Re: [asterisk-users] if statement recording - after hours

There are multiple ways to do time-of-day routing. ExecIf w/ IFTIME, GotoIfTime, and ExecIfTime. I put some examples below. Sincerely, Brian LaVallee On 9/12/14, 10:05, Eric Wieling wrote: See ExecIf in the output of core show applications. The IF function might be useful, see core show

[asterisk-users] FYI: Block Comments

) ; -- The parser stopped loading anything past the above mistake -- ; -- Missing that space started a block-comment - Arghhh! -- exten = _4X.,1,NoOp(This would NOT load either) ; -end Guess I have to change my highlight syntax, avoiding dashes in the future. Sincerely, Brian LaVallee

Re: [asterisk-users] Error: 'LENGTH' is not a recognized built-in function name

On 8/11/14, 11:31, Matthew Jordan wrote: On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat deepaksingh.ra...@gmail.com wrote: On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat deepaksingh.ra...@gmail.com wrote: Hi, I modified the query in res/res_config_odbc.c. Original: SELECT

Re: [asterisk-users] The plain old PBX functionality

the digital PBX features you're looking for, will involve two groups of settings. Configuration on the server -and- configuration on the phone. SIP phones are NOT dumb terminals, you have to configure them to operate how you want. Sincerely, Brian LaVallee

Re: [asterisk-users] OPTIONS Request without username - Forbidden

: INVITE,ACK,CANCEL,BYE,REGISTER,PRACK,UPDATE snip Accept: application/sdp Sincerely, Brian LaVallee On 6/25/14, 11:30 PM, Rafael Visser wrote: Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's

Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info

to manipulate the ISDN message via SIP, it all comes down to how the gateway handles the desired functions. Sincerely, Brian LaVallee On 6/26/14, 11:24 PM, Positively Optimistic wrote: We're using a Earthlink PRI converted to SIP via a MediaGateway. I assume the mediagateway will convert the headers

[asterisk-users] Multiple Servers: Multiple Peers: call-limit

. But, hoping there might be a simpler solution. Sincerely, Brian LaVallee -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] T1 Card RED ALARM

Since there are a number of setting that could be causing the alarm, AMI/B8ZS, SF/ESF, etc... Start with a loop-test, make sure the card can communicate with itself (using the current settings). Connect the following pins: 01 (RX-) -- 04 (TX+) 02 (RX+) -- 05 (TX-) Sincerely, Brian LaVallee

Re: [asterisk-users] Play announcement only once in a Call Queue after 10 seconds

Hi Jonas, While I don't work with queues, but you could playback announce-holdtime before putting the caller into the queue. exten = _X.,1,NoOp(Post Queue Announcement) same = n,Answer() same = n,Wait(10) same = n,Playback(announce-holdtime) same = n,Queue(real_queue) Brian On 6/25/14,

Re: [asterisk-users] Maximum number of users

connecting a sufficient number of PSTN connections to support those users. Sincerely, Brian LaVallee On 12/18/13, 11:45 PM, bilal ghayyad wrote: Hello; Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later? Regards Bilal

Re: [asterisk-users] Multiple IAX2 Trunks Load balancing

Are you looking for something like this? Note: This will continuously go between the two trunks until the caller hangs up, can be fixed by adding loop counter. ; ; extensions.conf ; [LOADBALANCE] exten = _X.,1,NoOp(Connect to least used trunk) ; - show active count exten = _X.,n,NoOp(Calls:

[asterisk-users] ARA: realtime: sip.conf: context

for extensions and does NOT work on the context field of the sippeers table, is there any field that can be used? Sincerely, Brian LaVallee ---=== ;# extconfig.conf ; [settings] ; sippeers = mysql,database,sippeers moresippeers = mysql,database,moresippeers extensions = mysql,database,extensions other

[asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3

a small unit that handles one or two DS3's. The advantage comes when you add the 29th DS1. With VT1.5 it's just adding a single channel, DS3 will require another whole DS3 to get an additional DS1. Sincerely, Brian LaVallee From: Nick Khamis sym...@gmail.com Reply-To: Asterisk Users Mailing

[asterisk-users] Initial REGISTER Request: Contains Credentials before 401

community know how to avoid sending the credentials until AFTER receiving a 401? Any suggestions would be appreciated! Sincerely, Brian LaVallee # === # sip.conf # Asterisk 1.8.15-cert1 # --- ; [general] ; ; - trucated ; register=accountnum

[asterisk-users] Initial REGISTER Request: Contains Credentials before 401: KDDI Japan

-users@lists.digium.com Subject: Re: [asterisk-users] Initial REGISTER Request: Contains Credentials before 401 Brian LaVallee wrote: My SIP provider is not happy that credentials (in the Authorization header field) are provided in the initial REGISTER request. The SIP provider ONLY wants

Re: [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database

It sounds like phpMyAdmin is NOT on the same server as the Asterisk DB. You will run into a couple possible issues when allowing remote MySQL access on the Asterisk server, You will need to set the MySQL user privileges to a specific host or a wildcard (%). Most common issue is the firewall,

[asterisk-users] qualify=yes: OPTIONS: How to Change?: `From: asterisk`

qualify enabled without sending the other end any reference to asterisk. Can anyone point me to a setting that will change or remove `²asterisk²` from `FROM:` in the OPTIONS message? Thanks, Brian LaVallee -- /etc/asterisk/sip.conf (Asterisk 1.8.15-cert1) [general] ; - Truncated [TRUNK

[asterisk-users] Thanks! qualify=yes: OPTIONS: How to Change?: `From: asterisk`

Thanks Jeremy! On 5/9/13 8:21 PM, Brian LaVallee wrote: When qualify is enabled on a trunk, the From line shows asterisk. See the SIP message below. I had the same annoyance/issue. fixed it in https://issues.asterisk.org/jira/browse/ASTERISK-17616 That's looks like the problem I