Hi below are my configs: pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> legacy pbx analog extensions. my dial plan is like callers dial into asterisk(span1) , hear an IVR option and they are connected to the agents via the legacy pbx (which is in sync with asterisk on span2)....This works perfectly fine until about 200 calls or so...After that time when asterisk tries to dial to the legacy pbx - the call drops with error "All are busy congested at this time" .the same is indicated on asterisk -rvvvvvvvvvv , but the spans are up and active at that time... can anyone throw some light on this ?>>> ZAPTEL.CONFspan=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62>>> ZAPATA.CONFcontext=pri-pstn switchtype=euroisdn pridialplan=local usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group=1 callgroup=1 pickupgroup=1 immediate=yes musiconhold=default signalling = pri_cpe channel => 1-15 channel => 17-31 context=pri-legacy immediate=yes group=2 overlapdial=yes signalling = pri_net channel => 32-46 channel => 48-62>>> EXTENSIONS.CONF ; ; Context PRI-Public ; [pri-pstn] ; include => default ; exten => s,1,Answer exten => s,2,Dial(Zap/g2/1888) ; Dial to legacy pbx and sends the 4 DID digits needed for the legacy pbx exten => s,3,Hangup ; ; Context PRI-legacy ; [pri-legacy] ; include => default ; exten => s,1,Answer exten => s,2,DigitTimeout,2 exten => s,3,ResponseTimeout,2 exten => _X.,1,Dial(Zap/g1/${EXTEN}) exten => _X.,2,Congestion -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081116/d776b40e/attachment.htm
Robert Boardman
2008-Nov-16 11:27 UTC
[asterisk-users] * + Legacy PBX works but strange problem
Sriram wrote:> > Hi > below are my configs: > pstn(e1)--->asterisk (span1)----->legacy pbx(connected via > span2)-----> legacy pbx analog extensions. > > my dial plan is like callers dial into asterisk(span1) , hear an IVR > option and they are connected to the agents via the legacy pbx (which > is in sync with asterisk on span2)....This works perfectly fine until > about 200 calls or so...After that time when asterisk tries to dial to > the legacy pbx - the call drops with error "All are busy congested at > this time" .the same is indicated on asterisk -rvvvvvvvvvv , but the > spans are up and active at that time... can anyone throw some light on > this ? > > >>> ZAPTEL.CONF > | > span=1,0,0,ccs,hdb3,crc4 > span=2,0,0,ccs,hdb3,crc4 > > bchan=1-15 > dchan=16 > bchan=17-31 > > bchan=32-46 > dchan=47 > bchan=48-62 > >>> ZAPATA.CONF > | > | > context=pri-pstn > switchtype=euroisdn > pridialplan=local > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > group=1 > callgroup=1 > pickupgroup=1 > immediate=yes > musiconhold=default > signalling = pri_cpe > channel => 1-15 > channel => 17-31 > > context=pri-legacy > immediate=yes > group=2 > overlapdial=yes > signalling = pri_net > channel => 32-46 > channel => 48-62| > |>>> EXTENSIONS.CONF > | > | > ; > ; Context PRI-Public > ; > [pri-pstn] > ; > include => default > ; > exten => s,1,Answer | > |exten => s,2,Dial(Zap/g2/1888) ; Dial to legacy pbx and sends the 4 DID digits needed for the legacy pbx > exten => s,3,Hangup > ; > ; Context PRI-legacy > ; > [pri-legacy] > ; > include => default > ; > exten => s,1,Answer > exten => s,2,DigitTimeout,2 > exten => s,3,ResponseTimeout,2 > exten => _X.,1,Dial(Zap/g1/${EXTEN}) > exten => _X.,2,Congestion| > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersyou need to pass the clock form the telco to the legacy pbx ie> |span=1,1,0,ccs,hdb3,crc4|Regards Robb
Steve Totaro
2008-Nov-16 13:55 UTC
[asterisk-users] * + Legacy PBX works but strange problem
On Sun, Nov 16, 2008 at 4:28 AM, Sriram <d_r_sriram at hotmail.com> wrote:> > Hi > below are my configs: > pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> > legacy pbx analog extensions. > > my dial plan is like callers dial into asterisk(span1) , hear an IVR option > and they are connected to the agents via the legacy pbx (which is in sync > with asterisk on span2)....This works perfectly fine until about 200 calls > or so...After that time when asterisk tries to dial to the legacy pbx - the > call drops with error "All are busy congested at this time" .the same is > indicated on asterisk -rvvvvvvvvvv , but the spans are up and active at that > time... can anyone throw some light on this ? > > >>> ZAPTEL.CONF > > > span=1,0,0,ccs,hdb3,crc4 > span=2,0,0,ccs,hdb3,crc4 > > bchan=1-15 > dchan=16 > bchan=17-31 > > bchan=32-46 > dchan=47 > bchan=48-62 > >>> ZAPATA.CONF > > > context=pri-pstn > switchtype=euroisdn > pridialplan=local > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > group=1 > callgroup=1 > pickupgroup=1 > immediate=yes > musiconhold=default > signalling = pri_cpe > channel => 1-15 > channel => 17-31 > > context=pri-legacy > immediate=yes > group=2 > overlapdial=yes > signalling = pri_net > channel => 32-46 > channel => 48-62 > > >>> EXTENSIONS.CONF > > > ; > ; Context PRI-Public > ; > [pri-pstn] > ; > include => default > ; > exten => s,1,Answer > > exten => s,2,Dial(Zap/g2/1888) ; Dial to legacy pbx and sends the 4 DID digits needed for the legacy pbx > exten => s,3,Hangup > ; > ; Context PRI-legacy > ; > [pri-legacy] > ; > include => default > ; > exten => s,1,Answer > exten => s,2,DigitTimeout,2 > exten => s,3,ResponseTimeout,2 > exten => _X.,1,Dial(Zap/g1/${EXTEN}) > exten => _X.,2,Congestion > >This is just a suggestion that has worked very well for me in the past when dealing with "Legacy" systems that have only "Analog" phones connected. Ditch the Legacy system and get some form of channel bank. If you want to go SIP to Analog, I have had great luck with Quintum Tenor AX. Since, you have a spare E1 port, you could simply terminate the analog lines to a tried and true channel bank. I have never looked for an E1 channel bank (30 port density) but I would assume they exist. If the Legacy system has proprietary, digital extensions, that complicates things a bit. Special apps running or connected on your Legacy system can usually be migrated and after that bit of growing pain, you have all the flexibility you want to customize. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081116/f30897f9/attachment.htm
hi Robert followed your points - but problem persists...everything goes well for sometime but after that - asterisk is unable to dial the pbx... any more thoughts thanks Sriram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081116/58e07307/attachment.htm
On Sun, Nov 16, 2008 at 9:41 PM, Sriram <d_r_sriram at hotmail.com> wrote:> hi Robert > > followed your points - but problem persists...everything goes well for > sometime but after that - asterisk is unable to dial the pbx... > > any more thoughts > >Post some outputs or logs ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081116/4b5f37b5/attachment.htm
Hi
-Executing [1 at custom-app:1]
Dial("Zap/13-1","ZAP/g2/3901") in new stack
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [1 at custom-app:2] Hangup("Zap/13-1","")
in new stack
== Spawn extension (custom-app,1,2) exited non-zero on 'Zap/13-1'
-- Hungup 'Zap/13-1'
this is the log i get when asterisk is unable to dial to PBX ... but when i
restart * everything again starts working fine until again 100 calls or so..
Rgds
Sriram
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Tony Nichols
2008-Dec-03 20:54 UTC
[asterisk-users] * + Legacy PBX works but strange problem
On Sun, Nov 16, 2008 at 8:55 AM, Steve Totaro < stotaro at totarotechnologies.com> wrote:> > > On Sun, Nov 16, 2008 at 4:28 AM, Sriram <d_r_sriram at hotmail.com> wrote: > >> >> Hi >> below are my configs: >> pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> >> legacy pbx analog extensions. >> >> my dial plan is like callers dial into asterisk(span1) , hear an IVR >> option and they are connected to the agents via the legacy pbx (which is in >> sync with asterisk on span2)....This works perfectly fine until about 200 >> calls or so...After that time when asterisk tries to dial to the legacy pbx >> - the call drops with error "All are busy congested at this time" .the same >> is indicated on asterisk -rvvvvvvvvvv , but the spans are up and active at >> that time... can anyone throw some light on this ? >> >> >>> ZAPTEL.CONF >> >> >> span=1,0,0,ccs,hdb3,crc4 >> span=2,0,0,ccs,hdb3,crc4 >> >> bchan=1-15 >> dchan=16 >> bchan=17-31 >> >> bchan=32-46 >> dchan=47 >> bchan=48-62 >> >>> ZAPATA.CONF >> >> >> context=pri-pstn >> switchtype=euroisdn >> pridialplan=local >> usecallerid=yes >> hidecallerid=no >> callwaiting=yes >> usecallingpres=yes >> callwaitingcallerid=yes >> threewaycalling=yes >> transfer=yes >> cancallforward=yes >> callreturn=yes >> group=1 >> callgroup=1 >> pickupgroup=1 >> immediate=yes >> musiconhold=default >> signalling = pri_cpe >> channel => 1-15 >> channel => 17-31 >> >> context=pri-legacy >> immediate=yes >> group=2 >> overlapdial=yes >> signalling = pri_net >> channel => 32-46 >> channel => 48-62 >> >> >>> EXTENSIONS.CONF >> >> >> ; >> ; Context PRI-Public >> ; >> [pri-pstn] >> ; >> include => default >> ; >> exten => s,1,Answer >> >> exten => s,2,Dial(Zap/g2/1888) ; Dial to legacy pbx and sends the 4 DID digits needed for the legacy pbx >> exten => s,3,Hangup >> ; >> ; Context PRI-legacy >> ; >> [pri-legacy] >> ; >> include => default >> ; >> exten => s,1,Answer >> exten => s,2,DigitTimeout,2 >> exten => s,3,ResponseTimeout,2 >> exten => _X.,1,Dial(Zap/g1/${EXTEN}) >> exten => _X.,2,Congestion >> >> > This is just a suggestion that has worked very well for me in the past when > dealing with "Legacy" systems that have only "Analog" phones connected. > > Ditch the Legacy system and get some form of channel bank. If you want to > go SIP to Analog, I have had great luck with Quintum Tenor AX. Since, you > have a spare E1 port, you could simply terminate the analog lines to a tried > and true channel bank. I have never looked for an E1 channel bank (30 port > density) but I would assume they exist. > > If the Legacy system has proprietary, digital extensions, that complicates > things a bit. > > Special apps running or connected on your Legacy system can usually be > migrated and after that bit of growing pain, you have all the flexibility > you want to customize. > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >I have noticed when connecting our legacy system to asterisk, the option " overlapdial=yes caused issues with only certain exchanges... and would appear randomly. It seems to add a "pause" of some 4 sec. when dialing. This would give you the "busy" error. -- A.G. (Tony) Nichols I.S. Manager -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081203/1727522e/attachment.htm