similar to: t38 ATA

Hello List, I think I've discovered a little bug in t.38 bug in 1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes. Asterisk always responds with a=T38MaxBitRate:2400. I've tried with Patton and Grandstream devices and the result is always the same. Patton ignores the parameter and sends the fax at 9600.

Dears I installed digium fax and followed the instruction at http://downloads.digium.com/pub/telephony/fax/README,And as you can see above that t38 is loaded I am using a call file to send fax1.tif file as fax to the gateway named add The problem that Addpac send always Receive 488 Not acceptable here,and lkindly find my debug attached Please advice. Thanks I Advance shark*CLI>

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

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>

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes extensions.conf [default] exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call) exten =>

Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can anyone imagine a scenario when enabling this parameter (even for peers that don't require it) can cause problems? Regards and thanks in advance, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference

> Zoa wrote: >> On friday we finally released Attrafax under a GPL2 license. >> It comes with its own set of modems and built in transparent gatewaying. >> The solution should be quite stable as long as the line quality is ok. >> (Some tools for measuring the line quality are included in the release, >> as well as some fax2mail scripts). >> >> There is

Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there ..... ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi There. I have the following setup : Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24 My problem is as follows : If I set up a very simple extensions.conf. when I dial from a fax machine, it seems as if no fax is being recognised. If I answer the call, I can hear the fax machine beeping. extensions.conf :

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

Receiving a fax with 1.6.1: == Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on 'DAHDI/4-1' -- Executing [fax at incoming-pstn-line:1] NoOp("DAHDI/4-1", "Fax Detected") in new stack -- Executing [fax at incoming-pstn-line:2] Goto("DAHDI/4-1", "incoming-fax,s,1") in new stack -- Goto (incoming-fax,s,1) --

Hi, I was wondering why the results from lm and dynlm are not the same for what I think is the same model. I have just modified example 4.2 from the Pfaff book, please see below for the code and results. Can anyone tell my what I am doing wrongly? Many thanks, Werner set.seed(123456) e1 <- rnorm(100) e2 <- rnorm(100) y1 <- ts(cumsum(e1)) y2 <- ts(0.6*y1 + e2) lr.reg <- lm(y2

Hi all, I have problem for receiving fax from multiple page fax that sent from fax machine (analog). The error is : WARNING T.30 Page did not end cleanly This is my dialplan [inboundfax] exten => s,1,NoOp(**** FAX RECEIVED from ${CALLERID(num)} ${STRFTIME(${EPOCH},,%c)} ****) exten => s,n,Set(FAXOPT(ecm)=yes) exten =>

hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found

2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great!

Hello, We have been working with the ReceiveFax application for some weeks now in order to receive faxes in T.38 and it works fairly well, but there are some faxes that for some reason we are not able to receive correctly. The asterisk version we are using is 1.6.0.6 with spandsp-0.0.5pre4 and the asterisk machine is behind a CISCO mediaGW to be able to communicate with the PSTN. The SIP call

I was wondering if anyone had some sample time series dgp code. I am particularly interested in examples of autoregressive processes and error correction model DGPs. I have attached a more specific example of what I mean. I have tried myself but would hoping someone had some more elegant code that would help me extend my own code. Thanks Luke Keele UNC-Chapel Hill Nuffield College, Oxford

Hello all, When I used commands "ecm.mix and dabipf.mix" to do a simulation (sample size is small 100), I got an error : Steps of ECM, missing value where True/False needed. I've checked the menu, and the option "prior" of ecm.mix said that if structural zeros appear in the table, hyperparameters for those cells should be set to NA. However, it didn't say how to do