similar to: Meetme Talker Optimization

mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >

Hello, all. I see there is an "o" option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because

Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsullivan at opensourcedevel.com

Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=612 at a10, 610 at a10 However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However,

Hello, all. After many pages of googling and testing in the lab, I'm still a bit perplexed about how to implement tls protection for the asterisk manager. manager.conf allows one to specify the cert file but one normally must also specify the private key file. If I simply enter the cert file: sslenable=yes sslbindport=5038 sslbindaddr=172.x.x.8 sslcert=/etc/pki/tls/certs/pbxc.pem ; path

Hello, all. I am delightfully slogging my way through installing and configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and admiring the product but I'm having a problem getting speex to install and I would very much like to use it. It is not available in menuselect and the problem appears to be with speex_preprocess_ctl: [root at pbx01 asterisk-1.6.1.1]# grep -i speex

Hello, all. Since implementing an iptables firewall between the Asterisk PBX and several SIP phones, the Asterisk PBX ability to "reinvite" has been broken even when the phones are on the same network (i.e., no firewall between the phones). We've been beating our heads against the wall thinking it was the complex rule set but it appears the issue is ip_conntrack_sip. Before I drop

Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the "s" extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general]

Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how

Hello, all. I am attempting to use IMAP voice mail storage in Asterisk 1.6.1.1 on CentOS 5.3 using Zimbra 5.1.6. I will not be using it as it has proved terribly unstable - Asterisk segfaults on every voice mail message although the message is successfully deliver to my email inbox - but I thought I should report it. Here are the errors from the Asterisk console: -- Executing [210 at

Hello, everyone. No need to read this message. I'm posting for documentation for other poor, ignorant slobs like me who are struggling to pull together the many technologies to make converged networks happen. Hopefully, this will help save someone else the time I spent. I started the below email until I realized I had solved multiple parts of a compound problem but not all at the same time.

Hello, all. I'm having a nasty problem with call parking in Asterisk 1.6.1.1 that smells like a bug. When the call returns, it seems to be returning to a "|" delimited extension and failing. Here is the output from the console: [Jun 26 22:20:42] NOTICE[7168]: chan_sip.c:18160 handle_request_invite: Call from 'tkeeley' to extension '56' rejected because extension

Hello, all. I must be brain cramping badly on our Asterisk 1.6.1.1 installation. Our MeetMe macros are working fine except they do not prompt for a PIN. So I made a very simple conference room: exten => 7777,1,MeetMe(123456,cMaAsx,123456) Shouldn't this prompt the user who dials 7777 to enter a PIN before entering the conference room whether or not a PIN is defined in meetme.conf? I

Hello, all. I've come across a nasty problem just as we are ready to put our first system into production. During our final testing, we were plagued with several "invalid extension" or "password incorrect" messages when we knew the information entered was correct. Upon investigation, we saw that DTMF signals were occasionally but not consistently duplicated. We might

Hello, all. I'm having a deeply frustrating time getting multiple parking lots to work and am wondering what I am doing wrong. I am using Asterisk 1.6.1.1. I defined two separate parking lots in features.conf as follows: [parkinglot_a100] ; SSI context => a100-parking parkpos => 900-920 findslot => next [parkinglot_a10] ; EBC context => a10-parking parkpos => 800-820

Just a thought as we explore the brave new world of converged voice and emails. Voice mail boxes typically hold a very small number of messages while email folders contain thousands. Do we need to rethink the traditionally small limits on voice mail boxes when storing in IMAP or are the messages counted separately? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1

Hello, all. The little bit of reading I've done on lua makes me eager to give it a try. However, when I try to install it (Asterisk 1.6.1.1 on CentOS 5.3), it is not available in menuselect. I have installed lua and lua-devel. I've seen very little about it in my Internet searches. What else must I do so that it installs? Thanks - John Oh, by the way, I'm having a similar problem

Hello, all. As we work through our design issues, we are very interested in moving immediately to real time database since we anticipate expanding our system in to a clustered system within a year or two. One of the biggest disadvantages we anticipate is the lack of configuration documentation. In other words, the .conf files are self-documenting. If I want to understand my dialplan, I read

Hello, all. We have successfully connected our new Asterisk 1.6.1.1 PBX to Vitelity's network and have been very happy with them thus far. However, we'd like to use domains in our sip.conf to facilitate routing in our multi-tenant environment. We also like to set allowexternaldomains=no for security. However, this breaks our inbound PSTN calling from Vitelity. Is it possible to use

Hello, all. I see that I can use the C flag in Dial() to reset the CDRs and plan to use this for outbound calling since our carrier does not bill us until the call is answered. We'd like to do this for inbound calls from the PSTN since we pay for inbound and outbound minutes. However, it doesn't seem to make sense to do this with Dial() because we have an automated attendant which