similar to: Dynamic IP Addrress Work Around

Every time I send an e-mail to the * list, I receive this "AntiSpam UOL" E-mail. is anybody else experiencing the same? How can I get rid of it? Uriel -----Original Message----- From: AntiSpam UOL [mailto:andersoncbr.sspam@uol.com.br] Sent: Wednesday, September 24, 2003 11:51 PM To: uriel@adelphia.net Subject: RE:RE: [Asterisk-Users] SIP / GrandStream Configuration Ol?,

Where can I learn the syntax for DBput and DBget? is it working with MySQL? do I need to set up tables? URiel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030508/a2598dc8/attachment.htm

vmail.cgi rocks (if I can borrow the expression for Mark Street). As Mark pointed out, the /vm/INBOX messages are created with 0700 security and vmail.cgi is not happy. Apache/cgi/vmail.cgi cannot play them unless I fool around with the Apache wrapper or chmod 755 *.* thefiles myself. (Tedious, that is why I like computers). Obviously this is not acceptable. I took a trip to the

I have been fooling around with DynExtenDB and run into two glitches. 1) The code is looking for (chan->dnis) and in my case I find (null). I forced (chan-dnis) to be the same as (chan->exten). So far so good. Now I can connect and talk. This lead me to the second glitch. 2) As soon as the call ends by hanging up, the code issues a (ast_spawn_extension). This causes asterisk to drop

All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards,

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

After rebooting my asteriks server, e-mail notifications are no longer being sent after a voice-mail is left. I can see the messages in /var/spool/asterisk/vm. has anybody had the same experience? how was it resolved? Uri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/fc494bb4/attachment.htm

I have seen a couple of messages on the Xten and the work done by William Walsh (Kudos). It is not clear in my mind the advantages of SIP phones versus using GnoPhones (once we complete the work for the Windows version). Since I lack the experience with IP SIP phones, can someone, high level, tell me when it makes sense to use them. Is it complicated to set up on the Asterisk side? Thank you.

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Can anyone give me presise instructions on how to compile cdr_mysql.so? When I initially installed asterisk on the system, I didn't have mysql installed. Since then I have installed mysql, created the database and table structure for cdr_mysql and placed the appropriate settings in the cdr_mysql.conf file. However when I do a show modules at the CLI I cannot find cdr_mysql.so.

I have two loop-start phone lines connected to an Adtran TA 750 channel bank. When a call comes into asterisk thru the first phone line and the call is redirect (in transit) back thru the second phone line, the channel bank will leave one or both lines opened (ie hangup with spawn a connection again). does any body have a suggestion: 1) I checked the cables into the channel bank from both phone

I was looking at the shell in /usr/sbin and the following statement does not seem right: while :; (read colon + semi colon). In C that would be 2xsemi-colon. is this correct in a bash shell? I am having problems with safe_asterisk. when it dies, it is not being brought up again. It works very well for stop/start/startus but does not bring up asterisk when it fails. -------------- next part

Below you will find, what I believe to be a typical setup with a T100P card. My question is - 1. Is this correct? 2. What kind of phones would be needed here... (Would you have to use Digital phones) And if so what would you recommend. PRI/T1----- | | | ---------------- | | | Channel Bank | | |

I have just installed R on a Windows NT system. Unfortunately I am unable to open any of the data files I wish to work with. I have tried using read.table and scan and to the best of my knowledge am using the correct syntax. The error message I receive is Error in file(file, "r") : cannot open file [file name] I have the data in text files in white-space delimited form. I put them

Hi I used to have a xDSL with a Cisco 677 router setup that connected via PPPoE to my ISP. I now have a different ISP that uses a xDSL modem setup which means my shorewall firewall machine external interface hooks up til my ISP by directly by DHCP and gets a real internet IP while before with my router setup . My question is: With my Cisco router setup i could from my LAN access my outside IP

Hello Walter, On Sat, 13 May 2017 13:08:17 +0200 "Walter H." <Walter.H at mathemainzel.info> wrote: > On 13.05.2017 00:29, Robert Moskowitz wrote: > > I have been working, for the past few years, with armv7 SOCs and have > a number of servers working. > > > > Intel, etal are catching up with ARM and I have seen ones like: > > > >

Hi everyone (Mark, Jim). I am new to the list but thanks to both Mark and Jim, I have being using "asterisk" since summer 2001. I am just updating my version that was a year old. Yes, I know, I got busy with other things like paying bills so I don't have to sleep with the dog anymore. Anyway, one of the frustrations I have been dealing with (keep in mind that my version of

How is a callgroup used in the dial plan?? I can't see an example in the extensions.conf.. > See configs/sip.conf.samples search for pickupgroup > > RTFSC (Sample Configs) > > > Jeremy McNamara > > > > > WipeOut . wrote: > > >Is there a way to pickup a call whn using a SIP phone? > > > > > > > >> From

Callgroups/pickupgroups are allocated per channel, not in the dialplan. sip.conf and zapata.conf are the two files you're interested in. -wade ---- Original Message ---- From: wipeout@linuxmail.org To: asterisk-users@lists.digium.com, Subject: RE: [Asterisk-Users] Using callgroups (was: Taking a call for someone elses extensionfrom my extension) Date: Sun, 20 Apr 2003 16:39:15 +0000

Hello, in my company we want to implement a Two-Factor authentication in servers with Solaris(8 or 9) and HP-UX (11 or 11i). The authentication server will be a RSA. We want to use OpenSSH to ask for authorization twice, first with PAM, and then to the RSA server. I like to know if somebody already implemented. Thanks for your help Uriel Quintana La informaci?n contenida en este correo