similar to: IAX2 ports usage

hi ! I have asterisk with Licensed G.729 codec enabled. Whenever I make a call using this codec a warning apears as, WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames

hi! I wanna do some arithmatic operations (addition and substraction -integer operation) inside extensions.conf. Is there a simple way to do this. If I do yy = ${xx} + 1 // say "xx" is initialized to '0' the resulting "yy" will show "0 + 1" Obiviously not the result I need. Any help !!!!! denzel.

Hi All, What the support that asterisk has to send/receive Faxes? Can I plug a FAX machine in the a FXS extension and send out Faxes? What's the codec I need to use? g.711? Also can we receive a FAX into a FXS extension in Asterisk PBX? Also I need to know if we can send/receive FSK data from/to an extension plugged into Asterisk PBX? For example if there's a phone model which can send

hi! I've got cisco 7960G working with * box. Calls could be Blind Xfered through the phone but not the supervised transfer( Message on the phone: Transfer failed). Even when I put the caller on hold and resume it later, I can't hear the other side but the otherside can hear me. (It shows as the line is connected though. Yet the respective caller entry blinks.) Any suggestions most

hi We are using snome 200 IP phone with *. It works OK. But after a period of time we can't hear any sounds for any icoming or outgoing calls. I've got two of these phones. Same symptoms occur to both of these( not at the same time ) and the problem remains until the phone is completely rebooted. Don't know whether this's * or Snomes' prob. Any help would be appreciated.

hi! I'm looking for a robust hardware IP phone which supports SIP protocol inorder to implement a call centre. Have anyone used CISCO SIP phones (eg:- 7960G ) with asterisk. From what I know these CISCO IP phones are very robust and feature rich. Yet I'm nervous whether * don't like CISCO at all. Thoughts are most welcome. denzel. -------------- next part -------------- An HTML

hi! I've got a asterisk system running with around 50 per calls per minute. I've connected * to internal pabx and outside telecom using E1 (ISDN pris). Sometimes calls disconect abnormally. Is this something we have to live with or is it a bug in CVS code ? denzel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

<P>hi,</P> <P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried&nbsp;editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P> <P>Thank you inadvance,</P> <P>Surajee</P> <P>&nbsp;</P><br> --------------This mail sent

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

<P>hi Everyone,</P> <P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P> <P>*CLI> <BR>&nbsp; == D-Channel on span 1 up<BR>&nbsp;&nbsp;&nbsp; -- B-channel 1 successfully restarted on span 1<BR>&nbsp;&nbsp;&nbsp; --

<DIV><FONT face=Arial size=2>hello All,</FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9</FONT></DIV> <DIV><FONT face=Arial size=2>to take an outside call through the hardware pbx, our

Hi All, I know, IAX is NAT friendly, but... I have a problem running gnophone from a box behind NAT firewall. I can register gnophone with * through NAT, but when I try to make a call it instantly disconnects. CLI iax show peers command tells me that peer is unreachable. However this peer is registred. Gnophone also tells me that it is registred. It seems that registration handshake has

Right now I've been working on setting up an Trixbox server on our internal network. Its behind the firewall, but I'd like to open up the firewall to it because we sometimes have developers working off site and I'd like them to be able to connect. Is this safe to do? I've got the "Allow Anonymous Inbound SIP Calls" box unchecked in freePBX. Is there anything else

Hello, using Asterisk 1.4.28, I encountered a problem with SIP RTP port allocation. I found some entries in mailinglist and bugtracker regarding this issue, but only old ones. My rtp.conf has [general] rtpstart=30000 rtpend=30100 so 100 ports available. I know that up to 4 ports per channel can be used and so up to 25 channels are possible. But even earlier I often get the error about

Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are "generic," and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a

What is the state of speex support in asterisk? I saw the codec seems to be there. Can speex be used on IAX2 links? Is there much work still to be done? many thanks, --J.

hi! I'm trying to get zapras working in GSM csd network. Whenever a dialup call is initiated from the mobile to the * gateway the following appears in the log and zapras terminates. Phone gives the error dialup not answered. ==> /var/log/messages <== pppd[2310]: Plugin zaptel.so loaded. pppd[2310]: Zaptel Plugin Initialized pppd[2310]: Using zaptel device 'stdin' pppd[2310]:

Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/attachment.htm

Just thought I'd share my success with chan_h323 and our Symbol NetVision phone (4046-100-US). Voice quality is excellent, and setup was trivial. The new NetVision firmware (4.21) is much better than the 3.x stuff. It gives the phone a whole new look and feel. The hardest (and longest) part was getting OpenH323 compiled. After that, H.323 ran out of the box. I simply uncommented

Hi, After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This