Hey Danny,<br><br>Yes, modifying the dial command fixed it.<br><br>Thanks alot.<br><br><br><br>Best regards,<br><br>Bobola O. Oke<br><br><div class="gmail_quote">On Wed, Mar 30, 2011 at 6:03 PM, Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span> wrote:<br>
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<p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">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.</span></font></p>
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<p class="MsoNormal"><b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b><span style="font-weight: bold;">On Behalf Of </span></b>Bobola Oke<br>
<b><span style="font-weight: bold;">Sent:</span></b> Wednesday, March 30, 2011
11:36 AM<br>
<b><span style="font-weight: bold;">To:</span></b> Josué Conti<br>
<b><span style="font-weight: bold;">Cc:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users]
Connecting Asterisk to Siemens Hipath 3750</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">Hi guys</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">Thanks alot for the support.</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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" target="_blank">http://www.voip-info.org/wiki/view/Siemens+Hicom</a></span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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.</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">Thanks.</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">Bobola</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">2011/3/16 Bobola Oke <<a href="mailto:okebobola@gmail.com" target="_blank">okebobola@gmail.com</a>></span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">2011/3/15 Josué Conti <<a href="mailto:josueconti@gmail.com" target="_blank">josueconti@gmail.com</a>></span></font></p>
<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">2011/3/14 Bobola Oke <<a href="mailto:okebobola@gmail.com" target="_blank">okebobola@gmail.com</a>></span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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.</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">On Fri, Mar 11, 2011 at 1:17 AM, Edwin Lam <<a href="mailto:edwin.lam@officegeneral.com" target="_blank">edwin.lam@officegeneral.com</a>>
wrote:</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">On 3/10/11 6:43 AM, Bobola Oke wrote:</span></font></p>
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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.</span></font></p>
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<p class="MsoNormal"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">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"><span style="color: rgb(136, 136, 136);"><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></span></font></span></font></p>
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<br>
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