similar to: IAX2 trunk on IPV6

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

Hi Is there any way to change the preferred audio playback format in asterisk (I'm using 1.8.25.0) i.e. first check for gsm, if doesn't exits then check for slin? Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex

Hi I'm using asterisk 1.8.25.0 on CentOS 6. I have compiled it with all the calendar modules: *CLI> module show like calendar Module Description Use Count res_calendar.so Asterisk Calendar integration 4 res_calendar_ews.so Asterisk MS Exchange Web Service Calenda 0 res_calendar_caldav.so

The Asterisk Development Team has announced the release of Asterisk 1.8.25.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.25.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.8.25.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.25.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Howdy, I'm setting up several gluster 3.12 clusters running on CentOS 7 and have having issues with glusterd.log and glustershd.log both being filled with errors relating to null client errors and client-callback functions. They seem to be related to high CPU usage across the nodes although I don't have a way of confirming that (suggestions welcomed!). in

Thanks Milind, Yes I?m hanging out for CentOS?s Storage / Gluster SIG to release the packages for 3.12.1, I can see the packages were built a week ago but they?re still not on the repo :( -- Sam > On 18 Sep 2017, at 9:57 pm, Milind Changire <mchangir at redhat.com> wrote: > > Sam, > You might want to give glusterfs-3.12.1 a try instead. > > > >> On Fri, Sep

Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sam, You might want to give glusterfs-3.12.1 a try instead. On Fri, Sep 15, 2017 at 6:42 AM, Sam McLeod <mailinglists at smcleod.net> wrote: > Howdy, > > I'm setting up several gluster 3.12 clusters running on CentOS 7 and have > having issues with glusterd.log and glustershd.log both being filled with > errors relating to null client errors and client-callback

Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten => s,n,Dial(SIP/peer1/number at domain2.com,60,r) [peer1] type=friend host=domain1.com fromdomain=domain1.com As a result in SIP packet uri: number at domain2.com@domain1.com I need: number at domain2.com I can't use "SIP uri dial", i need

FYI - I've been testing the Gluster 3.12.1 packages with the help of the SIG maintainer and I can confirm that the logs are no longer being filled with NFS or null client errors after the upgrade. -- Sam McLeod @s_mcleod https://smcleod.net > On 18 Sep 2017, at 10:14 pm, Sam McLeod <mailinglists at smcleod.net> wrote: > > Thanks Milind, > > Yes I?m hanging out for

We're using Asterisk 14.7.6 and I have a dialplan that ends like this: same => n,Dial(SIP/${EXTEN:0:4}@peer1) same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) same => n,Hangup() When peer1 hangsup, the priorities after the Dial are executed fine. But when the caller hangsup during the Dial, the cleanup steps aren't done. Why? I did read "Note that on a successful

I have considered opening a bug report on this, but wanted to get some feedback and make sure I am not missing something in the way of a simple work around. What is the scenario in which this impacts your implementation? Ours is the desire to use the same realtime SIP database for many asterisk servers, and route the call based on a "home server" value in the realtime database. The

I have been working through the book "Applied longitudinal data analysis: modeling change and event occurrence" by Judith D. Singer and John B. Willett. I have been working examples using SAS and also using it as an opportunity for learning to use R for statistical analysis. I ran into some difficulties in chapter 8 which deals with using structural equation modeling. I have tried to

Thanks, Anthony. I added both 'g' and 'F' options. Now, when the caller hangs-up, my cleanup code is run by both the caller channel and the peer channel, but I only want the caller channel to do that. Also, when the peer hangs-up, there is no execution of the priorities following the Dial. Finally, is there a way to reset all globals, maybe as a variant of "dialplan

Take an example where there is two sip users defined in sip.conf as follows; [peer1] Host=192.168.1.1 ... [peer2] Host=dynamic Secret=password ... [Peer3] Config not relevant ... The intention is to accept calls from peer1 without authentication (ip address authentication only), but require authentication from peer2 If by chance a SIP invite comes "From"

Hi, I too have found this matching to be frustrating. I would like it to behave as you describe. Doug -- Doug Meredith 506-854-7997 ext. 801 ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon Estep Sent: Thursday, January 04, 2007 1:50 AM To: Asterisk Users Mailing List - Non-Commercial

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

Hello I have two peers setup and working with x2 bricks each. They have been working via tcp for the last 4-5 months. I just got two Infiniband cards and put the on the peers. I want to change the transport type to rdma instead of tcp but I don't see an easy way to do this. Can you please help me with proper instructions. Best Regards Ivan Dimitrov

http://bugzilla.mindrot.org/show_bug.cgi?id=1282 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |RESOLVED Resolution| |WORKSFORME CC|