similar to: How is Context Determined when Transferring a Call?

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

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.

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 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

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

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

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

So I saw mention of a way to allow dialing using SIP URI's on Dave McNett's site at http://slacker.com/~nugget/projects/asterisk/page7 Wow, awesome, I can call anywhere now. However, I think there is a more elegant way of figuring out whether or not the local * server should handle a given domain. Specifically, Dave compares a series of domains within extensions.conf to figure out how to

Does anyone have any tricks to use some logic with SIP UA's codec negotiation based on the UA's IP? What I would like to do is have Cisco 7960's use g711u when they register with a local IP, and g729 when they register with a non-local IP. I was thinking about sip.conf and making two entries for each UA, one where the host=dynamic, disallow=all, then allow=g729; the other

Out internet connection was out this morning. It seems that the SIP extensions on our LAN were affected. Behavior like: Call comes in over POTS to a TDM400P, there is a delay then before the Cisco 79[46]0's start to ring. If we were lucky enough to get a call through, then we could not transfer the call, or place the call on hold, or park the call. Outbound calls seemed to have a delay

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 use Cisco 7940's and 7960's over here, loaded with SIP, and they do very nicely. Also, the SCCP channel for * is under heavy development, and may offer a future option to convert in that direction, too (SCCP, or skinny, is their native tongue, not SIP). We got our phones from John Putnam at Global Technology Solutions - competitive price, very prompt service, and delivered them with the

Paul, Ah, I see. Our echo is largly under control now. It took me a while to figure out the gains and get them tuned, and now the echo only leaves very small artifacts. Nonetheless, this still provokes the odd complaint here and there. We use VOIP for outgoing calls when our POTS lines are congested, and we find zero echo during those calls. Therefore, I assume that our handsets (Cisco

Well, I thought and hoped my issue of random hangups on our TDM400P were related to busydetect=yes in zapata.conf. The behavior of a call being hungup has not changed, however, since setting the busydetect option to 'no'. Again, the only affected user is my loud talker... What are some causes/solutions to seemingly random call disconnects on Zap channels that people have seen? I have