similar to: Overriding indications for IAX2 calls

Good morning everybody, Can someone explain to me the interconnection between these four things: indications.conf, SetLanguage(), zaptel.conf and ring-back ? If there is any !! :- ) I am having this case where some users cannot hear ring back from a DeadAGI script and it seems to be interconnected to these items. These users are from the iaxfriends table, they _can_ hear ring-back from a

Apologies if this is an old question; I've searched the list and the wiki but have not been able to find a definitive answer. I have an Aastra 480i phone registered with * 1.2.4; I want to generate UK ringback tones when the handset dials another internal extension. On my Zap channels, I have this in place by editing /etc/zaptel.conf; however I've had no luck with the Sip handset (I have

Hello all, Over the weekend, I setup and linked an Asterisk box at another site to the Asterisk box here. The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100 phones. The phones at the other end are Grandstream BT-100 SIP phones. The Cisco phones run SIP 7.1 (upgraded last Friday from 6.1), the Grandstream phones run 1.0.4.68. Both Asterisk boxes are running stable CVS

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thanks Andrew. ---------------------- indications.conf [general] country=uk [uk] description =

Hi, The iax.conf is below and the trace. Any ideas please? disallow=all allow=g729 trunk=yes qualify=yes qualifysmoothing=yes nat=yes canreinvite=yes context=OutboundWS transfer=mediaonly Executing [082449627 at private:1] Dial("SIP/919-094d6e60", "IAX2/ECom-iax/2782449627|60|") in new stack -- Called ECom-iax/2782449627 -- Call accepted by xxx.xxx.xxx.x (format

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

Good afternoon all, I was wondering if anybody knows of a way to limit the number of calls going out over an interface (or respond with some sort of 'circuits-busy' message?) The reason I ask is my outgoing bandwidth is only 128kbit and if there are any more than 2 calls going over the internet interface the QoS is reduced dramatically for all calls. Failing that, does anybody know if

Hi all, This is my first post here - I started with Asterisk a few days ago and have "fallen in love" - fantastic product. I've only got softphones connected at the moment - I'll probably order the FXO/FXS cards in about a month (and then think about getting some hardware SIP phones). Our current phone system is quite a few years old and isn't growing with us (when a single

Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it

I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk. It works ... Partially. We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA. First of all, when I make a SIP call to the unit with a simple Dial() command (no "r", so Asterisk shouldn't generate its ringback tone) I hear Asterisk's ringback tone anyway (I'm

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in

I've got a problem with audio output from the Asterisk console. I'd really appreciate any help. I'm simply trying to dial out to a phone on PSTN. My extensions.conf entry is as follows: exten => _1NXXNXXXXXX,1,Dial(IAX2/####@voxee/${EXTEN}) exten => _1NXXNXXXXXX,2,Hangup After starting asterisk and dialing, I hear a ringback tone through the console speaker, and the PSTN

Hello, I've got strange problems trying to run asterisk with MGCP ip phones (Thomson ST2030). Situation: "user A" <----- pstn -------> ASTERISK <----- mgcp ------> "user B" "User A", connected behind a PSTN, tries to call "User B". After dialing "User B"'s number, call comes to ASTERISK, ASTERISK contacts

Hi, everybody: I have just set up a system with a regional VOIP provider. I have two IAX channels to this provider. Incoming calls ring a configured SIP extension immediately, but outgoing calls are delayed for about 8 to 10 seconds before the remote PSTN end starts ringing: > -- Called [IAX2 channel] > -- Call accepted by [IAX2 provider IP] (format ulaw) > -- Format for

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.10.0.1 DAHDI-Tools-v2.10.0.1 dahdi-linux-complete-2.10.0.1+2.10.0.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete * Fixes an issue where

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.10.0.1 DAHDI-Tools-v2.10.0.1 dahdi-linux-complete-2.10.0.1+2.10.0.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete * Fixes an issue where

I am noticing that calls that come from our IAX pstn gateway provider and terminate to our Asterisk IVR do not receive ringing when an extension is dialed. For example: 1. An inbound PSTN caller calls our number 2. Asterisk answers and provides greeting 3. PSTN user dials extension of internal SIP phone 4. No ringback is heard from PSTN callers perspective 5. SIP user picks up or the

Typo in your OS79XX.TXT P00 ? instead of P0S !? -----Original Message----- From: Michael L?jtnant [mailto:ml@zyxel.dk] Sent: 17 August 2004 13:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you