search for: drivecarefully

Displaying 12 results from an estimated 12 matches for "drivecarefully".

2011 Mar 03
3
Testing from where number is...
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
2010 Nov 10
1
CentOS Digest, Vol 70, Issue 10
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!
2011 Apr 16
4
Jabber / GTalk / hints
-----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 :
2011 Apr 15
3
sip error logging
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
2011 Apr 16
4
Jabber / facebook chat?
-----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
2011 Mar 22
3
Act! Integration
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 --------------
2011 Mar 22
1
How to use Atxfer in AMI
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
2011 Mar 03
4
SIP Provider Recommendation in US
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:
2011 Mar 22
3
Asterisk PRI back-to-back connect
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
2011 Feb 18
3
FAX on PRI to MFCR2
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
2011 Mar 17
3
Call are established, but voices can't be heard
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
2011 Mar 28
8
CDR MYSQL missing field data
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