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
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
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 'xxxxxx' does not exist. Rejecting call on channel 0/31, span 2 That comes up each time a call is attempted from the siemens, the xxxxxx 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" <james@stemnetworks.co.uk> wrote:> 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" <jjones@danrj.com> 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
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 don?t know how to get the next bit coded. Any help appreciated ! Thanks James On 6/8/06 08:54, "Jon Farmer" <jon@bctech.co.uk> 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 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060806/10bb76b9/attachment.htm
Hi James,>James wrote;>When I hit 9 on the siemens it does not get a dial tone from asterisk, Iassume 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060806/5388527e/attachment.htm
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)" <chan@isysnetsolution.com> 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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060807/c7487e90/attachment.htm