similar to: Asterisk Fax

Hi all, I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine via a T.38 enabled trunk.? I've got t38pt_udptl = yes faxdetect=no in my sip.conf file.? The ATA has all of the T.38 options turned on, echo cancellation is off, as well as silence suppression off.? The only configured codec is u711.? When the user tries to send a fax, it gets to the point where it

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

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

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

Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten => 306,1,NoOp(Fax transmission) same => n,Set(FAXOPT(gateway)=yes) same => n,Dial(DAHDI/3) ----->FXS port to fax machine same => n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -->

In the simplest terms I can think of, I'm going to describe what I want to do and I want to know if it's possible in the current version of asterisk. Can I take a T38 call from an ATA, convert that back to analog and have asterisk screech that out on a POTS line to a remote fax machine. Would it work? And could I receive an incoming fax the same way? Please don't talk to me

Hi Guys, This is such an annoying issue whenever it comes up. The sender and receive always receive the source public IP no matter what in the IP packets but then SIP packets go out with something like 192.168.0.20. In this instance, a Bell Canada DSL modem is installed and I saw the SPA-2102 register properly but only to fail later on due to sending it's LAN IP to the Asterisk server. So,

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

Howdy All, I'm considering Asterisk / Digium as a replacement to my existing phone switch. I need to continue to be able to push analog lines between multiple buildings in a campus environment. The Digium Analog 410 Card manual states it's not recommended to go beyond 1500 feet distance for an FXS card, and no line should leave the building or be bundled. The 2400 Series Manual does

I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of "softfax" that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling

Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000 at mydomain to call 9000 at my domain(which is the call center)

Is it possible to do SIP<->Asterisk<->TDM in a single step with FFA? Or does FFA always use TIFF files? I'm using Free FFA on 1.6.2.15 and I want to be able to use SPA 2102 ATA's at the fax machines and send faxes directly over a PRI. Mark -- Mark Willis Star One Telecom Office: 1-800-889-7001 Cell: 210 880 5097 http://staronetel.com

Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root at ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk

Hello I am going to try the new Digium Fax for Asterisk product. I'm planning to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs. I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has any experience with these devices, or other recommendations, I would be grateful if you could share your experiences. Regards Ian

Hello, Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. P.S. Are both options part of the configs of Asterisk or need modules to be selected and installed before doing the

Anyone have any testing experience with T38 and HT-502 Grandstream? I just want to confirm that t.38 is working on this device. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110801/fea18d8c/attachment.htm>

Anyone knows what should be the configuration of the new solution of Digium for fax in order to send and receive faxes from PSTN to a fax machine through an ATA implementing T38 protocol?

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.

Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by

Hi list. I have Asterisk installed on a Debian 1.8 6 64-bit. What happens is the following, some channels are not being hangup properly. They run the hangup in dialplan, but the output of the command "core show channels" shows several channels with status "rsrvd." Checking the server's memory, the "top" command shows multiple processes and stopped using the