search for: chanunavail

...cro is pretty basic: [macro-STDEXT] exten => s,1,NoOp exten => s,2,Dial(SIP/${ARG1},15,Tt) exten => s,3,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(${ARG1}|u) exten => s-NOANSWER,n,Hangup exten => s-BUSY,1,Voicemail(${ARG1}|b) exten => s-BUSY,n,Hangup exten => s-CHANUNAVAIL,1,Voicemail(${ARG1}|b) exten => s-CHANUNAVAIL,n,Hangup exten => s-CONGESTION,1,Voicemail(${ARG1}|b) exten => s-CONGESTION,n,Hangup My issue is that there are probably only about 50 numbers active at one point in time and the numbers change frequently. I used the 22XX so that I didn...

...es on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be returned from the PSTN GW if the E1 is unavailable, the call goes out over our VOIP provider. Below is an example of a call that goes tires the PSTN GW, gets a DIALSTATUS of CHANUNAVAIL, and then calls our VOIP provider. The problem is that this call was actually busy, and the E1...

...i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) Why, if 5551234 is busy, is DIALSTATUS set to CHANUNAVAIL? Should it not be BUSY? And now with IAX2 (I am using a user:pass that is purposely invalid): exten => 1234,1,Dial(IAX2/user:WRONGpassword@peer/${EXTEN}) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) DIALSTATUS is NOANSWER instead of CHANUNAVAIL ......

...n outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired result would be for DIALSTATUS to get set to CHANUNAVAIL so it would then try any other trunks that I have configured. Further, I don't want simply change the logic to try the IAX on a DIALSTATUS=BUSY because then a truely busy destination number would get re-dialed three times. Thanks for any help, John [macro-dial-pstn-iax] ; exten => s,1,Set...

...allow=all allow=alaw&ulaw nat=no canreinvite=no qualify=yes -Softphones Xlite The PBX can't register to Net2Phone, and no calls are made and this is the log: -- Executing [s at macro-dialout-trunk:20] NoOp("SIP/9004-00000008", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack -- Executing [s at macro-dialout-trunk:21] Goto("SIP/9004-00000008", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:1] Set("SIP/9004-000000...

...n new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/5551212 -- Channel 0/1, span 1 got hangup, cause 28 -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s at macro-dialout-trunk:20] Goto("SIP/801-09b6e498", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:1] GotoIf("SIP/801-09b6e498", "1?noreport") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:3]...

...'s' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config] 2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config] 3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config] 4. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pt:ok) [pbx_config] [pt] 5. Dial(SIP/${ARG1}@${SIPPT},${ARG2},${OUTBDIAL}) [pbx_config] 6. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config] 7. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pstn:ok) [pbx_config] [pstn] 8. Dial(${PSTN...

...STATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL ,CONGESTION,ANSWER) exten => s-CHANUNAVAIL,1,NoOp("CHANUNAVAIL") exten => s-CHANUNAVAIL,2,UserEvent(ChannelUnavailable|Account: ${ACCOUNTCODE}^${CALLERIDNUM}) exten => s-CONGESTION,1,NoOp("CONGESTION") exten => s-CONGESTION,2,UserEvent(Congestion|Account: ${ACCOUNT...

How do you enable call waiting on sip phones? Ive looked and googled and can only find call waiting pstn phones butnot for sip. Is their a way of setting this up within the dailplan?

What is involved in provisioning Asterisk to use a multiline analog service from our local telco? I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk interprets and deals with two incoming calls and/or two outgoing calls? Thanks! jlc

1.6.2.0-rc1. I am having trouble with voice mail intermittently not working correctly on CHANUNAVAIL. (it may happen for other statuses too, haven't checked). Basically here's what happens: -- Executing [1651xxxxxx at mydids:1] Macro("SIP/ipkall-trunk-14838bc8", "phone,1651xxxxxx") in new stack -- Executing [s at macro-phone:1] Dial("SIP/ipkall-trunk-14838bc...

...but now I have stumbled on a behaviour difference I don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would detect this quickly (through the 'qualify' pings), mark the phone as 'Unavailable' and fail immediately with 'CHANUNAVAIL' when dialling this phone. With Asterisk 13 and chan_pjsip qualify still works for determining current phone availability (endpoint shown as 'Unavailable' shortly after disconnecting the cable), but the phone is being dialled like nothing is wrong ? Asterisk sends the INVITE and waits...

...w stack [Oct 10 08:29:10] WARNING[4365]: app_dial.c:1450 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s at macro-dialout-dahdi:5] NoOp("SIP/111-b4e05610", "Dial Status:CHANUNAVAIL") in new stack -- Executing [s at macro-dialout-dahdi:6] Goto("SIP/111-b4e05610", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-dahdi,s-CHANUNAVAIL,1)

...; s-NOANSWER,1,Voicemail(u${ARG1}@default) exten => s-NOANSWER,2,Playback(thank-you-for-calling&goodbye) exten => s-NOANSWER,3,Hangup exten => s-BUSY,1,Voicemail(b${ARG1}@default) exten => s-BUSY,2,Playback(thank-you-for-calling&goodbye) exten => s-BUSY,3,Hangup exten => s-CHANUNAVAIL,1,Voicemail(u${ARG1}@default) exten => s-CHANUNAVAIL,2,Playback(thank-you-for-calling&goodbye) exten => s-CHANUNAVAIL,3,Hangup exten => _s-.,1,Goto(s-NOANSWER,1) [macro-menuexten] exten => s,1,Set(DYNAMIC_FEATURES=automon) exten => s,2,Dial(${ARG2},20,Ttmw) exten => s,3,Goto(...

...g) exten => s-BUSY,2,BackGround(is-curntly-busy) exten => s-BUSY,3,SetGlobalVar(FOUNDME=BUSY) exten => s-BUSY,4,Goto(s-CLEANEXIT,1) ; ; exten => s-CANCEL,1,BackGround(canceled) exten => s-CANCEL,2,SetGlobalVar(FOUNDME=CANCEL) exten => s-BUSY,3,Goto(s-CLEANEXIT,1) ; ; exten => s-CHANUNAVAIL,1,BackGround(channel) exten => s-CHANUNAVAIL,2,BackGround(is-curntly-unavail) exten => s-CHANUNAVAIL,3,SetGlobalVar(FOUNDME=CHANUNAVAIL) exten => s-CHANUNAVAIL,4,Goto(s-CLEANEXIT,1) ; ; exten => s-CONGESTION,1,BackGround(channel) exten => s-CONGESTION,2,BackGround(is-curntly-unavail)...

...nswer: Unable to forward voice or dtmf -- Hungup 'Zap/1-1' [Feb 29 18:16:12] NOTICE[7365]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/1-1' not posted == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/5166-b7b0caa0' status is 'CHANUNAVAIL' 3) When I did "reload chan_zap.so" -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found [Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring switchtype [Feb 29 18:24:31] WARNING[7146]: chan_zap.c...

...724 reset_conf: Failed to reset conferencing on channel 1: Invalid argument -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (0:0/0/0) -- Executing [s at macro-dialout-trunk:21] NoOp("SIP/1990-00000001", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0") in new stack -- Executing [s at macro-dialout-trunk:22] Goto("SIP/1990-00000001", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL at macro-dialout-trunk:1] Set("SIP/1990-0000000...

...Dial("Zap/10-1", "IAX2/iax100|20|tr") in new stack May 25 01:19:21 NOTICE[17629]: app_dial.c:972 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp("Zap/10-1", "CHANUNAVAIL") in new stack -- Executing Goto("Zap/10-1", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-stdfwd3iax-notransfer,s-CHANUNAVAIL,1) -- Executing Goto("Zap/10-1", "s|400") in new stack -- Goto (macro-stdfwd3iax-notransfer,s,400) --...

..._6XX,103,Hangup exten => _04.,1,Macro(dial-mobile,${EXTEN}) [macro-dial-mobile] exten => s,1,SetGlobalVar(NumToDial=${ARG1}) exten => s,2,SetGlobalVar(theCHANNEL=ZAP/3) exten => s,3,Dial(${theCHANNEL}/${NumToDial},60,T) exten => s,4,Goto(s-${DIALSTATUS},1) exten => s,104,Goto(s-CHANUNAVAIL,1) exten => s-CHANUNAVAIL,1,SetGlobalVar(theCHANNEL=SIP/iinet) exten => s-CHANUNAVAIL,2,Playback(voip-warning) exten => s-CHANUNAVAIL,3,Dial(${theCHANNEL}/${NumToDial},60,T) exten => s-CHANUNAVAIL,4,Hangup ; zap/3 context [home] exten => s,1,Answer exten => s,2,SetMusicOnHold(d...

...ueu def, like this: [oneeighty_main] musiconhold = default timeout = 1 joinempty = strict leavewhenempty = strict strategy = rrmemory retry = 0 member => Agent/80014055 member => Agent/80014057 member => Agent/80014052 then the timeout DOES appear to be honored, except that Queue returns CHANUNAVAIL, and does not proceed to the next step in the dialplan. Again, Why? I'm just trying to find the right combination of configuration VOODOO that lets me ring each queue member basically once for 15s. Doug.