similar to: Audio lost on reinvite

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. 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

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 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

On Mon, Dec 15, 2014 at 3:34 AM, Recursive <lists at binarus.de> wrote: > <snip> >> For asterisk 1.6 & 1.8 you would need to set 'canreinvite=no', I don't know what Asterisk 13 will do with this setting. >> > I suspect Asterisk 13 will just ignore it. To make things worse, there seems to be a configuration directive named reinvite (not a typo); I

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. 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. 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. 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. 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

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. Our first major asterisk system is just about ready for production. However, we noticed that our outbound SIP callers did not receive rings when dialing cell phones. Land lines were fine. We "fixed" this by setting progressinband=no in sip.conf. However, I gather this places extra load on the asterisk server and I'm not sure that it always conveys accurate status

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 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

We were able to solve the below problem. I'll post it in case someone encounters the same issue. No need to respond or even read unless you see a better way. Thanks - John We have manually set callerID on our outbound lines to reflect the appropriate DID both for e911 and to be polite to folks we call, e.g.: exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=5197546340) exten =>

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. I've solved my own problem but will post it here in case someone else has the same misunderstanding in the future. We thought we had set up our meetme so that regular users entered the conference without a pin but could not speak to each other until the moderator arrived. We enforced pin entry on the moderator . . . or at least so we thought. If the moderator waited long enough

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