Re: [asterisk-users] Asterisk and Siemens Legacy PBX
James Arscott [EMAIL PROTECTED] writes: Hi Small progress, though combining the suggest below, enabling overlapdial and a few other things I have got the following : When you hit 9 on the simenes, you hear a dial tone. As soon as you hit another number to start dialling it complains with some generic error on the siemens handset. What I see from asterisk at the same time [...] -- Starting simple switch on 'Zap/62-1' -- Accepting overlap call from '697000' to 'unspecified' on channel 0/31, span 2 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: INFORMATION (123) [70 02 81 39]LI Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9' ] -- Processing IE 112 (cs0, Called Party Number) Hmm? That's not overlap dialing. You get the complete called number in one single Message. -- Processing IE 112 (cs0, Called Party Number) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Overlap Receiving, peerstate Overlap sending Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81]LI Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] ... and Asterisk's answer is: Unallocated number. It seems your Siemens PBX doesn't do it right. We had some issues with other PBXs and Asterisk when Asterisk was the NET-side. Try to reverse the roles, so that Siemens is NET and Asterisk CPE. That helped here with an Alcatel. cu, Wolfgang ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Title: Re: [asterisk-users] Asterisk and Siemens Legacy PBX Hi, thanks for this, something I had totally looked over because I saw the span 2 had gone from RED to OK. Zapata.conf - [channels] language=en ; Default context context=inbound-from-pstn switchtype=euroisdn signalling=pri_cpe rxwink=300 usecallerid=yes idecallerid=no callwaiting=no ;restrictcid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=no callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=9 pickupgroup=9 immediate=no musiconhold=default busydetect=no callprogress=no channel=1-15,17-31 context=inbound-from-siemens signalling=pri_net switchtype=euroisdn priindication=outofband group=2 channel=32-46,48-62 Zaptel.conf - loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 I now assume framing and timing are not right. Any help would be appreciated ! :) James On 7/8/06 03:48, (AstATN) [EMAIL PROTECTED] wrote: Hi James, James wrote; When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one. I might be wrong; My question is, are you sure your ISDN ( Asterisk span to Siemens ) is up logically? ISDN is no tone given, dial tone is actually produced by Legacy Side, when L1, L2 and L3 signals is up (eg coding, framing and timing), Legacy PBX will automatically make it self ready, and simulating the dial tone when user hit 9 to call out. I did try with Alcatel and Ericsson MD machine; both are simulating dial tone once L2 and L3 are working properly, so I assume that this is the Europe PBX standard. As fall as ISDN Legacy PBX is concern, it will throw out the digits if nothing wrong with the link. If possible, share with your Zapata.conf setting, may be group of us can help. Tq James wrote; Hi, I just realised I think I have missed a step Asterisk is not matching the extension from the siemens because the siemens has not even sent one yet, it is still waiting for a dial tone. When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one(dur!) How should I tell asterisk how to handle this, I have defined it a context and I know its making it this far, but I don9t know how to get the next bit coded. Any help appreciated ! Thanks James ___ --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 ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
James Arscott [EMAIL PROTECTED] writes: Asterisk is not matching the extension from the siemens because the siemens has not even sent one yet, it is still waiting for a dial tone. When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one(dur!) How should I tell asterisk how to handle this, I have defined it a context and I know its making it this far, but I don¹t know how to get the next bit coded. Any help appreciated ! Have you turned on overlapdial in zapata.conf? Most PBX's send numbers in overlap mode, yours does it also. Next, you should add something like this to your Siemens-context: exten = s,1,WaitExten(2) This waits for the numbers which should now come from the Siemens side. hth, Wolfgang ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Title: Re: [asterisk-users] Asterisk and Siemens Legacy PBX Hi, thanks I will try both of those, I had tried overlapdial before with no luck, but maybe its a combination of things I need. My concern is still over if L2 and L3 are up on my ISDN between the asterisk and siemens, I do my settings look right ? I thought my timing on span 2 maybe incorrect ? Thanks James On 7/8/06 09:54, Wolfgang Zweimueller [EMAIL PROTECTED] wrote: James Arscott [EMAIL PROTECTED] writes: Asterisk is not matching the extension from the siemens because the siemens has not even sent one yet, it is still waiting for a dial tone. When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one(dur!) How should I tell asterisk how to handle this, I have defined it a context and I know its making it this far, but I dont know how to get the next bit coded. Any help appreciated ! Have you turned on overlapdial in zapata.conf? Most PBX's send numbers in overlap mode, yours does it also. Next, you should add something like this to your Siemens-context: exten = s,1,WaitExten(2) This waits for the numbers which should now come from the Siemens side. hth, Wolfgang ___ --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 ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
James Arscott [EMAIL PROTECTED] writes: My concern is still over if L2 and L3 are Œup¹ on my ISDN between the asterisk and siemens, I do my settings look right ? I thought my timing on span 2 maybe incorrect ? If you do a pri show span 2 you get a short info about the state of the span. Next, you can try pri debug span 2 and watch the ISDN conversations. This should give you the info if your link is set up correctly. cu, Wolfgang ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Title: Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Hi
Small progress, though combining the suggest below, enabling overlapdial and a few other things I have got the following :
When you hit 9 on the simenes, you hear a dial tone. As soon as you hit another number to start dialling it complains with some generic error on the siemens handset. What I see from asterisk at the same time
-- Starting simple switch on 'Zap/62-1'
-- Accepting overlap call from '697000' to 'unspecified' on channel 0/31, span 2
-- Hungup 'Zap/62-1'
corp-phones-1*CLI pri show span 2
Primary D-channel: 47
Status: Provisioned, Up, Active
Switchtype: EuroISDN
Type: Network
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: -1
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3
And the debug from the pri span 2
Protocol Discriminator: Q.931 (8) len=29
Call Ref: len= 2 (reference 1/0x1) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Ext: 1 User information layer 1: A-Law (35)
[18 03 a1 83 9f]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 31 ]
[6c 08 01 80 36 39 37 30 30 30]
Calling Number (len=10) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number not screened (0) '697000' ]
[7d 02 91 81]LI
IE: High-layer Compatibility (len = 4)
-- Making new call for cr 1
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 125 (cs0, High-layer Compatibility)
Protocol Discriminator: Q.931 (8) len=14
Call Ref: len= 2 (reference 1/0x1) (Terminator)
Message type: SETUP ACKNOWLEDGE (13)
[18 03 a9 83 9f]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 31 ]
[1e 02 81 82]LI
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1)
Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ]
-- Starting simple switch on 'Zap/62-1'
-- Accepting overlap call from '697000' to 'unspecified' on channel 0/31, span 2
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 1/0x1) (Originator)
Message type: INFORMATION (123)
[70 02 81 39]LI
Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9' ]
-- Processing IE 112 (cs0, Called Party Number)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Overlap Receiving, peerstate Overlap sending
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 1/0x1) (Terminator)
Message type: RELEASE COMPLETE (90)
[08 02 81 81]LI
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1)
Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/62-1'
My config is currently as follows :
Extensions context :
[inbound-from-siemens]
ignorepat = 9
exten = s,1,WaitExten(10)
exten = _X.,2,Dial(Zap/g1/${EXTEN})
Zapata.conf :
[channels]
language=en
; Default context
context=inbound-from-pstn
switchtype=euroisdn
signalling=pri_cpe
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
;restrictcid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=no
cancallforward=no
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
rxgain=0.0
txgain=0.0
callgroup=9
pickupgroup=9
immediate=no
musiconhold=default
busydetect=no
callprogress=no
group=1
channel=1-15,17-31
context=inbound-from-siemens
signalling=pri_net
switchtype=euroisdn
priindication=outofband
overlapdial=yes
pridialplan=local
immediate=no
group=2
channel=32-46,48-62
Zaptel.conf
loadzone=uk
defaultzone=uk
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,2,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
Cheers
James
On 7/8/06 11:06, Wolfgang Zweimueller [EMAIL PROTECTED] wrote:
James Arscott [EMAIL PROTECTED] writes:
My concern is still over if L2 and L3 are up on my ISDN between the
asterisk and siemens, I do my settings look right ? I thought my timing on
span 2 maybe incorrect ?
If you do a pri show span 2 you get a short info about the state of
the span.
Next, you can try pri debug span 2 and watch the ISDN
conversations. This should give you the info if your link is set up
correctly.
cu,
Wolfgang
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
James Arscott wrote: I also tried just using s , this again did not work. I assumed the ‘Extension ‘’ in context’ part of my debug meant that the siemens is not sending, or asterisk can’t work out, what extension is being sent If that makes sense It means that whatever context you have defined for the Zap span can't find a extension with the number the siemens is dialling. Look at the zap span config and see what context is defined and then make sure that context has the right extenensions defined. Also to help me get my head around this, the ‘extension’ referred to that should be being sent from the siemens, is this going to be the number the siemens is dialing, if not, how do I get ‘access’ to that number? Yes its the number the siemens is dialling. My goal is to just allow the siemens to make any call it wants via the span 1 on the asterisk box, which is connected to a ‘real’ ISDN PRI. This is a everyday use for Asterisk :-) -- Jon Farmer Telford, Shropshire, UK ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Title: Re: [asterisk-users] Asterisk and Siemens Legacy PBX Hi, I just realised I think I have missed a step Asterisk is not matching the extension from the siemens because the siemens has not even sent one yet, it is still waiting for a dial tone. When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one(dur!) How should I tell asterisk how to handle this, I have defined it a context and I know its making it this far, but I dont know how to get the next bit coded. Any help appreciated ! Thanks James On 6/8/06 08:54, Jon Farmer [EMAIL PROTECTED] wrote: James Arscott wrote: I also tried just using s , this again did not work. I assumed the Extension in context part of my debug meant that the siemens is not sending, or asterisk cant work out, what extension is being sent If that makes sense It means that whatever context you have defined for the Zap span can't find a extension with the number the siemens is dialling. Look at the zap span config and see what context is defined and then make sure that context has the right extenensions defined. Also to help me get my head around this, the extension referred to that should be being sent from the siemens, is this going to be the number the siemens is dialing, if not, how do I get access to that number? Yes its the number the siemens is dialling. My goal is to just allow the siemens to make any call it wants via the span 1 on the asterisk box, which is connected to a real ISDN PRI. This is a everyday use for Asterisk :-) -- Jon Farmer Telford, Shropshire, UK ___ --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 ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Hi James, James wrote; When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one. I might be wrong; My question is, are you sure your ISDN ( Asterisk span to Siemens ) is up logically? ISDN is no tone given, dial tone is actually produced by Legacy Side, when L1, L2 and L3 signals is up (eg coding, framing and timing), Legacy PBX will automatically make it self ready, and simulating the dial tone when user hit 9 to call out. I did try with Alcatel and Ericsson MD machine; both are simulating dial tone once L2 and L3 are working properly, so I assume that this is the Europe PBX standard. As fall as ISDN Legacy PBX is concern, it will throw out the digits if nothing wrong with the link. If possible, share with your Zapata.conf setting, may be group of us can help. Tq James wrote; Hi, I just realised I think I have missed a step Asterisk is not matching the extension from the siemens because the siemens has not even sent one yet, it is still waiting for a dial tone. When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one(dur!) How should I tell asterisk how to handle this, I have defined it a context and I know its making it this far, but I don9t know how to get the next bit coded. Any help appreciated ! Thanks James ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Hi, thanks to the original poster, I redid all the cabling and immediately got the span to go OK between asterisk and the siemens legacy PBX. Only problem now is working out how to handle the calls from the siemens Worth pointing out at this stage I have no access to the siemens configuration, so I could be shooting blind. I put span2 (which is connected to the siemens) into its own context (inbound-from-siemens) and then tried to few simple attempts at receiving¹ the calls that the siemens is trying to make. However whatever I put all I get via the asterisk console is : -- Extension '' in context 'inbound-from-siemens' from 'xx' does not exist. Rejecting call on channel 0/31, span 2 That comes up each time a call is attempted from the siemens, the xx shows as whichever direct dial number tried to dial out on the siemens, which I initially was pleased to see, however I am now stumped at how I should try to get asterisk to deal with these calls, am I barking up the wrong tree ? Thanks James On 4/8/06 23:03, James Arscott [EMAIL PROTECTED] wrote: Hi, thanks I will try this tomorrow morning when the legacy PBX can be taken offline¹ for a few hours, any suggestions on specific asterisk configuration options that I may have missed to achieve this ? I am hoping its just the cable I am using Cheers James On 4/8/06 22:30, Jerry Jones [EMAIL PROTECTED] wrote: probably need a crossed t1 cable 1-4 2-5 On Aug 4, 2006, at 4:20 PM, James Arscott wrote: Hi, this is my first post, so go easy on me ! Sorry if this has been covered before, I could not find an answer that helped me. I am trying to achieve the following : Telco ISDN30e PRI - Asterisk with TE210P - Siemens HiPath PBX The siemens is a legacy PBX and I am not 100% of the modules etc inside it, it is being used in production at the moment and we have a need to put the Asterisk pbx as a gateway in between the ISDN and the Siemens. Ultimately this will help us move people from the legacy PBX to full SIP phones. We have many Asterisk PBX's working well using the TE210P + ISDN30e PRI, but I am unsure how to get the legacy PBX working with the 2nd span of the TE210P. I *assumed* that all I had to do was configure the 2nd span with pri_net and leave span 1 as pri_cpe and that would do the job, but when I do this and plug the siemens into span 2 I get a RED alarm on the span 2 and that's about it. Any tips on the most likely configuration that will work ? What configuration of CAT5 should I be using to connect the legacy PBX to span 2 ? Straight, crossed, etc. Many thanks in advanced ! James ___ --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 ___ --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 ___ --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 ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
James Arscott wrote: Hi, thanks to the original poster, I redid all the cabling and immediately got the span to go OK between asterisk and the siemens legacy PBX. Only problem now is working out how to handle the calls from the siemens Worth pointing out at this stage I have no access to the siemens configuration, so I could be shooting blind. I put span2 (which is connected to the siemens) into its own context (inbound-from-siemens) and then tried to few simple attempts at Œreceiving¹ the calls that the siemens is trying to make. However whatever I put all I get via the asterisk console is : -- Extension '' in context 'inbound-from-siemens' from 'xx' does not exist. Rejecting call on channel 0/31, span 2 That comes up each time a call is attempted from the siemens, the xx shows as whichever direct dial number tried to dial out on the siemens, which I initially was pleased to see, however I am now stumped at how I should try to get asterisk to deal with these calls, am I barking up the wrong tree ? No you are slowing barking up the right tree :-) The call is getting accepted by Asterisk in the context inbound-from-siemens. However it can't work out what to do with the call. You need to match the xx number with a extension number which is in the inbound-from-siemens context or another context included in it. For instance if the xxx number is 123456 you could use. [inbound-from-siemens] exten = 123456,1,Dial(SIP/101) to dial SIP phone 101 or if the numbers from the siemens follow a pattern ie they all start with 12 then you could use exten = _12,1,Dial(SIP/101) If you check the extensions.conf page at www.voip-info.org/wiki you will see loads of examples on how to construct a dialplan HTH Jon -- Jon Farmer Telford, Shropshire, UK ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Title: Re: [asterisk-users] Asterisk and Siemens Legacy PBX Hi Jon, thanks for the reply, good to know I am on the right track! I have actually already tried your suggestion, but using : _. To try and match what the siemens was sending, but this did not work, I know this is not advisable in production, but I was just trying to get something working before I came up with the proper solution. I also tried just using s , this again did not work. I assumed the Extension in context part of my debug meant that the siemens is not sending, or asterisk cant work out, what extension is being sent If that makes sense Also to help me get my head around this, the extension referred to that should be being sent from the siemens, is this going to be the number the siemens is dialing, if not, how do I get access to that number? My goal is to just allow the siemens to make any call it wants via the span 1 on the asterisk box, which is connected to a real ISDN PRI. James On 5/8/06 20:30, Jon Farmer [EMAIL PROTECTED] wrote: James Arscott wrote: Hi, thanks to the original poster, I redid all the cabling and immediately got the span to go OK between asterisk and the siemens legacy PBX. Only problem now is working out how to handle the calls from the siemens Worth pointing out at this stage I have no access to the siemens configuration, so I could be shooting blind. I put span2 (which is connected to the siemens) into its own context (inbound-from-siemens) and then tried to few simple attempts at receiving the calls that the siemens is trying to make. However whatever I put all I get via the asterisk console is : -- Extension '' in context 'inbound-from-siemens' from 'xx' does not exist. Rejecting call on channel 0/31, span 2 That comes up each time a call is attempted from the siemens, the xx shows as whichever direct dial number tried to dial out on the siemens, which I initially was pleased to see, however I am now stumped at how I should try to get asterisk to deal with these calls, am I barking up the wrong tree ? No you are slowing barking up the right tree :-) The call is getting accepted by Asterisk in the context inbound-from-siemens. However it can't work out what to do with the call. You need to match the xx number with a extension number which is in the inbound-from-siemens context or another context included in it. For instance if the xxx number is 123456 you could use. [inbound-from-siemens] exten = 123456,1,Dial(SIP/101) to dial SIP phone 101 or if the numbers from the siemens follow a pattern ie they all start with 12 then you could use exten = _12,1,Dial(SIP/101) If you check the extensions.conf page at www.voip-info.org/wiki you will see loads of examples on how to construct a dialplan HTH Jon -- Jon Farmer Telford, Shropshire, UK ___ --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 ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
probably need a crossed t1 cable 1-4 2-5 On Aug 4, 2006, at 4:20 PM, James Arscott wrote: Hi, this is my first post, so go easy on me ! Sorry if this has been covered before, I could not find an answer that helped me. I am trying to achieve the following : Telco ISDN30e PRI - Asterisk with TE210P - Siemens HiPath PBX The siemens is a legacy PBX and I am not 100% of the modules etc inside it, it is being used in production at the moment and we have a need to put the Asterisk pbx as a gateway in between the ISDN and the Siemens. Ultimately this will help us move people from the legacy PBX to full SIP phones. We have many Asterisk PBX's working well using the TE210P + ISDN30e PRI, but I am unsure how to get the legacy PBX working with the 2nd span of the TE210P. I *assumed* that all I had to do was configure the 2nd span with pri_net and leave span 1 as pri_cpe and that would do the job, but when I do this and plug the siemens into span 2 I get a RED alarm on the span 2 and that's about it. Any tips on the most likely configuration that will work ? What configuration of CAT5 should I be using to connect the legacy PBX to span 2 ? Straight, crossed, etc. Many thanks in advanced ! James ___ --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 ___ --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
Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Title: Re: [asterisk-users] Asterisk and Siemens Legacy PBX Hi, thanks I will try this tomorrow morning when the legacy PBX can be taken offline for a few hours, any suggestions on specific asterisk configuration options that I may have missed to achieve this ? I am hoping its just the cable I am using Cheers James On 4/8/06 22:30, Jerry Jones [EMAIL PROTECTED] wrote: probably need a crossed t1 cable 1-4 2-5 On Aug 4, 2006, at 4:20 PM, James Arscott wrote: Hi, this is my first post, so go easy on me ! Sorry if this has been covered before, I could not find an answer that helped me. I am trying to achieve the following : Telco ISDN30e PRI - Asterisk with TE210P - Siemens HiPath PBX The siemens is a legacy PBX and I am not 100% of the modules etc inside it, it is being used in production at the moment and we have a need to put the Asterisk pbx as a gateway in between the ISDN and the Siemens. Ultimately this will help us move people from the legacy PBX to full SIP phones. We have many Asterisk PBX's working well using the TE210P + ISDN30e PRI, but I am unsure how to get the legacy PBX working with the 2nd span of the TE210P. I *assumed* that all I had to do was configure the 2nd span with pri_net and leave span 1 as pri_cpe and that would do the job, but when I do this and plug the siemens into span 2 I get a RED alarm on the span 2 and that's about it. Any tips on the most likely configuration that will work ? What configuration of CAT5 should I be using to connect the legacy PBX to span 2 ? Straight, crossed, etc. Many thanks in advanced ! James ___ --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 ___ --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 ___ --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
