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Hey all! I had an experience trying to set up an E1 in Brazil which could help somebody. In Brazil is very common telcos to have just R2 digital as their primary signaling. As I were trying to set up an E100P, which does not support R2 yet, I had to test an other signaling which works perfectly with Asterisk. They call this signaling as RDSI, using ccs as framing and PA (primary access) as

Hi everybody,, I need to block incoming collect calls to my Asterisk box but I could not find out where to do that. Went to zaptel.h but I did not see any timing which can be applied to collect calls. Does anybody knows if I can set this up in Asterisk? I'm using an E100P connected to the PSTN and a T100P connected to a Zhone 100. Version: Asterisk CVS-05/30/04-16:28:04 thank you Oz

Hello, I am trying to see what I need to do SIP to H323 using G.729. I have Oh323 and SIP working with G711 fine. If I have a SIP client configured to use G729 and H323 client also to G729, how many license should I need to buy from Digium. Many thanks SW

Hi, Does anybody have ATA 188 working with any kind of fax machine? I've tried many different configuration following the Cisco Online Manual and I couldn't get this working with Asterisk. I were trying do change the ATA Connect Mode and Audio Mode reading the (http://www.cisco.com/en/US/products/hw/gatecont/ps514/ products_configuration_example09186a00800d698e.shtml) and allowing

Hi, We currently have an Asterisk box (P4 2.4 Ghz, 512Mb RAM, 1 T100P, Zhone Channel Bank and 1 E100P) connected to an ISDN running without any problems. That machine is working for about 1 year. Two days ago, we decided to switch that machine for two PowerEdge 600SC (HA) and we got some problems. The running machine (P4 2.4Ghz) has an Intel motherboard with four 32-bit 3.3v PCI slots,

Hey All, I need to forward an extension to an other depending on the current time but I could not get it done with GotoIfTime. What I'm trying to do is ring on the extension 1 if time is between 8:00AM and 2:00PM and on extension 2 if is between 2:01PM 11:00PM. exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333) exten => 111,2,Dial(${Person1}) exten => 111,3,Dial(Hangup) exten

Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz -------------- next part -------------- >

Hi all! Guys, I'm testing the Cisco ATA 188 with my Asterisk (version 09/10/03) and I faced with a low call volume heard just for people who is not "under" the ATA. I mean, if I call a person whose extension is connected at the ATA, he can hear me perfectly, but I get a low call volume. Is it possible to change this call volume? Can I do something in order to get a little high

Hi, I'm using Asterisk version (Asterisk CVS-05/30/04-16:28:04) on Debian Woody. Sometimes I get this warning message: Jun 14 13:32:41 WARNING[10251]: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When that is happening, Asterisk gets slow and close all remote active connections (asterisk -vvvcr). VoIP call alse gets bad at this time. That message appears many times,

Did anybody get this error message before: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When it's happening, Asterisk gets freezed and talkers can not hear each other. This message appears like in a loop at the server's screen. thank you Oz _________________________________________________________________ MSN Messenger: instale grátis e converse com seus amigos.

Hello.... I have the follow situatuion: < ISDN > | | V E100P |----------------| IAX2 / g729A |----------------| T100P | Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - - -> |--------------| | | | | | Zhone | ----------------- ----------------- --------------- Here's the situation: I receive calls from the PSTN