similar to: Why is the internet connection important to LAN and PSTN calls?

Howdy, How can you tell if RTP traffic has been reinvited/is bypassing an * server? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com

Is it possible to dial more than one extension with a different CID to each extension? I'm thinking macros might be needed, but I don't have a good handle on macros. Is it possible? Any hints? BTW - this would be used for showing an internal extension to one phone and a PSTN accessible number to another phone. Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road

Hola, What would a valid regexp in Asterisk be to identify a NANP number, i.e., NXXNXXXXXX? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:brent.torrenga at torrenga.com web:www.torrenga.com

I am thinking of turning on the NAT option in our Cisco phones (and the corresponding sip.conf modification) to allow the phones to be taken outside the LAN. Can anyone think of any reason not to just always turn on the NAT enabled option? I can't think of a reason not to always operate these phones with this enabled, since it would likely allow them to be taken outside our LAN and used.

I think A@Home allows a user to search a directory by either first OR last name, right? I don't know for sure since I don't run A@Home. I would like to offer that functionality in my system - and I'd have done it by now if there was a prompt where Allison asks "press 1 to search by first name, press 2 to search by last name". But I don't think that prompt exists. Can

Dear list (and more specifically Bret), I am getting one-way (inbound only) audio when trying to place a SIP call via voip.trxtel.com (i.e. 18005558355@voip.trxtel.com). The Cli spits out "== Forcing Marker bit, because SSRC has changed" 5 times after atempting a native bridge. I realize this is most certainly a NAT issue, the * server is behind one. Sip.conf has externip=, and

Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for

Hi, I have got my setup almost how I would like it now, but I have just two last remaining issues that I cant seem to find answers too so i'd be grateful if someone could help? 1) Since upgrading my Cisco 7960 SIP phone to P0S3-08-2-00 the phone now displays the IP address of my asterisk server alongside the caller ID of the incoming call. For example "0123456789@192.168.0.1",

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk

Has anyone had problems getting their preffered codecs on the Teliax web interface taking effect? I have two accounts, two separate yet similarly configured * servers. On one account the settings took right away - on another server I am getting no result. In fact, no matter what I change the settings to, only the old codecs are usable (otherwise * says it can't negotiate a codec). Teliax

Dearest List, I guess I missed this point: Is it true that if you change the echo canceler in zconfig.h, and then recompile/install your zap modules, that for this to be taken into effect by * you must then recompile/install *? I would have figured that the zap echo cancellation method was independent of *, and I don't recall seeing any docs mentioning either way. Sincerely, Brent A.

Howdy: Does echo only occur on analogue PSTN lines, or can it also occur on PRI and BRI lines? If so, for the same reasons? This is a part of our consideration to transition to BRI. Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com

I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the

We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am a total virgin to ISDN. I understand that we need two BRI circuits to provide four voice channels, and that the hardware to speak to the BRI circuits can be passive or active, with the active type being much more

It does on my test phone. Is your tftp server available? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Brent Torrenga Sent: Monday, April 17, 2006 11:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip Notify cisco-check-cfg - Does it still workwith 8.2? Has anyone else noticed that

>Brent, > >you mean, I could just remove the remark signs and number it 103, 104, >105, .... since it does not matter why it failed (busy, congestions) >(maybe for statistic purpose to add a log entry for the move to the next >provider). > >bye > >Ronald Yup. Take a look at the macro solution, too. I don't fully understand macros (I'm no programmer), and

Has anyone else noticed that notifying a 79[46]0 with cisco-check-cfg doesn't elicit any response from the phone using fw 8.2? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com

It doesn't seem as much broken as just annoying. I am holding off on upgrading until this resolves, but it doesn't seem to affect performance, anyways. BTW, some folks say that the server address only gets appended to the CID when a redirect or something comes about. Our experience here shows that the IP always gets appended. >Alexander Burke wrote: >> Just in case anyone here

What would cause this? It happened out of the blue: -- Executing VoiceMail("Zap/3-1", "u326@default") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/6' (language 'en') -- Playing

List, Per someone's suggestion (thanks, whoever you were) from this list, I implemented dnsmasq to prevent the issue of resolving DNS when the net connection goes down. This morning the net connection was down, and our * server didn't miss a beat. I recommend looking into: http://thekelleys.org.uk/dnsmasq/doc.html And http://www.enterprisenetworkingplanet.com/netos/article.php/3377351