Hi All, I setup auto dial on my asterisk server. The problem is asterisk does not wait for called party to answer the call but proceed to process the extension specifed in my .call file My sample call file : hannel: local/0124787924@outbound-reminder MaxRetries: 5 RetryTime: 300 WaitTime: 40 Account: Reminder context: remindem extension: s priority: 1 Set: MSG=0135.20070601.0124787924 Set: APPTDT=20070601 Set: APPTTIME=0135 Set: APPTPHONE=0124787924 Set: CALLATTEMPTS=5 Set: CALLDELAY=300 My outbound-reminder context: [outbound-reminder] exten => _01N.,1,Dial(Zap/g1/${EXTEN},20) My remindem context : [remindem] exten => s,1,Answer() exten => s,2,Wait(2) exten => s,3,Playback(custom/reminder5) Once asterisk start to execute .call file, my handset rings but the console shows Playback(custom/reminder5) Below shows some console messages : -- Attempting call on local/0124787924@outbound-reminder for s@remindem:1 (Retry 1) -- Executing [0124787924@outbound-reminder:1] Dial("Local/0124787924@outbound-reminder-5223,2", "Zap/g1/0124787924|20") in new stack -- Called g1/0124787924 -- Zap/3-1 answered Local/0124787924@outbound-reminder-5223,2 [Jun 1 02:50:09] WARNING[8651]: pbx.c:4976 ast_pbx_outgoing_exten: Local/0124787924@outbound-reminder-5223,1 already has a call record?? > Channel Local/0124787924@outbound-reminder-5223,1 was answered. -- Executing [s@remindem:1] Answer("Local/0124787924@outbound-reminder-5223,1", "") in new stack -- Executing [s@remindem:2] Wait("Local/0124787924@outbound-reminder-5223,1", "2") in new stack == Spawn extension (outbound-reminder, 0124787924, 1) exited non-zero on 'Local/0124787924@outbound-reminder-5223,2' -- Executing [s@remindem:3] Playback("Zap/3-1", "custom/reminder5") in new stack I really appreciate if some one can help me Regards ASLAY
aslay@pinwee.com.my wrote:> Hi All, > > I setup auto dial on my asterisk server. The problem > is asterisk does not wait for called party to answer > the call but proceed to process the extension specifed > in my .call file > > My sample call file : > > hannel: local/0124787924@outbound-reminder > MaxRetries: 5 > RetryTime: 300 > WaitTime: 40 > Account: Reminder > context: remindem > extension: s > priority: 1 > Set: MSG=0135.20070601.0124787924 > Set: APPTDT=20070601 > Set: APPTTIME=0135 > Set: APPTPHONE=0124787924 > Set: CALLATTEMPTS=5 > Set: CALLDELAY=300 > > My outbound-reminder context: > > [outbound-reminder] > exten => _01N.,1,Dial(Zap/g1/${EXTEN},20) > > My remindem context : > > [remindem] > exten => s,1,Answer() > exten => s,2,Wait(2) > exten => s,3,Playback(custom/reminder5) > > Once asterisk start to execute .call file, my handset > rings but the console shows Playback(custom/reminder5) > >I believe that it is because you are using zap lines to dialout. Zap lines are considered answered almost immediately. The believe digital and VoIP channels on the other hand have the call supervision that can distinguish when an answer is made. Any kind of dialout like that, I just use my sip service provider. -- Warm Regards, Lee
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Lee Jenkins > Sent: Saturday, June 02, 2007 11:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Auto Dial Problem > > aslay@pinwee.com.my wrote: > > Hi All, > > > > I setup auto dial on my asterisk server. The problem > > is asterisk does not wait for called party to answer > > the call but proceed to process the extension specifed > > in my .call file > > > > My sample call file : > > > > hannel: local/0124787924@outbound-reminder > > MaxRetries: 5 > > RetryTime: 300 > > WaitTime: 40 > > Account: Reminder > > context: remindem > > extension: s > > priority: 1 > > Set: MSG=0135.20070601.0124787924 > > Set: APPTDT=20070601 > > Set: APPTTIME=0135 > > Set: APPTPHONE=0124787924 > > Set: CALLATTEMPTS=5 > > Set: CALLDELAY=300 > > > > My outbound-reminder context: > > > > [outbound-reminder] > > exten => _01N.,1,Dial(Zap/g1/${EXTEN},20) > > > > My remindem context : > > > > [remindem] > > exten => s,1,Answer() > > exten => s,2,Wait(2) > > exten => s,3,Playback(custom/reminder5) > > > > Once asterisk start to execute .call file, my handset > > rings but the console shows Playback(custom/reminder5) > > > > > > I believe that it is because you are using zap lines to dialout. Zap > lines are considered answered almost immediately. The believe digital > and VoIP channels on the other hand have the call supervision that can > distinguish when an answer is made. > > Any kind of dialout like that, I just use my sip service provider. > > -- > > Warm Regards, > > Lee >This may be true with analog zap channels but not T1 PRIs. Additionally, some VoIP providers "answer" the call prior to initiating the second leg of the call. Who is your provider that does not give you an answer until the call is really answered? Last I checked, IAX.cc (now Vitelity) was giving me answered immediately. I am not sure that is the case anymore. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB
Hi,> I setup auto dial on my asterisk server. The problem > is asterisk does not wait for called party to answer > the call but proceed to process the extension specifed > in my .call fileNo problem with Auto Call> exten => _01N.,1,Dial(Zap/g1/${EXTEN},20)the problem with zap channel try callprogress with yes in zapata.conf It may cause another problem, "after remote party has picked up the call" and asterisk still does not know it. and in ringing status. if your dial plan work fine now, then no need to change rxgain. otherwise. Just Increase your rxgain value. try with different values and choose best one. if rxgain greater then desired value > ?? you my receive invalid report that remote party has picked up. if rxgain less then desired value < ?? you my receive invalid ringing report after call is answered. so adjust it according your requirement and also check noise and quality your PSTN lines. Regards Nasir Iqbal ICT Innovations
Hi,> I setup auto dial on my asterisk server. The problem > is asterisk does not wait for called party to answer > the call but proceed to process the extension specifed > in my .call fileNo problem with Auto Call> exten => _01N.,1,Dial(Zap/g1/${EXTEN},20)the problem with zap channel try callprogress with yes in zapata.conf It may cause another problem, "after remote party has picked up the call" and asterisk still does not know it. and in ringing status. if your dial plan work fine now, then no need to change rxgain. otherwise. Just Increase your rxgain value. try with different values and choose best one. if rxgain greater then desired value > ?? you my receive invalid report that remote party has picked up. if rxgain less then desired value < ?? you my receive invalid ringing report after call is answered. so adjust it according your requirement and also check noise and quality your PSTN lines. Regards Nasir Iqbal ICT Innovations
Dear Sir, Thank you very much ASLAY ----- Original Message ----- From: "Nasir Iqbal" <nasir@ictinnovations.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, June 03, 2007 5:43 PM Subject: Re: [asterisk-users] Auto Dial Problem> Hi, > > > I setup auto dial on my asterisk server. The problem > > is asterisk does not wait for called party to answer > > the call but proceed to process the extension specifed > > in my .call file > > No problem with Auto Call > > > exten => _01N.,1,Dial(Zap/g1/${EXTEN},20) > > the problem with zap channel > > try callprogress with yes in zapata.conf > > It may cause another problem, "after remote party has picked up the > call" and asterisk still does not know it. and in ringing status. > > if your dial plan work fine now, then no need to change rxgain. > otherwise. > > Just Increase your rxgain value. try with different values and choose > best one. > > if rxgain greater then desired value > ?? you my receive invalid report > that remote party has picked up. > > if rxgain less then desired value < ?? you my receive invalid ringing > report after call is answered. > > so adjust it according your requirement and also check noise and quality > your PSTN lines. > > > Regards > > Nasir Iqbal > > ICT Innovations > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users