similar to: budgetphone

(this time with subject....) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not

I'm trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I'm new to Asterisk I can't get the error why this is not working. To me it all

Hello all, I've been trying everything I could find, but I can't seem to get my * server connected to my SIP provider (budgetphone.nl). Here's my sip.conf: [budgetphone] port=5060 bindaddr=0.0.0.0 context=from-budgetphone register => 31307110000:secret@budgetphone.nl/500 type=friend host=budgetphone.nl fromuser=31307110000 secret=secret fromdomain=budgetphone.nl

Hi, I am currently trying out the asterisk@home (version 1) release of Asterisk, and I want to configure it as follows: Calls from regular telephony network (PSTN) come in through my VoIP provider over SIP and outgoing calls to the PSTN should be routed through the ViOP provider onto the PSTN network. I thus have no direct PSTN connection, but only a SIP connection. Incomming calls

Hi, here what i have: [2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004] Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 on RedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL. Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives me the following error, -- Executing Dial("SIP/2001-8a8e", "Modem/ttyI0:998004|20|r") in new stack

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Hi there, I'm having a hard time getting outbound calling to my SIP-->PSTN gateway. I continuasly get the following result in my log files: Jun 5 10:07:50 WARNING[1568]: No such host: t2y Jun 5 10:07:50 NOTICE[1568]: Unable to create channel of type 'SIP' Jun 5 10:07:50 VERBOSE[1568]: == Everyone is busy/congested at this time I make the following context in my

I've got a strange situation with musiconhold. It works if I dial my extension 6000: >From extensions.conf: exten => 6000,1,Answer exten => 6000,2,MusicOnHold() Debug output if I call 6000: -- Executing Answer("SIP/gs1-b6ee", "") in new stack -- Executing MusicOnHold("SIP/gs1-b6ee", "") in new stack -- Started music on hold,

Hi, I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. -- Michiel van Baak http://lunteren.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

Hi, I have small confusion in my configuration which is I had some DID's like 044578900-04457999. I was configured dial plan below mention. exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten =>

I'm no longer on the DEV mailing list, but: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist http://svn.digium.com/svn/asterisk/branches/ -- /Nick -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, I am making outgoing call from asterisk to GSM network with the help of VoIP trunk(SIP trunk) then I am not geting any caller ID at destination end. Is this the asterisk issue or VoIP trunk issue? Is this is due to asterisk then how we solve it? I already user Set(CALLERID(num)=XXXXXXXXXXX) in dialplan. ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer

Hello, I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring but I don't hear the caller and the caller doesn't hear me (all IP Phones have the same problem). This problem appear also if the call is directly send to the second E1 of the digium card who is connected to an IVR. It does not depand on the charge of the server (I have the problem with only one call).

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1&t=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate it all here, I've posted my logs and conf files on that thread, too. Problem is that while

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

The Asterisk Development Team has announced the release of Asterisk 1.6.0.27. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Note that support for the 1.6.0 and 1.6.1 branches are moving to security fixes only, scheduled for the first half of May 2010. The Asterisk development team recommends that all users of Asterisk 1.6.0 and 1.6.1

The Asterisk Development Team has announced the release of Asterisk 1.6.0.27. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Note that support for the 1.6.0 and 1.6.1 branches are moving to security fixes only, scheduled for the first half of May 2010. The Asterisk development team recommends that all users of Asterisk 1.6.0 and 1.6.1

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

Sir, i am using goautodial server , bri card is showing ok but when i try to call that showing below , This configuration is in doing in dubai , so kindly help me how can connet the call from this , what is my mistake is in this :::chan-dahdi.conf [channels] #include dahdi-channels.conf language=en context=default usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes

Hi all, I have this weird problem. I'm running asterisk 1.0.3 on Debian Sid (official debian package). We have 2 fritz ISDN cards. All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial("SCCP/michiel-00000004",