similar to: No sound with Playback() or Background()

Hello, I'm wondering how to make it possible for the user to cancel the last entered digit, if he made a mistake. For example, a user calls and starts entering 1...2...4, then he should be able to press, lets say *, to cancel 4 and enter i.e. 3. Thanks Jake -- ------- Domeny w ULTRA NISKICH cenach: ------- .pl - 29 zl, .com.pl - 22,50 zl, reg - 7,50 zl

Hello, I have two asterisk servers and I would like to merge their dialplans. I thought DUNDi would be a natural choice. I created the following configuration on the first server: iax.conf Code: [dundi] type=user dbsecret=dundi/secret context=dundi-local dundi.conf [general] ttl=4 autokill=yes cachetime=30 entityid=00:06:5B:8E:B0:08 secretpath=dundi bindaddr=XXX.XXX.XXX.XXX port=4520

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/104a07f3/attachment.htm -------------- next part -------------- Hello In my IVR menu whenever user select the option number 1 then it should jump to echo context, I think call did jump to "echo" context but I always get the following warning and I hear couple of beeps and then

I've just started consulting for a SME client based in Reston Virginia. They don't know it yet but they are going to need a hosted asterisk service and some DID's. Email me if you are able to provide 10 DID's in Reston (must be able to be ported away!!) and hosted Asterisk with end user configurable IVR etc. Probably only 5-8 users at the moment BUT... they'll be

Hi, I'm hoping someone has a suggestion for us. We have an ITSP that sends inbound traffic to us. Unannounced to us last week they started alternately sending traffic from two IP addresses, instead of the one we knew about. Some calls would pass, and others would be dumped as unauthenticated. I added the 2nd IP to the sip.conf file to allow for this, and everything was fine

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

Hello, I have a setup like this: exten => s,1,Ringing exten => s,n,Wait(3) exten => s,n,Answer exten => s,n,Set(TIMEOUT(digit)=6) exten => s,n,Authenticate(11111) exten => s,n,DISA(no-password|my-context) exten => i,1,Playback(invalid) exten => i,n,Wait(1) exten => i,n,Goto(s,5) exten => t,1,Hangup I need to be able to get back to the

Hi list users I?m trying to do an incluye statement from the Database In my dialplan I have different contexts that defines common services for example ---------------------------------------------------------------------------- ----------------------- [basic_services] exten => 100,1,VoicemailMain() exten => 600,1,Playback(demo-echotest) exten => 600,2,Echo() exten

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) =====> Asterisk ====> FXO (Nokia 7610) ====> Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone?

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

Hello! I'd like to set up dovecot to keep namespaces in database, keep more than one namespace per user. I try to create the simplest rule (even without tuple in db table) to get new namespace, unfortunately it doesn't work. My user_query is: SELECT '/dane/domeny/%d/mail/%n/' as home, dovecot_typ_skrzynki ||':'|| '/dane/domeny/%d/mail/%n/' ||'.'||

Hi, I have been trying to workout how to use the call pickup. So Far, I have the following in zapata.conf [channels] signalling => fxo_ks context => local pickupgroup=1 callgroup=1 channel => 1-3 When I dial *8# all I hear is busy tone. What have I missed? thanks Jay.

Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for "ivr menu".)

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations <sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to doing a transfer from an analog extension to a SIP extension but until the moment I was not successful. I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer => #1 atxfer => *2 verifying previously that the extension uses the arguments "tT" with the Dial

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are