search for: 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 working....

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...

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot g...

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 username=31307110000 And my extensions.conf: [from-budget...

...1 VERBOSE[1563]: -- Executing [1;36;40mSetCIDNum[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m", "[1;35;40m31437110323[0;37;40m") in new stack May 8 10:47:11 VERBOSE[1563]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m", "[1;35;40mSIP/XXXXXXXXXX@budgetphone.nl|30|r[0;37;40m") in new stack May 8 10:47:11 DEBUG[1563]: SIMPLE DIAL (NO URL) May 8 10:47:11 DEBUG[1563]: Outgoing Call for XXXXXXXXXX May 8 10:47:11 DEBUG[1563]: XXXXXXXXXX is not a local user May 8 10:47:11 VERBOSE[1563]: -- Called XXXXXXXXXX@budgetphone.nl May 8 10:47:11 DEBUG[...

...lso make calls, mainly only sip, i don't mean trixbox or freepbx. anyhow my main problem is the sipgate.de thing, below are my sip.conf and extension.conf thx for any help sip.conf: [general] language=en disable=all register => user:pass at voip.eutelia.it/number register => user:pass at budgetphone.nl/number register => user:pass at sipgate.de/number register => user:pass at sip.voiparound.com/number register => user:pass at sip.webcalldirect.com/ register => user:pass at ixcall.net/number register => user:pass at freedigits.net/number register => user:pass at sip.messagenet...

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

...SetCallerID(${T2Y1}) exten => s,2,SetCIDName(${MYNAME}) exten => s,3,Dial(SIP/${ARG1}@t2y,${ARG2}) exten => s,4,Playback(new/acnt-or-cir-busy-now) exten => s,5,Hangup I do have a [t2y] reference in my SIP.conf file [t2y] type=friend secret=PASSWORD username=USERNAME host=budgetphone.nl dtmfmode=rfc2833 fromuser= USERNAME fromdomain=budgetphone.nl username= USERNAME insecure=very nat=yes qualify=no register=yes I tried changing to change the reference t2y in the context in extensions.conf to @budgetphone.nl and @sip.budgetphone.nl, but then I get a authorization...

...P/voipbuster-ac66 -- Started music on hold, class 'default', on SIP/voipbuster-ac66 -- Stopped music on hold on SIP/voipbuster-ac66 But.... If somebody rings me and I put him on hold he hears nothing: Debug output for incoming call: -- Executing SetCallerID("SIP/gw02-mci.budgetphone.nl-42ba1908", ""prive" <xxxxxx>") in new stack -- Executing Dial("SIP/gw02-mci.budgetphone.nl-42ba1908", "SIP/sipura&SIP/gs4") in new stack -- Called sipura -- Called gs4 -- SIP/sipura-7685 is ringing -- SIP/gs4-4a86 is ring...

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

-----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 -