similar to: Testing from where number is...

-----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 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 --------------

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 becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-night operations from the reputable providers. --Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey Guys! We have two Asterisk with A102D Sangoma cards now i want to connect them back-to-back over PRI line via Cross-cable so what would be the configuration specially timing source and all? anybody did it before like this ? I want to make sure everything before putting in production.. (saving my downtime) -S -------------- next part -------------- An HTML attachment was

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

Hi, I am having a little problem and I hoped maybe I could get some help here. I deployed a Asterisk 1.8 server of my own to make SIP calls just between my friends. The server is configured with a public IP address. My friends and I are using "Acrobits Softphone for iPhone" as a client. I am using its push service which is hooked up to my Asterisk server. Now, the current situation is

Hello, I have Asterisk-1.8.3.2, dahdi-linux-complete-2.4.1+2.4.1, and libpri-1.4.11.5 installed and running on a Ubuntu 10.04 server all built from source. Everything is working nicely except one small issue. The CDR records are stored in the CSV file correctly and complete. The MySQL storage is working as it should and is automatically updating all the fields except the CLID field. I have

Hello! Im new to Asterisk configuration and I have few questions regarding its configuration. I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free calls from each of 4 pstn lines... Can I configure Asterisk to call thru pstn line that has free minutes? For example Outgoing calls are going through PSTN 1 for 60 minutes. When I use all of these free minutes - outgoing

i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '8888246463' rejected because extension not found. but the extensin existed -- Bayardo S?nchez Garc?a Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail:

i create inbound confi my confi is: [incoming] exten=> 18888246463,,1,Dial(SIP/8003,60,rT) exten=> 6463,1,Dial(SIP/8003,60,rT) exten=> 18888246463,,n,Wait(5) exten=> 18888246463,,n,Hangup but y calling and send this error in my CLI: [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '8888246463' rejected

i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 "Service Unavailable" back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION'

Anyone else having problems connecting to http://downloads.digium.com/pub/ ?? Jonn

I need help need recording all call for my pbx but i am a novato in asterisk my confi for record is: exten=>_NXXXXXXXXX,n,Set(CALLFILENAME=CLIENTE-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}) exten => _NXXXXXXXXX,n,MixMonitor(${CALLFILENAME}.gsm,m) exten => _NXXXXXXXXX,n,Dial(${TRUNK_CLIENTE}/${EXTEN}) -- Bayardo S?nchez Garc?a Web Developer - Internet

I have a problem when I call a good record but I make a call to return to the same number I erased the previous record, and I replaced with the last call -- Bayardo S?nchez Garc?a Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanchezg at hotmail.com Skype: bayardo.sanchez This email is intended

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!

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug

Hi, I have a Samba server, it's main goal is to store documents of all users of the network. Certain users abuses and save mp3, mov, jpg, gif and other files that must be saved in other file server, so I need to restrict the those type files and allow my users save only office files like .doc, .docx, .xls, .ppt, .pdf thanks for your help. Bayardo.