similar to: Playing sound before dialing

Hi, I'm using http://www.portunity.net/ I configured now asterisk with the following setup: iax.conf: register => XXXXXXX:YYYYYYY@iax.iaxport.de [portunity-out] type=friend host=iax.iaxport.de username=XXXXXXX secret=YYYYYY context=incoming-portunity notransfer=yes [guest] type=user context=default ;callerid="Guest IAX User" And in extensions.conf: [default] ;exten =>

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

>From the local machine the following command is executed: /usr/local/bin/rsync -aHnuv serverX:/ / --exclude-from=/rsync.exclude --rsync-path=/usr/local/bin/rsync --ignore-existing > /var/tmp/rsync.stdout 2> /var/tmp/rsync.stderr I have never used the rsync command. The above command was used by a former sysadmin to "synchronize" two servers. However when I ran the command

On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > Can't comment on this one, as I never use AGI to dial. > My AGIs just set the context, extension and priority, > and exit to the dialplan to do any dialling. (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537) I would like to do this, but I am having trouble figuring out how. I have

Hi guys, Any idea why I am getting this error when someone tries to send me a T38 Fax?

Hi All, i have created one prepaid PERL AGI script to integrate asterisk users in our current Oracle Billing System. I am using $AGI->exec('Dial', $dialstr); in script after getting the MAX time out for the priticular call. But when the channels increase on my asterisk more than 50-60 asterisk get crashed and i am suspecting the cause is of AGI Script. because when i check ps on

Looking at the code, it would appear that the 'callstart' column of the cdrs table should really be called 'callend': $dialstr = "IAX2/$res->{path}/$phone|30|HL(" . ($maxtime * 60 * 1000) . ":60000:30000)"; $res = $AGI->exec("DIAL $dialstr"); $answeredtime =

Hi everybody, I sent an e-mail this morning regarding SIP / IAX2 with no ring-back, I now succeeded to pin-point the problem, here it is, if I dial a provider directly from extensions.conf I get ring-back, if I dial from an AGI script I don't get the ring-back but it calls anyway. I use 1.0.9. Any hint would be appreciated ! Thanks, Frederic ;Calling this one does not give me ring back

Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 -> IVR answers and puts caller to the chosen queue -> Someone picks up the phone (Internal ext. 321) -> CallerID shown on customers

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem

Hi, I am having a strange problem with enum and AGI. Here is what happens: I have in my agi something like that: foreach my $resolver ("e164.arpa", "e164.info", "e164.org") { my @enums = get_enums($phone, $resolver); foreach my $enum (@enums) { $dialstring = $enum .

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

I just got my wife this phone, and when calls come in on the queue it will not ring, if I dial the ext it rings, here is the config from the tftp server: Thanks unidencom.txt # UIP200 Mass Configuration System Generic File # Notes: # 1. Lines start with '#' are comments # 2. To leave a field value unchanged (as saved on local phone), leave value to blank. # 3. To set a field's

Hi Saulius, The spec you sent me is simply a Megatec excerpt: http://www.networkupstools.org/protocols/megatec.html So Socomec is simply a megatec rebranding (selling by simply putting its name on). As per the NUT compatibility list (http://www.networkupstools.org/compat/stable.html), Sicon are supported by the powercom driver. But since it's a megatec unit, you should preferably try the

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

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

Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars =

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone written an extension that will take a 12 digit number, convert it to an IP address so that you can make a sip call to it. Chris