Hi guys Thanks alot for the support. I have successfully connected the HiPath3750 to the E1 lines and everything is working fine with the appropriate dial plans. I used Josue's config and the info I got from here http://www.voip-info.org/wiki/view/Siemens+Hicom
Well, not everything is working fine though.. The asterisk server seems to 'generate' the ringing tones as opposed to using the tones from the various other external numbers that I am calling. For example, if I call a phone number that is switched off, it rings for a while and then I get a service unavailable message on the IP phones. What can I do to get the normal "the number you have dialed is switched off". I am in Nigeria if that information is useful in this situation. Thanks. Bobola 2011/3/16 Bobola Oke <[email protected]> > Hey Josue, > > Thanks alot. I will be expecting the configuration samples. From your > response, I guess QSIG would be better for more functionality between the > two PBXs then.. > > Yes, this is my first implementation of asterisk and the support I have had > from the mailing lists (some just by searching the archives) has been > nothing short of wonderful. Thanks guys. > > Hoping to hear from you soon. > > Best regards, > > Bobola O. Oke > > > 2011/3/15 Josué Conti <[email protected]> > >> Hello Bobola, thanks for your response. >> So, I'm using Euroisdn with a Siemens HiPath 3750 and Qsig with Siemens >> HiPath 4000. >> Because we don't need to "facility enable" in this case (HiPath 3750) just >> ANI interchange between user's, ok? >> In another response I was send to you a configurations sample for Asterisk >> and Siemens may you look this? >> One more time, best regards and good luck in your project. >> If you need please contact us. >> >> Josue >> >> >> 2011/3/14 Bobola Oke <[email protected]> >> >>> Thanks guys, >>> >>> I got the layer1 link up. >>> >>> Edwin, I will make a cable from this link that you have posted and see if >>> that also works. Presently, I just did a 'manual' connect of the ends to get >>> the layer1 up. >>> >>> Josue, many thanks for your response. Searching through this list >>> archives, I see that you must have done alot of integrating asterisk with >>> Siemens PBX. >>> >>> Guys, what do you advise I use for the upper layer protocols, QSIG or >>> EuroISDN, to connect the asterisk PBX and the Siemens PBX? What are the pros >>> and cons of using either protocol. Working sample configuration files are >>> highly appreciated + what the PBX guy has to configure on the Siemens side. >>> >>> Thanks alot. >>> >>> >>> >>> On Fri, Mar 11, 2011 at 1:17 AM, Edwin Lam >>> <[email protected]>wrote: >>> >>>> On 3/10/11 6:43 AM, Bobola Oke wrote: >>>> >>>>> >>>>> The telco has a DB9 terminated interface straight to the PBX and I >>>>> cannot make >>>>> sense out of the interface for the PBX. What kind of interface is this? >>>>> How do I >>>>> connect the RJ48 of the PRI cards to make this whole setting work. >>>>> >>>> >>>> searching through this list's archive and found this: >>>> >>>> http://lists.digium.com/pipermail/asterisk-users/2006-December/174258.html >>>> >>>> >>>> -- >>>> Edwin Lam <[email protected]> >>>> Systems Engineer, OfficeWyze, Inc. >>>> Ph: <%2B1%20415%20439%204988> >>>> <%2B1%20415%20439%204988><%2B1%20415%20439%204988>+1 >>>> 415 439 4988 Fax: <%2B1%20415%20283%203370> >>>> <%2B1%20415%20283%203370><%2B1%20415%20283%203370>+1 >>>> 415 283 3370 >>>> http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20 >>>> >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>> http://www.asterisk.org/hello >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> >> >
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