/httpd/htdocs/upload/scantest2.tiff,dz) (Retry 1) == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 > Channel SIP/gafachi1a-0000000a was answered. > Launching SendFAX(/srv/httpd/htdocs/upload/scantest2.tiff,dz) on SIP/gafachi1a-0000000a -- Channel 'SIP/gafachi1a-0000000a' sending FAX: -- /srv/httpd/htdocs/upload...

Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo

...l:7] PlayTones("Zap/4-1", "ring") in new stack -- Executing [s at incoming-dial:8] Dial("Zap/4-1", "SIP/100&SIP/101&SIP/102&SIP/107&SIP/111,20,tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Called 100 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Called 101 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using UDPTL TOS bits 184...

...A's and Asterisk remain the same, just switched the faxes to the oposite end of the path. Doing this comfirmed the original results, sharp to cannon is reliable, cannon to sharp is unreliable. What I observe faxing from sharp to cannon: path sets up as ulaw RTP, cannon answers, RTP switched to UDPTL, fax completes Faxing from cannon to sharp: path sets up as ulaw RTP, sharp answers, RTP switches to UDPTL only half the time when UDPTL is active, fax completes, when the path stays with RTP, fax always fails If I understand correctly, Asterisk switches the media stream to UDPTL when it hears va...

Asterisk Project Security Advisory - AST-2016-003 Product Asterisk Summary Remote crash vulnerability when receiving UDPTL FAX data. Nature of Advisory Denial of Service Susceptibility Remote Authenticated Sessions Severity Minor...

On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote: > Has anybody any idea why asterisk drops the media stream in the 200 OK? > The channel has been T38_ENABLED before! Or is it necessary to add more > debug code? Who does the negotiating? > Only asterisk or is pjsip doing some parts, too? Asterisk does the T.38 negotiation and produces the answer SDP, PJSIP does the SDP

Hello! unchanged asterisk crashes during udptl / t.38 negotiation with telekom - they do not support t.38 / udptl. In detail: fax client -> asterisk -> telekom -> easybell -> asterisk -> fax server Fax server sends t.38 reinvite via asterisk to easybell. Session Description Protocol Version (v): 0 Owner/Creator, Sessio...

...ection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Executing [3008384e0 at sipgate-test:1] Answer("SIP/sipgate-00000016", "") in new stack -- Executing [3008384e0 at sipgate-test:2] Goto("SIP/sipgate-00000016", "home,447,1") in new stack -- Goto (h...

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...

...he provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens? * The fax from SPA112 to Asterisk cmd ReceiveFax works well, * The fax from Asterisk cmd SendFax to PSTN fax works well, * However, the fax from SPA112 to PSTN fax doesn't work. Using udptl debug, I can see packets between Asterisk and both sides (SPA112 and PSTN fax) but it seems that faxes can't agree how to send image. == sip.conf: [general] tcpenable=yes videosupport=yes transport=udp,tcp dtmfmode=rfc2833 qualify=yes directmedia=no allowguest=no alwaysauthreject=yes rtcache...

All; I'm running Asterisk 1.8.15-cert3 with the newest version of spandsp. I've even tried unloading that and using Digium's FFA module but I receive the same error on an outbound transmission: [2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL (SIP/XXXXXXXXXXX_outbound-00000000): Buffer overflow detected (59 + 127 > 175) I only get this with one specific upstream provider. Has anyone seen this before? Any help at all would be greatly appreciated. Regards; John -------------- next part -------...