similar to: sendURL

Is anyone successfully using asterisk-prepaid-0.3.1? I try to configure but doesn't work. It said that you need to do a few step, copy a few files and that is. Please, if someone has any tips about the configuration, answer me. Sebastian

recently added support (with bug) for SendURL for SIP channel causes problem with nokia phones, as I reported in http://bugs.digium.com/view.php?id=9821 it was quickly resolved, but because I can't find any RFC what it is doing/how to use it, I would like to ask here, if someone using this feature, or do you know according to what RFC this was added, please let me know. thanks PJ

Hi Folks, I am wanting advise on a good soft-phone on Linux. I have looked at Gnophone but cannot seem to get it to compile under debian sarge. I am now looing at sipXphone seem to be picking up that it is not that stable, but perhaps someone here can advise on what softphone I can use on Linux. Thanks in advance, Hamish ------------------------------------------------------------------- |

Would like to deploy * in a small help desk environment (five to ten people) using call queues and some sort of CTI interface to pop Remedy screen data in front of the help desk person receiving the call. Data to be popped would be based on CallerID. Anyone doing something similar? Anyone interfacing to an external Remedy system? Any reference sites that I could read/learn more of the

I am having a really hard time getting gnophone working with asterisk. Gnophone tries to register with my server but there is no response. I can direct incoming calls to gnophone but if gnophone answers them, asterisk does not recognize it. Here is my configuration: iax.conf [jambo] type=user host=dynamic defaultip=136.159.99.100 permit=136.159.99.100 username=jambo secret=fubar

Hi all, I installed all needed RPMs by GnoPhone to be installed without problems but when attempting to install GnoPhone itself I get this message: # rpm -Uvh gnophone-0.2.4-1.i386.rpm error: Failed dependencies: mozilla >= 0.9.2 is needed by gnophone-0.2.4-1 libgtkembedmoz.so is needed by gnophone-0.2.4-1 libgtksuperwin.so is needed by gnophone-0.2.4-1 I'm using

On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: > Hello, and thank you for registering at gnophone.com. Your login > information is listed below: > > Username: miernik > Password: ******* > IAX Phone Number: 17002916107 > > Please login as soon as possible to > http://x.linux-support.net/directory/ to complete the

How do I get Gnophone to register to my Asterisk server? I have set up iax. conf as follows: [tim] type=friend ;username=tstornes host=dynamic ;defaultip=207.194.60.56 secret=1111 context=from-iax callerid => "Tim" <5000> auth=plaintext qualify=10 permit=0.0.0.0/0.0.0.0 and extensions.conf includes a section in the context from-iax: exten => 5000,1,Dial(IAX/tim/s|100|r)

Hi! I'm using asterisk with a few kphone SIP-Clients. The registration process seems quite OK. But there are some problems: Calling other registered users is possible, but the rtp-stream is not reaching the right port, so you can hear nothing. In ethereal you can see, that the SIP/SDP fields addresses different ports at each client, so client A sends to port 32000 but client B listens on

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

DOes anyone have any good suggestions as to good SIP or IAX clients for linux? I have set up and am currently testing asterisk in a controlled environment. I have gnophone running on one of my boxes but the gnophone site has been down. So I can't seem to fing the IAX and Ix-devel rpms or the gsm and gsm-devel rpms. So that prevents me from setting up gnophone on another box. I am

I've been reading all I can in order to try to implement an asterisk setup. In speaking with someone the other day, they advised they thought there was a way to make a SIP or softphone (gnophone) go straight out to the vonage network through asterisk. I already have vonage set up with an ATA 186. What I'm wondering is is there some way that I could direct dial from a softphone

Hi everybody, I have two toll free numbers pointed to my asterisk server. My toll free number provider gave me two 7 digit dnis numbers. Both numbers land in the extensions. How to make the softphone (xlite) know that the call has landed through which number? I think the differentiating stuff is the dnis numbers. Is there any way, where I can notify the softphone in regard with the dnis number?

I want to be able to demo * on a notebook at a client's site. This means no FXO gateways; just 2 sip phones (like SNOM) and maybe a softphone (GnoPhone?). I already have RH9 running on my notebook. I would like to have one SIP phone dial and go through IVR before making a choice and ringing the other phone extensions. Of course the notebook would have to be running Asterisk. How can i setup

Hi All, I know, IAX is NAT friendly, but... I have a problem running gnophone from a box behind NAT firewall. I can register gnophone with * through NAT, but when I try to make a call it instantly disconnects. CLI iax show peers command tells me that peer is unreachable. However this peer is registred. Gnophone also tells me that it is registred. It seems that registration handshake has

hello everybody well...while trying to make gnophone to gnophone call using IAXtel's PBX server......i am not being able to establish a connetion.... possibly whats happening is, we are not being able to transmit our message packets properly and result is its not being able to establish connection n is resending frmaes again and gain here is message we are getting Tx-Frame Retry[000]

hello all i'm interested in setting up a small pbx using asterisk and the primary goal is keeping the cost down. the general layout of the net is as folows: * 4 phone lines (2x isdn+ 2x analog) [or 2x isdn + 1x analog, as one might be put aside for a traditional phone/fax with no fancy stuff] * a server box * several client hosts (all linux with x) currently, the isdn lines are unused (to

I hope this isn't something newbiesque and Steve will denigrate me. . . I just built gnophone, and I'm having trouble figuring out just how to sync the "Telephony Preferences" in gnophone and iax.conf on my asterisk server. I am heading on a trip and I'm pretty sure I'm going to be behind a NAT gateway, and my asterisk server is *separately* behind a NAT gateway. I

Hi All, as good? I am trying to make a call for the Unicall channels and after the exchange of signalling and sending of the digits asterisk locks up with the sending of the signalling "E" and the TELCO says that asterisk would have to send signalling "F", as to make for asterisk to send signalling "F"? The TELCO says that the signalling "E" is

The subject pretty much says it all. Does Asterisk support DNIS, and if so, what kind of connection is required? (T1, PRI) I've got a wink start T1. I've read comments that say the DNIS will be seen as an extension, but I'm seeing each digit of the DNIS as a separate extension. So in my case I send DNIS of 12345, Asterisk will jump from extension 1 to extension 2 to extension 3 to