similar to: zap not coming online on fedora 8

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Ok.. will be there... -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Thursday, October 04, 2007 12:50 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 39, Issue 12 Send asterisk-users mailing list submissions to

hi all, how to set the caller id facility for the TDM400p card. Please help me thanks, sandeep.s ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Tuesday, January 15, 2008 3:09 PM Subject: asterisk-users Digest, Vol 42, Issue 51 > Send asterisk-users mailing list submissions to > asterisk-users at

Dears; To patch libpri: I just place the patch file in the libpri source directory and then I run make and make install? Or I need to compile the dahdi and asterisk also? If the problem stayed, do I have to go for previous libpri version? Or for previous dahdi version and asterisk version? Regards Bilal ----------- > bilal ghayyad wrote: > > But I am afraid it is a bug because I

ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: > Admittedly I have not used the ExternalIVR app. Is it any good? > > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, > it can do it, but boy it is UGLY. There's also the fact that you can't > call Backgound() in a macro, which forces you to use Read() which >

Hi, I am new user on asterisk (for that matter linux) and i have lot of embedded programming experience. We have a new project from our client, to design a box that takes the telelphone line as input and route the line to the respective user with different ring tones. The box should be programmed by the users with buttons. Features. 1. I should be able to store some .wav files for different

Dear; Thanks a lot for guiding me. Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch? Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5): libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch It gave me that patched detected as shown below (example of one file, and I got same for other files): patching file

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > Are you using the wifi on on the cellphone? The peer IP is showing as > 192.168.200.3 which is not a routable address. Unless things have changed, > double NAT configurations do not work. Hi Steve, My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct in Internet. But maybe my Provider does a

getting calls, but I can only send calls from my main machine IP address so I can't control where I am sending calls to. I am hoping to have this developped somehow (a per SIP peer bindaddr and bindport), even if it means some bounty. I can't imagine this being this difficult, so a few of us who need this putting a couple hundred dollar would probably do it. Mike > -----Original

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

I'm using SixTel as a test (Opened account w/ $10) and am happy with them so far... In their basic service package, they don't charge a monthly fee, and it's outbound only, and you get charged for every minute. I paid for a DID, which is $1.50 or so per month, and it lets me receive inbound calls, which I also pay for by the minute.. I don't mind this for a service like this

n active project, than a dead one. Otherwise who is going to patch vulnerab= ilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What

with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

your sip communication altogether. Have you tried changing IP address of your asterisk server? If changing IP works, then probably your provider has blocked you sip communication by IP only. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-23 7:22 AM, "bilal ghayyad" <bilmar_gh at yahoo.com> wrote: Hi All; I have my friend that use his mobile (Nimbuz) to connect for the

Hi Danny; Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager? Regards Bilal ------------------------- It depends on how you are configured. The gui interfaces using Asterisk Manager, so you get the Same IO from the gui that you would get from a native manager session. -----Original Message----- From: asterisk-users-bounces at

On Sun, Jun 7, 2015 at 10:05 AM, Luca Bertoncello <lucabert at lucabert.de> wrote: > Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > > Are you using the wifi on on the cellphone? The peer IP is showing as >> 192.168.200.3 which is not a routable address. Unless things have >> changed, >> double NAT configurations do not work. >> >

I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to