similar to: Another Problem!

hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to

hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee

hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with, 4 port station interface card ,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) <------------------------------>

hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030906/ed1d46cf/attachment.htm

<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

hi All, quit new to asterisk, can anybody tell me whether Wildcard E400P and Wildcard E100P support R2 CAS protocol. if they do, what is the value, i should set to 'signalling' parameter in the zapata.conf file? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, This question is about DBPut and DBGet, Can i put about 1000 keys in a single family, (only once for the lifetime) for ex. exten => _X.,5,DBput(family/key1=${val}) ... exten => _X.,5,DBput(family/key1000=${val}) like above and if i later retrieve it, randomely, with inbound calls, will it affect performance? Surajee -------------- next part -------------- An HTML

<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

<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; --

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198

hi, The prob is when we ring 2 channels simultaneously, only 1 channel is actually ringing. In our configuration, the Asterisk box is connected to an E1 channel bank, where 15 analog extensions are conencted to channelbank inturn. We tried following, Dial,Zap/g4/444&Zap/g4/448|20|t Heres the output, -- Executing Dial("IAX2[trunk10@trunk50]/1",

dear All, I am jus installed centOS 5.0 from the distrubution to be as our mail server i have installed the following sendmail-8.13.8-2.el5 mailsacnner 4.61.7 spamassassin 3.2.1 clamav-0.91-1.el5.rf and every thing is workin fine but i want to implement domain keys so i downloaded the dkim-milter-1.2.0 and uncompress it edited the site.config.m4 file n folled intructions in the

Hi All! I have been messing around with this problem for about a week but to no avail! The following data has been cut down in order to make my question reproducible. The alldat data frame includes 2 columns: 1 date column and 1 factor column (equity names)).

Dear ALl, Sorry for my posting here but do apprecite if u guys help me.. I have already CenOS 5 and been used as my Primary DNS and primary sendmail mail server and is working perfectly fine i also have another Centos MAchine being used as my secondary DNS server and been workin fine.. i want to use this machine as my backup mail server and has gone to the DOCS have configured DNS and

hello *'s, Astcc not workin what is correct format for defining 1-database 2-brands 3-trunks 4-routes i define all these things but not workin may be i define in wrong format.I have FXO card installed.can anyone implement it and also my sip phone generates very loud noise wat is that i tried several settings but not hear any voice just noise. sip.conf [general] context=from-sip port=5060

Dear All, I am curently running a CENTOS server with the following setup CentOS release 5 (Final) Kernel 2.6.18-8.1.8.el5xen on an i686 bind-9.3.3-10.el5 sendmail-8.13.8-2.el5 apache httpd-2.2.3-11.el5_1.centos.3 MailScanner ver 4.66.5 ClamAV 0.92 SpamAssassin version 3.2.4 SquirrelMail ver 1.4.13 Now all this have been workin perfectly fine for a long time. now if i do a yum upgrade i see the

hi, can we have arrays in contexts? i tried like this, but didn't work :-( declaration myarray[0]=192.168.3.4 myarray[1]=192.168.3.1 usage myvalue = ${myarray[${myval}]} pls tell a way to do this Thanx a lot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030521/f8b61c89/attachment.htm

hi guys i have CentOS-5.1 with samba-3.0.25b-0.el5.4 and i took this workin well smb.conf from slackware12 ( commands to be done befor this ): # useradd wbc -c /home/wbc -g users# smbpasswd -a wbc# mkdir /export# chmod u+rwx,g+rx,o+rx /export# chown wbc.users /export the conf file:[global]workgroup = WORKGROUPnetbios name = sanshirosecurity = share [share]comment = Datapath = /exportforce user

Hello Group, I am new to asterisk but learn a lot about it to this mailing list and wiki currently i am facing problem about sip phone i have "PA 1688" chipset ip-phone and i have iptel.org sip account i registered locally and through iptel.org comfortably my problem is that when i called from my sip phone to analog or any number after connection my sip phone generates very load scartchy