similar to: Problem with IAX2 dialin with portunity

Hi, I have configured asterisk now with ENUM lookups which are working really perfect. Now I want to play a small soundfile before dial the number to inform the caller which protocl is used (SIP, IAX2 or ISDN). How can I do this? With Playback it doesn't seems to work: [iax2-sipport-out] ; with leading 3 using IAX-sipport exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out) exten

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

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

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)

Hello, I've got a problem at the moment, that setting "transmit_silence = yes" seems to have no effect on Asterisk 1.8-Certified. Although it's enabled and "core show settings" confirms, that it is really enabled, there are no RTP packets sent by Asterisk when waiting for DMTF input or when "Wait()" is called. Also, there seems to be a small gap of 2 or 3

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, after some corrections in my settings IAX2 dialin seems to work now. I get the incoming call, but i cannot here anything or can speak. (If I take the call the other side see that the connection is established if I close the call the other site is seeing it too) If I press hold in Idefisk the other side can hear MoH but not me. Asterisk print in the CLI interface that he starts MoH. The

On a SIP phone is it possible to enter the dialplan when the user picks up the phone without having to wait for the user to press an extension? Is is possible to do something like [sip-test] s,1,Answer s,2,Playback(welcome) s,3,WaitExten(30) 1,1,Noop(exten 1) ... t,1,Goto[s,2] --------------------------------- Be a PS3 game guru. Get your game face on with the latest PS3 news and

I would like to use a genuine Windows 2000 server as my PPTP server (getting poptop to work correctly is annoying the hell out of me) which needs to authenticate off of a Samba server configured as a PDC. A Windows-based PDC has a flag for "dialin access allowed" for each user. I searched the Samba archives and see several posts over the years that say it "sounds easy enough to

I have a samba ldap pdc set up. (2.2.8a). I have a windows domain member that is joined to the domain running ras. When users try to dial in to the server they get "Does not have Dial in permission/rights". Is this an option that samba recognizes at this point? Is there a way to tell windows that a user has dial in privileges? Sean Cook Kinex Networking Solutions

Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number (username and password) and it wants to call it to get to Internet. Since it is connected to

Hi all .... I Have installed Bering LRP on Many sites and I am very pleased with the capabilites of shorewall. Howerver I came across a prob that I am unaware ot its solution. Using shorewall 2.0.2f Kernel 2.4.24 On one Site LRP box serves internet outgoing connections through ( static IP ) a DSL line AND an incoming dial-in PPP conection. My shorewall configuration Is based upon the fact

Hi, i was able to solve my problem. During my playing around with * and capi i changed several options in config files. I did this while my * was running. To test if my changes where successful i entered "reload" on * console. This didn't help. But after i stopped asterisk and startet it again, everything worked perfect. So it seems that doing a reload while asterisk is running

Hello, I am trying to use a HFC-PCI (CCD/Billion/Asuscom 2BD0) card in TE mode to dial-in and out with ISDN. The problem is I can not get the card to dial out with a Zap channel. Dial-in is working. I am using bri-stuff 0.1.0-RC4 (but tried also RC3 and RC2k). I tried all combination of "immediate", "overlapdial", "pridialplan". I earlier also managed to dial out

Hi I've 3 iax connections to my provider , each of them have own DID , PH1<----| | \/ PH2<-->|-----| <---------------------------> |----|<-- DID1 | A1 | <---------------------------> |ISP |<-- DID2 PH3<-->|-----| <---------------------------> |----|<-- DID3 I had iax phone on each of this connection , but now I want to terminate all

What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. Any assistance? -----Original

Hello List, because this is my first post to this list, i'd like to introduce myself. My name is Thorsten M?ller, 26 years old and live near Frankfurt/Main in germany. Okay, now to the reason for this posting: I just installed my first asterisk server (Debian 2.2 Kernel 2.4.18-686) with AVM ISDN Fritz PCI card (passive). I followed the configuration on http://voip-info.org and the

This may not strictly be an asterisk question, but not sure where else to post ... I have an Asterisk test server setup with two firefly clients, one on the local lan and one on an external ip address. Both clients are setup the same way and voice calls work fine. The asterisk console reports a "Registered" message for the external client at about one minute intervals but the

Hi.... Apologies for the length of this, but any help would be greatly appreciated ;-)... I have installed and configured a Digium E100P on my Asterisk PBX and I have connected this to an 8 port E1 VoIP gateway on a Cisco 6509. There is also an external E1 link from our Telco plugged into one of the other ports on this gateway and this gateway is in turn registered with a Cisco CallManager

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29