search for: lavallee

...From line shows "asterisk". See the SIP message below. I would like to keep qualify enabled without sending the other end any reference to "asterisk". Can anyone point me to a setting that will change or remove `?asterisk?` from `FROM:` in the OPTIONS message? Thanks, Brian LaVallee -- /etc/asterisk/sip.conf (Asterisk 1.8.15-cert1) [general] ; - Truncated [TRUNK] ; - Truncated qualify=yes ; ; end -- IP 4.4.4.4.sip > 3.3.3.3.sip: UDP, length 573 ............................OPTIONS sip:server.carrier.tld SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aBcDeFgHiJkLmNo;rport M...

Hi, I'm testing: $ sudo nsd-control status version: 4.0.1 verbosity: 2 I found a loop problem with this record: * IN CNAME none ("none" means no matching record in zone and therefore match * again) Queries that use "* CNAME" will result in a loop. The response will use TCP and will be limited to 65k bytes $ dig @127.0.0.1 sdfgsfg.test.com ;; Truncated,

...quot;${DIALSTATUS}"="CHANUNAVAIL"]?SERVER01,${EXTEN},1) same => n,NoOp(yes, it's incomplete) I've though about passing the variable between the middle servers in a SIP message, side communication channel. But, hoping there might be a simpler solution. Sincerely, Brian LaVallee

On Tue, 31 May 2016 09:41:37 -0700 Michael Graczyk <mgraczyk at google.com> wrote: > UHJ is an interesting way to preserve compatibility with non-ambisonic > playback systems. However, I have not seen it generalized to higher > orders. I expect that its popularity will decrease as HOA becomes more > and more common. If UHJ becomes popular in the future, we could > specify

I have a Perl AGI script updating some values to database like recorded file path, unique ID and callerid. When I run the script with test dialplan, its not updating to database. Whereas database connection is fine, when I run agi debug I see only Tx packets not Rx packets, firewall is also OFF. Any other specific reason why there is no Rx. The same script working in one more Asterisk machine.

Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131218/7fbbc3c8/attachment.html>

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 related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

...lines into the QQnorm-Plot and the TA-plot under Gaussian distribution * stratification (2.0-2) Sophie Baillargeon http://crantastic.org/packages/stratification This package performs univariate stratification of survey populations. The main function implements a generalization of the Lavallee-Hidiroglou method of stratum construction. The generalized method takes into account a discrepancy between the stratification variable and the survey variable. The determination of the optimal boundaries also incorporate, if desired, an anticipated non-response, a take-all stratum for large...