similar to: MySQL and VoiceMail again

Most of my outbound calls are handled by macro contexts: [macro-pstn] exten => s,1,Wait(.5) exten => s,2,PlayBack(beep) exten => s,3,PlayBack(silence/1) exten => s,4,Dial(${ARG1}/${ARG2}${ARG3}) exten => s,5,Playtones(Congestion) exten => s,6,Wait(3) exten => s,7,Hangup which is called like exten => _920[1289]XXXX,1,Macro(pstn,Zap/g1,9,${EXTEN:${TRUNKMSD}}) Because of

where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) also anyone got a fix for the horrible speaker phone on the 200's

Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14

Why is the SELECT statement below putting a "?" in for the username? I am using today's CVS. Jan 13 18:48:41 WARNING[7570]: res_config_odbc.c:105 realtime_odbc: SQL Execute error! [SELECT * FROM sip_buddies WHERE name = ?] Full dump: Sip read: REGISTER sip:198.88.216.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK-ffmzndpfrao2;rport From: "Mike's Peoria

Is anyone signed up with Vonage and using an asterisks box?? Also what VOIP providers would anyone recommend? -- James Moran Potential Technologies http://www.potentialtech.com

Anyone know what protocols support a fax machine i.e. g.729, g.711, etc? ---------------------------------------- Michael Shuler

I'm trying to switch from one call to another incoming call from PSTN. When I'm getting a "beep" I press flash but instead of swithing to the second call, I'm getting a dial tone. even if I press *0, I cannot connect to the second call. Anybody had this problem? Tx, Bogdan

>I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN?

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets and 400 analog units. For the analog units I have quotes for 9 ADIT 600 48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used neither. Which is the best choice? The price difference is not that great. I am looking at Citelinks 24 port Handset Gateway for the Nortel Digital units. (Any other suggestions

Hi, I have Asterisk running with the VoiceMail. Using the latest CVS. I have my extensions.conf setup so that if a SIP caller dials *99 the VoicemailMain() as follows: exten => *99,1,Wait(1) exten => *99,2,VoicemailMain() A couple days ago I installed the MySQL/Voicemail support described at http://www.voip-info.org/wiki-Asterisk+voicemail+database Now for some reason

Searched the archives thoroughly... Can't find this specific problem... Simple setup with Asterisk on RedHat. No voice cards in the box, 2 SNOM 200 phones... Phones seem to work well, can leave VM, Message Waiting Indicator lights up but when I try to retrieve messages the call terminates and the following happens: -- Executing VoiceMailMain("SIP/520-a25e",

Can not check voice mail-box externally. I'm trying to log-in externally (from PSTN line) to check my "voice-mail" so I created context to authenticate log-in ... exten => 7,4,Authenticate(01894546) exten => 7,5,DISA(4789|disa-access) Authentication works OK, I get inside dial none enter my mailbox extension but it doesn't accept my mailbox password even though it is the

Hi, I'm running the latest CVS HEAD version of asterisk, and I'm experiencing hangups during voice conversation. This happens quite regularely and often. The problem is in dsp.c, line 1235, where it says accum /= len; But `len', at this point, is 0, resulting in a SIGFPE. The routine ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is setting p->fr.datalen to

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

Hello All, I have Polycom IP 500 phones which I would like to have message waiting indicators on. So far, I have my system running well but the problem I am seeing is that MWI doesn't seem to tell my phone that it should display a MWI state. The light does not show when you have message nor is there any indicator on the text lines of a message waiting. The wiki doesn't cover this

I have successfully implemented 1204 in semi production environment. Just want to share that it works very well, through the firewall (NATed). Unfortunately, it can not register with the server (and authenticate) but otherwise everything is fine. The audio quality is very good. Regards, Wojtek -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

Hey All, We are looking to expand our usage of Asterisk and I am trying to make as much of the configuration dynamic as I possibly can. The only part that I'm having problems with is sip.conf. I can get asterisk to register each extension with our local SER SIP proxy dynamically by using the "sipfriends" table in the database, but I'm having trouble with the message waiting

(Second try to send this. I wonder if the first one gotten eaten by a spam filter; I'll link to patches instead of attaching them.) Here are the tincctl patches I've been working on. They apply to http://www.tinc-vpn.org/svn/tinc/branches/1.1@1545. I intend to commit them once the crypto stuff's fixed. Since they're basically done, I'm emailing them now for review and in case