similar to: Can you dial with different CID's?

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

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

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

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

I'm used to seeing the same versioning (maybe I've been gone too long) Is zaptel 1.2.15 the right one for asterisk 1.2.16 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/e57842ef/attachment.htm

Jean-Louis curty to Asterisk More options 4:38pm (7 minutes ago) Hi everybody, I'd like to know if anybody tried to write a xml doc to monitor the number of calls in Q, when working with an ACD it's convenient to see how many calls are waiting so the agent can speed up the conversation when it gets too busy :-) I was wondering if it was poss to display this info on a display of a

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 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

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

Hey, all. I'm trying to set my CID such that, internally, I see a four-digit extension (which is also handy when checking VM), but externally, I see the full 10-digit number. So I plugged these lines into my extensions.conf: exten => _XXXXXXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2) exten => _XXXXXXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1}) exten =>

Is there anyway to force an autopickup on a cisco 7940 / 60 from the dialplan ? My problem is that I am originating a call from the AMI, with the internal user being called first, and then connecting to external user. However, sometimes the internal user doesn't pick up the phone, so the call is never placed. I need to know the results of the call so I need to be able to either a) get

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.

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

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 a weird situation. A polycom phone is configured to use system pbx1 as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three systems have identical sip.conf files. The phone is registered on pbx1. I shut down the Asterisk application on pbx1. I make a call. The phone sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The phone sends

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