similar to: Authentication woes.

I'm a bit further along but haven't figured out why Authentication is still failing. I've tried a telnet to port 143 and openssl connection to 993. The command I issued, per the debugging page on the wiki, is: a login info at aesoft-sbcs.com crap Here is a snapshot from my logs (yup second try and blank lines to make it easier for me to read). Oct 7 08:17:20 mx0 dovecot:

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we

I'm working at the next part of the virtual domains mail server. I'm moving this account (raanders at acm.org is a forwarder) which has a bunch of procmail rules to file into folders. My question is if it is more efficient is use? if { ... } elsif { ... } elsif { ... } else This seems to be the way many of the example scripts do it but I found at least one that used if

> From: "Roderick A. Anderson" <raanders at cyber-office.net> > Subject: [Dovecot] RHEL5/CentOS5 YUM repo, rpm, or spec file for 2.0? > > I don't remember sing any mention come across the list reference the > Subject line and nothing shows up within the first three pages of a > Google search. > > Anyone know of a YUM repo. RPM or spec file for

Anybody used the above phone with asterisk I have one working ok for calls, but having a problem with voice mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? -- Clive Email :

Anybody know of a UK supplier of "Voip Telephony with Asterisk" " by Paul Mahler ? I've searched on the web, and the only suppliers I can find are US based, and the postal charge is as much as the book. cheers -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel : 0845 0043366 Alt : 01952 402032 SIP : 84416002@voiptalk.org Mobile : 07970 856261

Hi Just uploaded source from cvs (CVS-HEAD-11/27/04-12:56:51) Zaptel and libpri make install works ok, but I get the following error when running make install in asterisk directory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ........... make *** [rpt.o] : Error 1 What have I done wrong ? (Its got to

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

FYI/Heads up, I /just/ received what looks like a phishing attempt for information about Open Source PBX usage. It says it comes from Digium but all the links (including the one for digium.com) point elsewhere. Rod --

All, We are in the process of trying to setup our network to use Verizon's SIP "trunking" product. They say that since Asterisk is not on their certified list of approved devices, we need to go through a field trial to get it approved before allowing us to use their service. Where we are at is getting the design approved. We are trying to watch our budget at the same time. We

Anyone tried installing Asterisk in a AWS server? \\||/ Rod --

I have a rather simple setup running under Asterisk 1.4. I'd like to move it to a new install of 1.6. Before I bring it online are there any gotchas I should look for? A Gotcha README would be better but searching with Google and the forums, for me, gets hits that deal with hardware issues -- cards etc. Nothing about depreciated/changed commands. TIA, Rod --

As I said in a previous reply the server is going great. In fact I can even send mail via it. (On the really old server I'm moving from I couldn't get authentication for outbound to work.) I now have a couple of small issues to deal with before moving completely off the old system. Virtual domains aliases? My reading seems to indicate that Postfix only handles aliases in one

I was thinking about all the old email I have and don't want to get rid of and how much space it is taking on my mail server. The idea came to me to install Dovecot on a server at home then I could drop mail into I want to save. It doesn't need to receive mail except what that is moved to it. Anyone see problems with doing this? Thanks, Rod --

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

Hi Having my first go at compiling Asterisk from cvs source. Compiled and installed zaptel ok Running make asterisk returns the following error message /usr/bin/ld cannot find -lssl collect2: ld returned 1 exit status The last part of the compile messages on screen are- editline/libedit.a db1.ast/libdb1.a stdtime/libtime.a -ldl -lncurses -lm -lresolv -lssl There is obviously something I have

I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal

Is anyone succesfully running Asterisk behind verizon residential DSL? I seem to be having some problems with my Asterisk server switching to Verizon. I'm attempting to do some troubleshooting, but I'm really interested in knowing of anyone's setup that already has Asterisk working with Verizon residential DSL. Thanks AJ ------------------------------------------------------ This

I am running samba version 1.9.18p4 on a sinix machine (unix svr4). My NT workstations use the german code page 850. In earlier versions of samba (1.9.17p2) the character mapping worked just fine, using character set = iso8859-1 client code page = 850 in my smb.conf. Since upgrading to samba 1.9.18p4 this don't work any more. Is this a known problem? I tripple-checked my smb.conf entries,

Hi! updated to p4 the other day, and now the "character set = iso8859-1" option doesn't work anymore. I didn't change anything in the smb.conf file but this: unix password sync = True time server = yes and it worked in 1.9.18p3 :/ Has anyone else seen this? Samba runs on FreeBSD 2.2.6 from the ports collection. (I don't think that password sync works for this