<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div style="RIGHT: auto"><SPAN style="RIGHT: auto">i can make bouns mints to asteri<VAR id=yui-ie-cursor></VAR>sk and elastix just give me the ips and i will add mints send the ip or host to </SPAN></div>
<div style="RIGHT: auto"><SPAN style="RIGHT: auto"><A href="mailto:civic_tito@yahoo.com">civic_tito@yahoo.com</A></SPAN></div>
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<DIV style="BORDER-BOTTOM: #ccc 1px solid; BORDER-LEFT: #ccc 1px solid; PADDING-BOTTOM: 0px; LINE-HEIGHT: 0; MARGIN: 5px 0px; PADDING-LEFT: 0px; PADDING-RIGHT: 0px; HEIGHT: 0px; FONT-SIZE: 0px; BORDER-TOP: #ccc 1px solid; BORDER-RIGHT: #ccc 1px solid; PADDING-TOP: 0px" class=hr contentEditable=false readonly="true"></DIV><B><SPAN style="FONT-WEIGHT: bold">From:</SPAN></B> Alexandre Rodrigues <alex454@gmail.com><BR><B><SPAN style="FONT-WEIGHT: bold">To:</SPAN></B> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> <BR><B><SPAN style="FONT-WEIGHT: bold">Sent:</SPAN></B> Thursday, March 22, 2012 6:28 PM<BR><B><SPAN style="FONT-WEIGHT: bold">Subject:</SPAN></B> Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk 1.8.4.2 "tooooooooooooooooo"<BR></FONT></DIV><BR>Hello,<BR><BR>Facing the same problem with the following debug skinny log:<BR><BR> -- Asked to indicate 'Stop tone' condition on channel
Skinny/500@duba-23<BR>Received Alarm Message: 32: Name=SEP001XXXXX Load= SCCP11.8-3-4SR1S :<BR>Invalid SCCP message! : ID :83<BR>Received Alarm Message: 32: Name=SEP001XXXXX Load= SCCP11.8-3-4SR1S :<BR>Invalid SCCP message! : ID :83<BR><BR>Did you solved the problem?<BR><BR>Thanks in advance,<BR><BR>Alex,<BR><BR>2011/6/25 bilal ghayyad <<A href="mailto:bilmar_gh@yahoo.com" ymailto="mailto:bilmar_gh@yahoo.com">bilmar_gh@yahoo.com</A>>:<BR>> Hi All;<BR>><BR>> Again, the Cisco IP Phones 7942G and using Skinny:<BR>><BR>> I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file.<BR>><BR>> The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination answer).<BR>><BR>> Also if we call to these phones, and we pickup
handset of the 7942G, I am hearing toooooooooo and no voice (no one hear voice .. source and destination are not hear).<BR>><BR>> What about be? Is it related to skinny channel that does not work?<BR>><BR>> In that case, skinny channel is not working fine and that means, I have to use SIP !<BR>><BR>> Did any one face like this problem?<BR>><BR>> Another problem, if I did changes in the extensions.conf and I need to reload, then I can not reload only the extensions.conf, I have to do reload and that will cause a reset for the Phones.<BR>><BR>> Any advise? Did anyone tried skinny and faced those problems?<BR>><BR>> Regards<BR>> Bilal<BR>><BR>><BR>> ----------------<BR>>> wow I think someone needs to just spend some time reading<BR>>> and playing. Getting these phones working is not rocket<BR>>> science and there are similarities with how to do firmware /<BR>>> config
pushes.<BR>>><BR>>> Not to sound mean but RTFM<BR>>><BR>>> Sent from my iPhone<BR>>><BR>>> On Jun 21, 2011, at 7:45 PM, Warren Selby <<A href="mailto:wcselby@selbytech.com" ymailto="mailto:wcselby@selbytech.com">wcselby@selbytech.com</A>><BR>>> wrote:<BR>>><BR>>> > On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad <<A href="mailto:bilmar_gh@yahoo.com" ymailto="mailto:bilmar_gh@yahoo.com">bilmar_gh@yahoo.com</A>><BR>>> wrote:<BR>>> > Dear Warren;<BR>>> ><BR>>> > Please, keep all discussions to the list.<BR>>> There's no need to email me personally about this.<BR>>> ><BR>>> > <snip><BR>>> ><BR>>> > cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written<BR>>> that it is SIP IP Phone load) and<BR>>> cmterm-7942_7962-sip.9-2-1.zip which is written that it is<BR>>> SIP IP Phone firmware files only.
So what is the difference<BR>>> between them "the load and the firmware"?<BR>>> ><BR>>> > The .sgn file is basically just a zip container that<BR>>> the Cisco Call Manager uses. You'll want to grab the<BR>>> zip file, extract the contents of the file into your tftp<BR>>> root directory. The latest firmware that I've used was<BR>>> 8.5.2, in which most everything I needed worked for<BR>>> me. I don't know specifics about the later versions of<BR>>> Cisco's SIP releases.<BR>>> ><BR>>> > Now, when I need to do the upgrade for the Phone, then<BR>>> I have to determine in the xml files the needed firmware?<BR>>> ><BR>>> > You should have, at least with firmware 8.5.2, the<BR>>> following files in your tftproot directory after unzipping<BR>>> the zip file:<BR>>> ><BR>>> > apps41.8-5-2TH1-9.sbn<BR>>> >
cnu41.8-5-2TH1-9.sbn<BR>>> > cvm41sip.8-5-2TH1-9.sbn<BR>>> > dsp41.8-5-2TH1-9.sbn<BR>>> > jar41sip.8-5-2TH1-9.sbn<BR>>> > SIP41.8-5-2S.loads<BR>>> > term41.default.loads<BR>>> > term61.default.loads<BR>>> > XMLDefault.cnf.xml<BR>>> > SEP[_MAC-ADDR_].cnf.xml<BR>>> ><BR>>> > I provide samples of the last two files on the blog<BR>>> post mentioned earlier. The last file, that starts<BR>>> with SEP, should contain the actual mac address of the phone<BR>>> you are trying to provision. So, for example, it would<BR>>> be SEP0003C9DD5624.cnf.xml, if the mac address of your phone<BR>>> was 0003.C9DD.5624. The example files are pretty much<BR>>> all you need, just go through them and change any location<BR>>> specific variables (such as _USER_, _IPADDR_, or _PASSWD_)<BR>>> to the proper values for your
environment.<BR>>> ><BR>>> > Once you've got your tftp server setup properly with<BR>>> all of the appropriate config files, plug your phone in and<BR>>> follow the instructions at the bottom part of my blog post<BR>>> that explain how to get the phone reflashed to the SIP image<BR>>> and registered to your asterisk server.<BR>>> ><BR>>> ><BR>>> > --<BR>>> > Thanks,<BR>>> > --Warren Selby, dCAP<BR>>> > <A href="http://www.selbytech.com/" target=_blank>http://www.SelbyTech.com</A><BR>><BR>><BR>> --<BR>> _____________________________________________________________________<BR>> -- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> --<BR>> New to Asterisk? Join us for a live introductory webinar every Thurs:<BR>> <A
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