similar to: T38 re-invites issue

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

Hello list, I read on voip-info.org that Asterisk 1.4 support T38 passthrough. So I guess this means that I can have a Grandstream HT503 with T38 support and an analogue faxmachine on the other side of my Asterisk and a T38-account with a ITSP on the other side of my Asterisk machine, right ?! The fax coming from the faxmachine passes the HT503 to my Asterisk and my Asterisk sends the fax to

Add more vDPA callbacks. [s/g]et_vq_state is added in patch 1, including a new control mechanism to read data from the DPU. suspend is added in patch 2. Alvaro Karsz (2): vdpa/snet: support getting and setting VQ state vdpa/snet: support the suspend vDPA callback drivers/vdpa/solidrun/Makefile | 1 + drivers/vdpa/solidrun/snet_ctrl.c | 324 +++++++++++++++++++++++++++++

Add more vDPA callbacks. [s/g]et_vq_state is added in patch 1, including a new control mechanism to read data from the DPU. suspend is added in patch 2. Alvaro Karsz (2): vdpa/snet: support getting and setting VQ state vdpa/snet: support the suspend vDPA callback drivers/vdpa/solidrun/Makefile | 1 + drivers/vdpa/solidrun/snet_ctrl.c | 324 +++++++++++++++++++++++++++++

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 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

The callback sends a resume command to the DPU through the control mechanism. Signed-off-by: Alvaro Karsz <alvaro.karsz at solid-run.com> --- drivers/vdpa/solidrun/snet_ctrl.c | 6 ++++++ drivers/vdpa/solidrun/snet_main.c | 15 +++++++++++++++ drivers/vdpa/solidrun/snet_vdpa.h | 1 + 3 files changed, 22 insertions(+) diff --git a/drivers/vdpa/solidrun/snet_ctrl.c

Dear Mr. Cardon! Na štvrtok, 9. novembra 2017 18:29:03 CET Denis Cardon via samba napísali: > > > IO would like to ask for help with diagnose why my Samba version > > 4.7.0-git. > > 23.4e3f0fb9d15SUSE-oS13.3-x86_64 in openSUSE Tumbleweed can not > > authentificate me on Windows server 2003 domain > > > > in /etc/fstab I have working combination - smb

Hi I'm trying to get the spa 8000 used with a fax machine using t38 passthru i have tried with 1.6.2 and 1.8.3 and is still a no go the provider i use is 012 in israel wich supports t38 (i use it with ffa) could anybody give me a clue how to get this working if it should t38pt is set to yes in sip.conf Thanks, Israel -------------- next part -------------- An HTML attachment was

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

Hi everyone, I have been trying to get T.38 Faxing to work with Iristel sip trunks for last few days but havn't been sccussful. I am using Asterisk 1.6.2.8 and SpanDSP 0.6. Here is what I see in the tcpdump capture: 1. Call come in from the trunk as regular voice call with g.711 codec 2. Asterisk answers the call and recognizes the CNG and sends the call to fax extension 3. Eventually

Would you mind repeating the test with canreinvite=no set for all you phones and mobile phones? What is your upload bitrate? Is it guaranteed? I would try also to test the PMTU: Try: ping -M do -s 2000 ${ip address of the sip server} You should receive icmp asking for lowering the packet size. The LTE phones could have lower MTU and thus overcome PMTU problem. Marek 2020-06-22 21:48

Hello, I found the problem and also the workaround. Clearly, since it was working with chan_sip it should not be dialplan problem, but sip stack problem. I have line=yes set up. After asterisk restart the old registration is not unregistered and new one is registered with different line value. Then incoming invites and qualify requests are sent to all the registrations and there the problem

Hi, I'm a student doing my masters thesis and trying to work with S-Net version 1.0 (tool used for statistical analysis and visualizations of internet traffic) that is written in R. The problem is this: I have a perl script calling a function defined in an R script that is located in a directly different from where the perl script is. This gives the following error. >>> echo

Hi Jason, > > + /* Overwrite the control register with the new buffer size (in 4B words) */ > > + snet_write_ctrl(regs, buf_words); > > + /* Use a memory barrier, this must be written before the opcode register. */ > > + wmb(); > > > At least you need to use smp_wmb() but if you want to serialize MMIO > writes you can simply use

Hello, I have a issue between asterisk and windows based VoIP system (Client). Vendor SIP Server --> My asterisk --> Client Here is ethereal trace between asterisk and client. 1 0.000000 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23 <sip%3A1978525648 at 192.168.4.23>, with session description 2 0.042380 192.168.4.23 -> 192.168.3.222

Hello, after migration from chan_sip to res_pjsip I get strange behavior when receiving call from the outside world. When call is received, it is replicated multiple times. Two of that calls get to the phone. So the phone is ringing on both lines. When having only Dial function in dialplan I am able to place call. But when creating some dialplan procedures containing VoiceMail I get phone ringing

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last

Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 -> SIP ATA (T38 enabled) -> fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys

Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James