similar to: re: questions about sip requests to asterisk 1.2

We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at sales@amarfone.com. Ehsanul Karim

hello all, i need some help and after trying the wiki i'm even more confused than i was. i'm trying to set up call forwarding and running into problems... i want the most basic call forwarding imaginable. 1. caller dials extension (say, 154) 2. dialplan is updated to forward caller's extension (based on CALLERIDNUM) to voicemail, instead of ringing his endpoint. 3. caller is

Hi all, i'm trying to integrate cepstral and asterisk, and i have a problem i'd appreciate any help with (i know it's a bit tangential, but i figure this is the place with the most knowledge of app_swift and asterisk). I've installed swift from cepstral.com with alison's voice, and it works fine, from the command line i can do swift "hello there" -o test.wav and then

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

Hi all I have a couple of questions maybe you guys can help me with them I have sip phones , SER server , Asterisk. what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone -> SER -> ASTERISK -> SER -> PSTN 2) sipphone -> SER ->ASTERISK ->PSTN on the first option i am trying to return the call to the ser

hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered

Hello all, I'm having a bit for a problem with the dial command limit option. I have the following dial command (executed from inside the a2billing agi) AGI Script Executing Application: (Dial) Options: ( IAX2/username@voipjet/18005551212|30|HL(60000:20000:00000)0) Now, from what i read in the wiki, this is supposed to limit me to one minute (60000 ms), and warn me when there are 20

Hi guys, I have a weird thing happening sometimes with users calling from a GrandStream phone through Asterisk onto a PSTN. Sometimes after a user hangs up a call on a GrandStream phone the phone starts ringing after a couple seconds. When the call is answered there is no one there. Anyone had this before ? Kindest regards David Wilson _______________________________ D c D a t a Tel +27 33 342

hello all, i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to 1.23 requests from SER to asterisk die quietly, no

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

Jeremy There is NetApp simulator that may help you ! Check now.netapp.com -- Yair

Hello: I noticed in few references that asterisk and ser and complementary. Meaning asterisk handles connections to PSTN and voicemail but SER is better for routing SIP traffic. Is anyone using just asterisk for production purpose. Meaning serving a high number of callers. Is it mandatory to use SER behind asterisk? your feedback would appreciated. -Gaid -------------- next part

Hello, I have a problem, I can't install the package 'mgu74av2cdf'. I downloaded the zip file, yet when asked the R console to install it from a zip file, I got the answer: "Error in file(file, "r") : unable to open connection In addition: Warning messages: 1: error -1 in extracting from zip file 2: cannot open file `mgu74av2cdf/DESCRIPTION'

Hello, I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring but I don't hear the caller and the caller doesn't hear me (all IP Phones have the same problem). This problem appear also if the call is directly send to the second E1 of the digium card who is connected to an IVR. It does not depand on the charge of the server (I have the problem with only one call).

My SIP seems to be configured correctly as I can dial out and my minutes show up on my broadvoice bill, but whenever anyone calls my broadvoice # inbound they just get a busy signal. I dont get anything in the logs saying anything came in from broadvoice at all. Has anyone had this/simmilar problem with inbound from Broadvoice? Any suggestions? Thanks Neri -------------- next part

I started to learn the R language, but I didn't suceed to use an external file. Let say that I have an excel file called "test1.xls" in the directory "C:/program files/R/rw2000/external_files" that looks like that: name mark yair 80 yosi 70 ... In the appropriate directory I wrote this: x<-read.delim("test1.xls") or this:

Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal.

Hello We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since. We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline: It does show up in asterisk a few seconds after the UIP200 reboot: -- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200 but

https://bugzilla.samba.org/show_bug.cgi?id=3718 Summary: RSync should verify permission/time before commiting a change Product: rsync Version: 2.6.5 Platform: Sparc OS/Version: Solaris Status: NEW Severity: major Priority: P3 Component: core AssignedTo: wayned@samba.org

Hi Yair, Please let me if you managed to fix the DTMF tone issue, which you were experiencing couple of months ago. If not can you share any advancement. I'm currently experiencing the same issue, I can make outbound calls but DTMF will not work when dialing IVRs. My configuration is asterisk@home 1.5, registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833.