similar to: asterisk / channel bank

Hello: I would like to know what kind of hardware is needed to have a good performance with asterisk for 12 concurrent VoIP calls ? Also if anyone had used the Adtran Total Access 750 can you give me some opinions about it ? thanks, Ivan

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

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

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?,

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

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

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

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

I have one of my asterisks off an xDSL modem with a dynamic IP address. The IP address keeps on changing so I am curious as to what other people are doing to work around this issue. I am contemplating to register for a dynamic DNS service but then my communications become dependent on the availability of this service. The "host=dynamic" for SIP seems interesting. Any suggestions?

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

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 2280 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20031012/70396f74/smime.bin

Hi, Maybe someone has seen this before... I've installed a new T100P, but it doesn't seem to work. I've attached the T100P to an Adtran 750 using a crossover cable. The Adtran shows a red alarm on the T1 interface. The Adtran has been set to factory defaults with FXS cards in 1-3 and an FXO card in 5. The T100P shows no signs of life--the leds are not lit. Should I be seeing

Jay, I have had a lot of trouble with the FXO ports on Adtran TA750. Unless the incoming POTS lines have a balance impedance, they will buzz very bad. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Sunday, April

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.

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,

It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk ? Somebody offered me that hardware, but I do not know if thats good hardware for Asterisk. rgs, Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030820/4a9e4608/attachment.htm

Hi all, I'm currently evaluating ideas for a phone system for our new office and I have a copy of Asterisk along with a couple of H.323 phones setup in the lab here. I'm very impressed - thanks for the great software! (I was especially impressed that the IAX connection to Digium worked out of the box like that - someone at Digium no doubt has a voicemail from me this morning!) ;-) What

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

Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank Thanks Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/1131cad4/attachment.htm