similar to: Question for someone running hylafax off *.

I think I'm very close to getting IAXModem and Hylafax going, but my current inbound hylafax logs show this: Feb 23 10:09:37.98: [ 3638]: MODEM <Empty line> Feb 23 10:09:37.98: [ 3638]: MODEM TIMEOUT: waiting for v.21 carrier Two questions - 1) Does anyone know what step I missed here? (I.e. please help!) 2) Is there a document I should be working off of? Google doesn't seem to

I have all three running on the same box. I say OT because it appears asterisk is doing it's job just fine. It must be an IAXmodem or faxgetty (hylafax) problem When faxes work, they look great. I have ten IAXmodems setup with different ports and they register fine. I have ten faxgettys that startup fine. I start the IAXmodems and then faxgettys in inittab. They are setup as a roll

Hi List I've a CID lookup hooked onto an inbound route (i m using trixbox) ...it runs well but it returns the value as "CIDNAME<CIDNUMBER>" ... if i just want to display the CIDNAME [leaving the quotes and <CIDNUMBER>] .. how can i do it ? do i have to edit some macro in extensions.conf ? rgds Sriram -------------- next part -------------- An HTML attachment was

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

In trying to get the asterisk and faxing working I had to resolve to using iaxmodem and hylafax. I have incoming working, but outgoing the other fax rings but it would appear from web searches that the fax signals are too low to be "heard" I can read about rxgain and txgain for zapata. my fax setup goes direct from aterisk <-SIP-> SIP Provider <- Fax Machine -> It never

On the subject of faxing - is there a way to interface Asterisk and Hylafax running on the same box? I have tried the directions at- http://www.voipinfo.org/wiki/view/T38modem+configuration+with+Asterisk and I just can't get it to work. Does anyone know of a way? (Even mention paid software)... Michael

Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif Unsuccessful Asterisk Command: same => n,System(mutt -s "New fax" elder.arohuanca at

hi folks. i'm experimenting with iaxmodem + hylafax using DID to determine where to send the fax to it's final destination. however i have difficulties passing the DID information from iaxmodem to hylafax. in extensions.conf: exten => _XXXX,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r) exten => _XXXX,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r) exten => _XXXX,n,Busy exten => _XXXX,n,Hangup

Hi everybody, since some time I am looking for a current and reliable solution to send and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction with Asterisk. For testing I am using a HFC-ISDN passive PCI-card, in production a Digium Dual T1/E1 PCI-card will be used. I run CentOS 5.5 (Kernel 2.6.18) and Asterisk 1.4 (but I also can use 1.8) but did not find any solution where I

Hi all, I know Hylafax is an application and not Asterisk but I'd like to post a problem found in configuring such application and Asterisk. I am able to reveive fax,but , I can't receive it in e-mail. Although I put my e-mail in /etc/hylifax/Dispatch I can't receive. Anybody know where I must to add something else in order to make it works! Thanks in advanced!! Att,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a ?crit : > I think the "488 Not acceptable here" is occurring because the channel > connecting through is not T.38 capable, that will be the IAX channel > from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

Hello, I'm trying to fill CID data into the astdb using AsteriskWin32's asterisk.exe, to no avail: The batch file stops after the first line, and just waits: ---------------------------------------- rem c:\cygroot\mystuff>import.bat rem rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 "My cellphone"' rem rem Asterisk module

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with

Hi All, I'm looking for a way to filter the AstDB cidname family to show only those entries with a specified area code in the Asterisk CLI. If this were a SQL database it would be something like: SELECT number, name FROM cidname WHERE number LIKE '1234%' I've tried "database show cidname 1234*" and substituted "%", "$", "-" for the wildcard

Admittedly this is probably pretty stupid of me, but there are just some things I can't understand by reading documentation. Any suggestions or recommendations about how to handle my problem are greatly appreciated. I'm trying to achieve the same functionality as my Nortel PBX, without rewriting much 'C' code. In my dialplan I'd like to compare two variables as a means of

I am using Sipura 841 phones and Asterisk CVS-v1-0-06/14/05. Whenever a call comes in with blank CIDName or CIDNum the phone reports the respective variable as "asterisk". I can manually set the variables to whatever I want: CIDName (alpha-numeric) & CIDNum (Numeric). But if I try to make them blank, or null, or maybe throw some alpha characters into CIDNum, they get reported

Hi All, I need to use partial matches on the CIDNAME family I have stored in AstDB. For example, an organisation might have several numbers with the same area code and the same first few digits: 1234 567890 1234 567889 1234 567824 ... I'd like to store these (e.g.) as CIDNAME/12345678* (where "*" is a wildcard) so that I can retrieve the organisation name from

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>