similar to: SIP Provider Recommendation in US

Hi! My customer want's to allow calls to landlines in EU and US and disallow calls to cells in EU. Rest of countries are blocked. Country blocking is easy... Is there a service that allows checking phone number? Maybe some specific Enum? I ask for number and server responds with info, for example: "Cell Phone, Belgium" or "Land Line, Germany". -- Piotr Gorski

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

I listened to your email using DriveCarefully and will respond as soon as I can. Download DriveCarefully for free at www.drivecarefully.com Sent via my BlackBerry from Vodacom - let your email find you!

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

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

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

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.

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 recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' - No matching peer found my logger.conf

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

Is there any integration for ACT! and asterisk? I've googled for hours and haven't been able to find anything. Thanks David [cid:image001.png at 01CBE88E.66E8E450] -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110322/ecc019cc/attachment.htm> -------------- next part --------------

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

Hi folks, I repeat "as is" the title of a post someone did a few months ago, since I am facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900

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

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 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