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...========== Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack -- Called <FWDNUMBER>:@iax2.fwdnet.net/612 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00014ms SCall: 00003 DCall: 00000 [65.39.205.121:4569] VERSION : 2 CALLED NUMBER : 612...

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside

...(language 'en') == Auto fallthrough, channel 'SIP/100-081d9478' status is 'UNKNOWN' ========================================= extension.conf ======================================== ;exten => $name,$priority,$application() [globals] ALANL=SIP/100 OUTBOUNDTRUNK=Zap/1 FWDNUMBER=867*** ; My FreeWorldDialup Number FWDCIDNAME="Alan Lord" ; My CLI FWDPASSWORD=****** FWDRINGS=${ALANL} ; Phone to ring FWDVMBOX=1000 ; Voice Mail Box (not yet setup) [zap_incoming] ; Channel defined in zapata.conf exten => s,1,Answer( ) exten => s,2,Set(TIMEOUT(digit)=5) exten =&g...

Hello, In my sip.conf I have: ;Register and forward FWD numbers to internal extensions register => FWDNUMBER:PASSWORD@fwd.pulver.com/9500 Which should register Asterisk at FWD and then when any calls are made to FWDNUMBER those calls should be forwarded to extension 9500 as specified in the extensions.conf. What I am getting is it is trying to dial the 9500 (IAX Firefly) client twice when a call is made...

...5036 bandwidth=high disallow=lpc10 jitterbuffer=no tos=lowdelay context=from-iax ;Register with IAX Peers ; register => 398xxx:xxx@iax2.fwdnet.net [free_world_dialup] type=user auth=rsa inkeys=freeworlddialup extensions.conf [globals] PHONE1=SIP/1001 PHONE2=SIP/1002 FWDNUMBER=398xxx [from-iax] ;Handle iax calls from FWD to my FWD# exten => {FWDNUMBER},1,Dial(${PHONE1}&${PHONE2},20,tr) exten => {FWDNUMBER},2,Hangup If anyone has any other suggestions I'd be very grateful :) -Cheers Max. -- Max Lock

Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast. My config is quite simple, I'm just following examples and the Wiki: I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

...erisk Event Logger restarted Asterisk Ready. *CLI> -------------------------------- I have configured some sample sections in my extensions.conf file to test asterisk features: ----------------------- [demo] exten => 1,1,SetCallerID("My Self-Testing") exten => 1,2,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${FWDNUMBER}) ; call my own FWD number using iax subscription at FWD. I'm receiving this call in X-Lite soft phone configured at another PC (not at the asterisk box) ; This woks great! I can "dial 1@demo" from console and get an incoming call at the...

...net ; US toll free access exten => _1800.,1,Macro(tollfree,${EXTEN}) exten => _1866.,1,Macro(tollfree,${EXTEN}) exten => _1877.,1,Macro(tollfree,${EXTEN}) exten => _1888.,1,Macro(tollfree,${EXTEN}) [macro-tollfree] exten => s,1,SetCallerID,${FWDCIDNAME} exten => s,2,Dial,IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/*${ARG1},60,r) exten => s,3,Playback(invalid) exten => s,4,Hangup It works fine when I dial from inside, but when I try to dial via DISA I get congestion, I cat get in, system accepts my passwords but gives me congestion. Why? -- #Joseph

...s fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten => _2431XXXXXXXXXX,2,Dial,IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/**${EXTEN},60,r) exten => _2431XXXXXXXXXX,3,Playback(invalid) exten => _2431XXXXXXXXXX,4,Hangup exten => _2431XXXXXXXXXX,103,Bus The output is: -- Executing SetCallerID("Phone/phone0", ""Chris"") in new stack -- Executi...

...'s context...) exten => s,n(dial_time),NoOp(======================> Chan Type ${Protocol}) exten => s,n,NoOp(======================> Chan Name ${Phone}) exten => s,n,NoOp(======================> Channel User's context ${CalledUsersContext}) exten => s,n,Dial(Local/${FWDNumber}@${CalledUsersContext}/n) Results at console on verbosity 9: SIPPEER() Works as advertised when I dial a SIP phone which has been call forwarded -- Executing NoOp("Zap/1-1", "======================> Chan Type SIP") in new stack -- Executing NoOp("Zap/1-1&quot...

I have configured iax.conf and extensions.conf as instructed on FWD website (http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76) and I can successfully receive calls and make test calls to 612, 613, etc. The problem is that I can not make a call to another FWD user. Here is what asterisk says: -- Executing [393xxxxxx@default:1] Set("Zap/1-1",

....10.8.252 ; ; Sample Registration for IAX + FWD ; ; To register using IAX with FWD, it must be enabled by visiting the URL ; http://www.fwdnet.net/index.php?section_id=112 ; ; Note that you need an extension in you default context which matches ; your free world dialup number. Please replace "FWDNumber" with your ; FWD number and "passwd" with your password. ; ;register => FWDNumber:passwd@iax.fwdnet.net ; ; ; You can disable authentication debugging to reduce the amount of ; debugging traffic. ; ;authdebug=no ; ; Finally, you can set values for your TOS bits to help improve ;...

...Username Perceived Refresh State 65.39.205.121:4569 596146 <Unregistered> 60 Timeout My section from extensions.conf is: ; Outgoing to FWD/IAX network, prefix calls with 7 exten => _777.,1,SetCallerId,${FWDCIDNAME} exten => _777.,2,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:3},60,r) exten => _777.,3,Congestion I'm using Asterisk 1.0.2 Any help would be very welcome! Thanks Mike

Guys. Can you dial 800 and 888 toll free numbers using FWD? how do you dial them cause I tried using 1800xxxxx and 1888xxxxx and I simply get a "nobody can asnwer the call" signal on asterisk. Can you dial 800 toll free from FWD?

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040802/d9eed8fc/attachment.htm -------------- next part -------------- ? Hi Can anyone give suggestion why we need STUN while using asterisk behind the NAT. Regards Shan.

Hello, I am trying to setup my Asterisk box to accept calls from the FWD network. I've followed all the config advice / samples I've found on the web. Making calls to devices on the FWD network from my Asterisk box works flawlessly, but whenever I try to call my Asterisk box from a FWD client I get a busy signal, and a "Call Disconnected" 486 error. What's odd is that I

...ait("Zap/1-1", "1") in new stack -- Executing BackGround("Zap/1-1", "custom/pls-wait") in new stack -- Playing 'custom/pls-wait' (language 'en') -- Hungup 'Zap/1-1' Here is my incoming extensions.conf dialplan: [globals] FWDNUMBER=223611 ; your calling number FWDCIDNAME="Robert Webb"; your caller id FWDPASSWORD=password ; your password FWDRINGS=IAX2/rwebb ; the phone to ring FWDVMBOX=2002 ; the VM box for this user ANALOGPHONE=zap/2 OFFICEPHONE=SIP/2002 LAPTOPPHONE=IAX2/rwebb VMBOX=2000 PSTNOUT=zap/5 [fromPSTN]...

Hi, I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911) does anyone know how to fix this, any ideas,? does anyone from mexico has done this? Any comment will be highly appreciated, Regards, Claudio -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ecallerid=no callerid=asreceived channel => 3 context=incoming_88097074 channel => 4 ============================================================================= extensions.conf: ================ [general] static=yes writeprotect=no [globals] ;IAXINFO=xxxxxxxxxxxx:xxxxxxxxxx ;FWDNUMBER=xxxxxxxxx ;FWDPASSWORD=xxxxxxxxxxxxx ;NUFONEUSER=xxxxxxxxxxx ;INTERCOMPASSWORD=xxxxxxxxxxxxx ;IAXCOMMPASSWORD=xxxxxxxxxxxxxx #include "passwords" ; above info is in that file! CONSOLE=Console/dsp ; Console interface for demo TRUNK=Zap/g1 ; Trunk interface TRUNKMSD=1 ; MS...