similar to: Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.

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)

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

> Asterisk Project Security Advisory - ASA-2007-010 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Two stack buffer overflows in SIP

> Asterisk Project Security Advisory - ASA-2007-010 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Two stack buffer overflows in SIP

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.

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:

Dear List I fear I stumbled over a bug in asterisk 13.14.1. My 'phones' are roaming around, sometimes some are connecting from ipv6 enabled networks, another time they are not. If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat problems. I have not specified a transport in the endpoint section, so that the appropriate transport which corresponds to the registration

Hi I'm attempting to use ICE to be able to present all possible RTP transports to peers. 16.28.0~dfsg-0+deb11u2 (I know it's old, but unfortunately Asterisk was removed from debian 'stable' and the version in 'sid' is just broken (opus + voicemail don't work anymore). But I ran into an issue when the peer is running rtpengine: Asterisk offers: a=candidate:H9da13901

Asterisk Project Security Advisory - Product Asterisk Summary Re-invite with T.38 and malformed SDP causes crash. Nature of Advisory Remote Crash Susceptibility Remote Authenticated Sessions Severity Minor

Hello, I need help to solve a problem that I am having using Asterisk 13, PJSIP and T38. My setup is as follows: SIP Provider --> Asterisk 13 --> Patton --> Physical Fax I need to get the fax directly in T38 to Patton. The provider sends me the fax in T38. If I receive the T38 fax on Asterisk (using an hylafax device), I can properly receive the fax. If I send a T38 fax with Asterisk

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple

Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP

Hello, I use Asterisk 13.1 with a SIP trunk with a provider When transmitting or receiving a fax in T.38 through the trunk with the provider I always get this warning: WARNING [8204]: 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

Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called

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:

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, 2 weks ago I asked questions about PJSIP and T.38 but got no replies. I upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having the same issues. In the trace below, I'm sending a fax from Hylafax server through iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw) connected to the PSTN via ISDN; the

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

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.

Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over 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