<div>Hi guys</div><div>Thanks alot for the support.</div><div> </div><div>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 <a href="http://www.voip-info.org/wiki/view/Siemens+Hicom">http://www.voip-info.org/wiki/view/Siemens+Hicom</a></div>
<div> </div><div>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.</div>
<div> </div><div>Thanks.</div><div> </div><div>Bobola</div><div> </div><div>2011/3/16 Bobola Oke <span dir="ltr"><<a href="mailto:okebobola@gmail.com">okebobola@gmail.com</a>></span><br></div><div class="gmail_quote">
<blockquote style="margin: 0px 0px 0px 0.8ex; padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="gmail_quote">Hey Josue,<br><br>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..<br>
<br>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.<br>
<br>Hoping to hear from you soon.<br><br>Best regards,<br><br>Bobola O. Oke<div><div></div><div class="h5"><br><br><div class="gmail_quote">2011/3/15 Josué Conti <span dir="ltr"><<a href="mailto:josueconti@gmail.com" target="_blank">josueconti@gmail.com</a>></span><br>
<blockquote style="margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="gmail_quote">Hello Bobola, thanks for your response.<br>So, I'm using Euroisdn with a Siemens HiPath 3750 and Qsig with Siemens HiPath 4000.<br>
Because we don't need to "facility enable" in this case (HiPath 3750) just ANI interchange between user's, ok?<br>
In another response I was send to you a configurations sample for Asterisk and Siemens may you look this?<br>One more time, best regards and good luck in your project.<br>If you need please contact us.<br><br>Josue<div><div>
</div><div><br><br>
<div class="gmail_quote">2011/3/14 Bobola Oke <span dir="ltr"><<a href="mailto:okebobola@gmail.com" target="_blank">okebobola@gmail.com</a>></span><br><blockquote style="margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="gmail_quote">
<div dir="ltr">Thanks guys,<br><br>I got the layer1 link up.<br><br>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.<br>
<br>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. <br><br>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. <br>
<br>Thanks alot.<div><div></div><div><br><br>
<br><div class="gmail_quote">
On Fri, Mar 11, 2011 at 1:17 AM, Edwin Lam <span dir="ltr"><<a href="mailto:edwin.lam@officegeneral.com" target="_blank">edwin.lam@officegeneral.com</a>></span> wrote:<br>
<blockquote style="margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="gmail_quote">
<div>On 3/10/11 6:43 AM, Bobola Oke wrote:<br>
<blockquote style="margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="gmail_quote">
<br>
The telco has a DB9 terminated interface straight to the PBX and I cannot make<br>
sense out of the interface for the PBX. What kind of interface is this? How do I<br>
connect the RJ48 of the PRI cards to make this whole setting work.<br>
</blockquote>
<br></div>
searching through this list's archive and found this:<br>
<a href="http://lists.digium.com/pipermail/asterisk-users/2006-December/174258.html" target="_blank">http://lists.digium.com/pipermail/asterisk-users/2006-December/174258.html</a><br><font color="#888888">
<br>
<br>
-- <br>
Edwin Lam <<a href="mailto:edwin.lam@officegeneral.com" target="_blank">edwin.lam@officegeneral.com</a>><br>
Systems Engineer, OfficeWyze, Inc.<br>
Ph: <a href="tel:%2B1%20415%20439%204988" target="_blank"></a><a href="tel:%2B1%20415%20439%204988" target="_blank"></a><a href="tel:%2B1%20415%20439%204988" target="_blank"></a><a href="tel:%2B1%20415%20439%204988" target="_blank">+1 415 439 4988</a> Fax: <a href="tel:%2B1%20415%20283%203370" target="_blank"></a><a href="tel:%2B1%20415%20283%203370" target="_blank"></a><a href="tel:%2B1%20415%20283%203370" target="_blank"></a><a href="tel:%2B1%20415%20283%203370" target="_blank">+1 415 283 3370</a><br>
<a href="http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20" target="_blank">http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20</a></font><div><div></div><div><br>
<br>
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