Hey Danny, Yes, modifying the dial command fixed it.
Thanks alot. Best regards, Bobola O. Oke On Wed, Mar 30, 2011 at 6:03 PM, Danny Nicholas <[email protected]> wrote: > What does your Dial command look like? If you are using the ,r option, > Asterisk will generate it’s own ringing noise even on a dead or busy line. > > > ------------------------------ > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Bobola Oke > *Sent:* Wednesday, March 30, 2011 11:36 AM > *To:* Josué Conti > *Cc:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Connecting Asterisk to Siemens Hipath 3750 > > > > 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 > > > > > > > > > > -- > _____________________________________________________________________ > -- 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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