search for: trunk_1

...is what happens when i make a call, i have put xx on the numbers and passwords. The dialplan strips the 0 in front of the number. -------------------------------------------------------- -- Executing [0043401xxxx at numberplan-custom-2:1? ] Macro("SIP/400-08280ae0", "trunkdial|SIP/trunk_1/043401xxxx"? ) in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/400-08280ae0", "SIP/trunk_1/043401xxxx") in new stack -- Called trunk_1/043401xxxx [Jan 11 14:33:16] WARNING[2439]: chan_sip.c:11918 handle_response_invite: Received response: "Forbidden&q...

...disallow=all allow=ulaw dtmfmode=inband canreinvite=no reinvite=no context=frombandwidth nat=no [bandwidth.com_outbound] host=216.82.224.202 port=5060 type=peer disallow=all allow=ulaw dtmfmode=rfc2833 nat=no fromuser=11234567890 extensions.conf [globals] ;?irrelevant stuff trunk_1 = Dahdi/g1 trunk_2 = SIP/trunk_2 OUT_2 = SIP/bandwidth.com_outbound ;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it added all the phones when Asterisk calls agents on a Queue. [frombandwidth] ;exten = _+1.,1,Set(GROUP()=SIPGROUP) exten = _+1.,1,NoOp(FromCount=${GRO...

...ultiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1 trunk_2 => SIP/trunk2 trunk_3 => SIP/trunk3 [macro-trunkdial] exten => s,1,Dial(${trunk_1}/${ARG1}) exten => s,2,Hangup() exten => s,102,Dial(${trunk_2}/${ARG1}) exten => s,103,Hangup() exten => s,203,Dial(${trunk_3}/${ARG1}) exten => s,204,Hangup() [from-...

Hello. I don't know if this is a problem, but I was expecting a different behavior. Users, have to dial "0" to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial "012345" there is an error and the call isn't made and the error is "handle_request_invite: Call from 'XXX' to extension '012345'

...07] netsock.c: == Using SIP RTP CoS mark 5 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Sending to 147.135.32.221 : 5060 (NAT) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Using INVITE request as basis request - 31007e-31 at 147.135.32.221 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Found peer 'trunk_1' for '4144038968' from 147.135.32.221:5060 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: <--- Reliably Transmitting (NAT) to 147.135.32.221:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-5...

Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0 allowexternaldomains=no allowexternalinvites=no Do I have to set allowexternalinvites or allowexternaldomains to yes to accept INVITEs from my ITSP? I've already configured the system to...

...rname = abc.com ....where could the asterisk be picking it up from? We have more than 10 such entries (all with same host = provider.sip.com value) and when as INVITE is challenged, the Asterisk does match the correct trunk and seems to send out correct Auth credentials...but not the one below.. [trunk_1] ;register to SP allow = ulaw ;context = test dialformat = ${EXTEN:1} canreinvite = no hasexten = no hasiax = no hassip = yes host = provider.sip.com insecure = very port = 5060 registeriax = no registersip = yes trunkname = test trunkstyle = customvoip username = 3035551122 disallow = gsm,g726,ala...

...hat. In the past we could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf...

...conf: ;! ;! Automatically generated configuration file ;! Filename: extensions.conf (/etc/asterisk/extensions.conf) ;! Generator: Manager ;! Creation Date: Sun Jun 10 16:08:39 2007 ;! [general] static = yes writeprotect = no autofallthrough = yes clearglobalvars = no priorityjumping = no [globals] trunk_1 = SIP/trunk_1 trunk_2 = SIP/trunk_2 trunk_4 = SIP/trunk_4 trunk_6 = IAX2/trunk_6 trunk_7 = IAX2/trunk_7 trunk_8 = SIP/trunk_8 [inbound] exten => s,1,NoOp(Inbound Call CallerID: ${CALLERID(all)}) ;exten => s,n,Answer(SIP/gs486) exten => s,n,Dial(SIP/gs486,30) exten => s,n,Hangup() ;exte...

...rd_vmenu,n,Playback(vm-intro) exten = record_vmenu,n,Record(${var1}) exten = record_vmenu,n,Playback(vm-saved) exten = record_vmenu,n,Playback(vm-goodbye) exten = record_vmenu,n,Hangup exten = play_file,1,Answer exten = play_file,n,Playback(${var1}) exten = play_file,n,Hangup hasbeensetup = Y [DID_trunk_1] include = default [numberplan-custom-1] plancomment = DialPlan1 include = default include = parkedcalls [timebasedrules] *******part of extensions.conf that was added by asterisk-gui (svn)******* *******part of users.conf that was added by asterisk-gui (svn)******* [trunk_1] allow = all contex...

..."no", but that didn't help either. I tried adding some device-specific configuaration to sip.conf, and now my calls are rejected with a status code of "404 not found". This is what I added in sip.conf: [6102] type=friend canreinvite=no host=dynamic dtmfmode=rfc2833 [trunk_1] type=peer host=192.168.20.58 canreinvite=no dtmfmode=inband What am I doing wrong? Hagai. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070521/79d73e4a/attachment.htm

...hat. In the past we could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf...

...[4667]: app_macro.c:337 _macro_exec: Executed application: Gotoif == Auto fallthrough, channel 'SIP/6006-015d0004' status is 'BUSY' -- Executing [9075763441 at DLPN_DialPlan1:1] Macro("SIP/6006-015d0004", "trunkdial-failover-0.3|Zap/g1/075763441|Zap/g2/075763441|trunk_1|trunk_2") in new stack -- Executing [s at macro-trunkdial-failover-0.3:1] Set("SIP/6006-015d0004", "CALLERID(num)=6498287700") in new stack [Jun 6 13:31:41] DEBUG[4825]: app_macro.c:337 _macro_exec: Executed application: Set -- Executing [s at macro-trunkdial-failo...

...ter calling and getting re-routed to the boss I call and it goes through. 3. We cannot recieve DTMF from via;talk, have tried auto, rfc2833, and inband without success with any of them, and yes we had via:talk change their end too. Here is the users.conf entry or the connection to via:talk. [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = <phone number> secret = blablabla trunkname = via:talk ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = <phone number> authuser = <phone number> insecu...

...ng to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI metadata busydetect = yes busycount = 3 busypattern = 480,620 ringtimeout = 8000 answeronpolarityswitch = no hanguponpolarityswitch = no callprogress = no progzone = in usecallerid = yes cidstart = ring pulsedial = no cidsignalling = v23 flash = 750 rxflash = 1250 mailbox =...

Thank you for the reply. On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <danny at debsinc.com>) commented about RE: [asterisk-users] Configuring Softphone: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz > Sent: Wednesday, December 08, 2010 1:27 PM > To: Asterisk