similar to: Upgrading from 1.6.1.1 to 1.6.1.2

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 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. 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]

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'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. 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. 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

Hello, all. I've been playing with MeetMe and talker optimization seemed like a great idea. I activated it as follows: exten => 201,1,MeetMe(100201,cTo) However, although I can see who is the talker on the CLI pbx01*CLI> meetme list 100201 User #: 01 1001 Denise Dion-Sullivan Channel: SIP/1001-1e1db7c8 (not talking) 00:00:33 User #: 02 1000 John A. Sullivan III

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. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

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, 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 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, 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. 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. We are having a problem where audio for sip channels is dropping upon reinvite. Perhaps it reflects a misunderstanding of what reinvite does. We are running Asterisk 1.6.1.1 on CentOS 5.3. SIP is set to canreinvite=nonat. We have tried RTP with strictrtp set to both yes and no. We have also tried extending the Asterisk rtp port range to accommodate the differing default ranges of

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 running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email

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