similar to: T.38 Faxing and Asterisk

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

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

Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and

Hi All, I'm in the lab with Asterisk 1.6.1.12 and several ATA's testing T38. I hit a snag with the Grandstream HT502. It only seems to nail up a session at 9600bps. The Grandstream GXW4104 nails up consistently at 14400bps. I'm using the same equipment in the same configuration, just switching out the ATA. I have the latest firmware on each unit. Any ideas on what could cause

Hi list, I'm evaluating a private telephony scenario of about 20 locations - 300 phones, 50 FAX machines. Initial overview points to the installation of asterisk at three locations connected to the PSTN via ISDN PRI. All other locations, small by themselves, would get SIP phones managed by asterisk, since there is good IP connectivity between all sites. Now on to the

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Fax over T38 is failing, on the same system it worked with Callweaver. What do I need to post to be get further assistance please?

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 Asterisk users, We are having MANY but MANY problems configuring an analog fax machine to work properly on Asterisk, the first thing we do was to plug in the Fax analog machine to the FXS port of the Digium TDM410P, we set echocancel=no on zapata.conf and also faxdetect=yes on general section, but our Asterisk CRASH every time we try to send/receive fax! We are using Asterisk 1.4.21 and

Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081127/b41ca08b/attachment.htm

Hi all, I am running centos 5.1 and I wish to change the domain name and dnsdomainname of my PC. currently the settings are-- $ hostname sipx.com $ hostname --fqdn sipx.com $ domainname (none) $ dnsdomainname com I have searched in the net for tips but everywhere only the hostname change is provided. I need to change/set the domain name and the dnsdomain name on my pc to sipx.com and this

Hi, I have been trying to get T.38 to work with clients behind NAT for the past week but with no success. I have an asterisk server on the public internet and several Grandstream (I tried Linksys too) HT502 ATAs behind NAT in different locations. I tried every possible combination of NAT, canreinvite, t38pt_usertpsource entries, I even tried asterisk 1.4.19, 1.4.24.1, 1.4.25 all with the same

It seems that Spandsp has everything in it (when you include rxfax and txfax) to be a T.38 termination when used with Asterisk 1.4? And if so, what version of Spandsp? What version of IAXModem (so I don't have to also deal with T38Modem)?

Hi All, I have a telephony project for which I need to build a prototype to demo for management. The prototype must work on a GSM phone network. In the demo system, a call from GSM phone comes into the demo box. The demo box runs CallWeaver. Callweaver picks up the GSM call, answers it and plays a sould file, then dials out to a second GSM phone somewhere and connects them so they talk. My

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

Greetings- Working with the T.38 gateway functionality that is sparsely documented [1] , I'm attempting to get the following functional: Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box in question) -> SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from

Hi Vardan I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same. as you said there might be mistake in sip.conf so i am pasting both servers configuration here.. 1- nasir.server.com [abc] username=abc type=friend secret=mysecret nat=yes mailbox=12234568 incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=payasyougo

1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545

Hi I am trying to your wine with sipXecs. We are testing Cisco ATA-186. When we attempt to update the ATA software via sipX we send a text file to the sipX server but it needs to be convert from a text file into a binary via the cfgfmt.exe tool I installed sipX and wine on the same box and used wine config tool to point or grab the tool. Wine puts it or a copy of it in the system32 folder.