similar to: 'Bookmarking' a place in a sound file

Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412121

Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

Hello, If you have canreinvite=no and a peer sends you a re-invite, what will Asterisk reply with? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180

Hello all, I have a problem where problem with one way audio, and I think it's related to "a=sendonly" and a re-invite. Can anyone please assist? The scenario is as follows.... - We send an INVITE to a peer, and it replies with a "100 Trying", and then a "183 Session Progress" message containing "a=sendonly". - Asterisk plays the caller music on hold,

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

have adaptors compatible with Asterisk, but explicitly say in the product titles that they're unlocked, which I think is the key. On Thu, Dec 17, 2009 at 4:16 AM, Brian Cline <Brian at nw.brian.fm> wrote: > Hello, > > I'm running Asterisk v1.6.1.11 internally with a few Linksys SIP > phones and will be receiving a machine containing a Dialogic card > for a

Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. It comprises of an online service run by us, and a lightweight and easy-to-install client on your side. Specifically of interest to Asterisk users is the monitoring of SIP registrations, and automatic blocking

Sipml5 works. You need to have TLS enabled on asterisk web socket. Mitul On 12-Mar-2015 12:47 PM, "David Cunningham" <dcunningham at voisonics.com> wrote: > Hello, > > Can anyone recommend a particular online WebRTC phone for testing with > Asterisk? > > We tried: > > - JsSIP, but even with the "enable video" checkbox disabled it sends video >

Hello, We have servers running Asterisk 1.8.20.1 and 11.7.0, and on both setting maxsecs in voicemail.conf doesn't seem to have any effect. A voicemail keeps recording after the specified time, and when the caller hangs up the voicemail is saved in the mailbox. Are we doing something really silly? Here's the voicemail.conf. We have tried 'voicemail reload' and restarting

Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was

Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> wrote: > Hello, > > Does anyone know a way with chan_sip to tell Asterisk to use a specific IP > address for its end of the communication for a specific

Hi Dovid, We can change the SDP in Kamailio, but Asterisk will still send its RTP from its default address. The remote end is strict about accepting RTP from the specified source and won't accept it. Have you any suggestions to solve that problem? Thank you. On Fri, 30 Oct 2020 at 14:49, Dovid Bender <dovid at telecurve.com> wrote: > Why not use OpenSips/Kamailoo in between?

Hi all, We have a project involving voice recognition, and will need a vocabulary of 10,000 words (actually names). Can anyone recommend a product that works with Asterisk? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed...

Hi everyone, We have an Asterisk 1.4.17 user who has problems with sometimes not getting a ring tone on the calling phone. We're considering setting progressinband = yes, but would like to know how much extra CPU load this will require? If anyone can give something even roughly specific (eg "30% increase") that would be great, rather than just "lots". Also, are there any

Hello all, We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups which previously worked fine have stopped working. Can anyone advise if there has been a change in how pickups work? Here is an example where 1000101 is trying to pick up a call to 1000103: <SIP/product-local-00000005>AGI Rx << EXEC Dial "Local/1000103 at product-pickup

OK, thank you George. On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote: > > > On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hi George, >> >> Thank you for the response. I'm a little unclear on what you mean by a >> transport. We're using chan_sip, not pjsip.

Hi David, It seems like a network issue to me, As it's unable to connect the other node and getting timeout. Few things you can check- * Check the /etc/hosts file on both the servers and make sure it has the correct IP of the other node. * Are you binding gluster on any specific IP, which is changed after your update. * Check if you can access port 24007 from the other host. If

Hello, Does anyone know a way with chan_sip to tell Asterisk to use a specific IP address for its end of the communication for a specific device? Something like: [device] type = friend host = 11.22.11.22 ouraddress = 33.44.33.44 This is for use on a server with multiple IP addresses. There is the "extenip" setting, but it's really designed for NAT, and can only appear in the

Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x