similar to: DTMF Detection Problems on VGSM channel

Dear Forum Members, I just purchased two VoiSmart GSM cards. Tried to install one of them on my Fedora Core 5 system, The compilation was not smooth, as expected, but after a small fix, it went through. Then I put two SIM cards in the card's slots. Then I loaded the modules. Then I started the Asterisk. After all I configured the vgsm.conf file according to my settings, that is just changed

Hello, Could anyone help me to figure out the following questions, please: 1. Whenever there is an incoming DTMF signal on the Zap channel, where does the processing actually take place: In Asterisk?; or in Zaptel Drivers? 2. I'm having a problem of double (or sometimes tripple) detection of a single DTMF received whenever I'm calling in through a Mobile Phone. I guess I need to make

Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I" from voismart (http://www.voismart.it/) but the driver is very bad (compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it

Dear Asterisk Comunity, I'm thinking about developing a callback application based on the following scenario: 1. Customer Calls the outgoing number which is a PSTN line connected to my Zap channel 2. Asterisk captures the Caller ID and calls back the customer. 3. As soon as the customer picks up the phone, asterisk plays a promt to enter the Destination number. 4. Asterisk Connects the

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Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere

Hi, is there a possibility to tell zaptel or Asterisk to modify the DTMF sensibility? The problem what i have is that the Asterisk don't get all Numbers which the analog-FAX dial, let say the FAX dial 123456789 the Asterisk get to number 24679. I think that can be to DTMF Tone duration or the Frequenzy. so you got yna idea what it could be? Thx for helping me. Bye Timm

Hi, I've configured samba with ldap support and all is going well with normal users. I wanted to know how to give special right/privileges to users using any windows tool. I tried to use w2k's support tool (Active Directory Users and Computers) and nt's srvtools(usermgr) but none of these worked. win2k says "can not connect to pre-windows 2000 domain" nt connects and

Hi, We are running a samba fileserver, access controlled using posix acl (right 770, with users/groups on the filesystem level. Therefore samba shares look like this: [share] path = /srv/academic read only = no writable = yes create mask = 0770 directory mask = 0770 Now certain users complain that they cannot access certain folders, but looking at the folders from the linux fileystem, their

I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061028/87014bd9/attachment.htm

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hola, compañeros. Pido ayuda con algo que sé que tiene que ser simple, pero la presión de tener que sacarmelo de encima me simplifica a mí demasiado y no me doy cuenta. Tengo una matriz de datos en la que tengo características tipo factor, necesito trasponer esa información a una matriz de datos binarios, en función de algunas combinaciones de esas variables. Unas combinaciones tienen que pasar a

Hello all, I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233445 at 192.168.1.100 SIP/2.0 From: "123456789"<sip:*123456789*@192.168.1.100>;tag=333333333 To: <sip:*6700112233445 at

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

Hi, Try this: lines1<- readLines(textConnection("gene1 or1|1234 or3|56 or4|793 gene4 or2|347 gene5 or3|23 or7|123456789")) lines2<-readLines(textConnection(">or1|1234 ATCGGATTCAGG >or2|347 GAACCTATCGGGGGGGGAATTTATATATTTTA >or3|56 ATCGGAGATATAACCAATC >or3|23 AAAATTAACAAGAGAATAGACAAAAAAA >or4|793 ATCTCTCTCCTCTCTCTCTAAAAA >or7|123456789

Hi, I purchashed a Telular Phonecell Fixed Cellular Terminal. I hooked it up to my wildcard fxo card. I can receive calls and these calls are passed on to the Asterisk Calling Card application. My problem is that i can't get DTMF to work properly. If a pin number is 484443543639 i get 4844444333544336639. how can i sort out this problem. Please would like ur urgent assistance.

Hello, I am using Asterisk SVN, a cellular phone, and chan_mobile to run a small home PBX with two analog telephones connected to a Linksys ATA using SIP. It works great (except for some Bluetooth adapter bugs that I am still trying to beat...seems the misaligned audio detection still needs work), but I have encountered an interesting issue. If I am using an automated system that accepts input

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger