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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 development project (in a nutshell, the card receives analog calls while the accompanying software handles automated prompts, etc). The Dialogic card is not SIP-based but will work with an analog line, so I'm looking into adapters that act

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?

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 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

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

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.

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. > > Do you mean a device in sip.conf, using bindaddr to set the address to > bind for that device? We've only used bindaddr in the

I don't know in that specific output what happened. Your best course of action is to add further logging or step through the logic with all of the knowledge you have of the RTP streams to understand what is happening. On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like

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

Hi Joshua, Thank you for that. From the code it kind of looks like STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) { ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n", Our call shows: #

It's probably best if you read the logic[1]. There's an entire comment that talks about how it works. [1] https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Could you confirm if the 5 second period for learning a new audio stream > is a minimum

Hi I am looking for a source for an OpenVox card. Has anyone purchased through http://www.voiplink.com or do you normally use another vendor or OpenVox.cn directly? thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090705/6d90b996/attachment.htm

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. Do you mean a device in sip.conf, using bindaddr to set the address to bind for that device? We've only used bindaddr in the [general] section before, but if it will work in a device that could be the answer. On Fri, 23 Oct 2020 at 00:13, George Joseph

I installed SPA942 and SPA2101, and experimented with TFTP and HTTP provisioning. It all went smooth for many hours. But then all of a sudden it stopped reading configs from both from TFTP and HTTP. Now I am trying to troubleshoot and cant't find the problem. Once in a while, it does read from TFTP and/or HTTP, but then again, stops reading at all. My other phones, i.e. Grandstream and Aastra

Hi David, Gluster supports live rolling upgrade, so there is no need to redeploy at all - but the migration notes should be checked as some features must be disabled first. Also, the gluster client should remount in order to bump the gluster op-version. What kind of workload do you have ? I'm asking as there are predefined (and recommended) settings located at /var/lib/gluster/groups . You

Hi Joshua, Could you confirm if the 5 second period for learning a new audio stream is a minimum or a maximum? The unusual call flow in question results in Asterisk learning a new audio stream when we don't want it to, and having a minimum of say 2 seconds of audio would help avoid this. Thank you! On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp <jcolp at sangoma.com> wrote: > On

Hi David, On Dec 24, 2019 02:47, David Cunningham <dcunningham at voisonics.com> wrote: > > Hello, > > In testing we found that actually the GFS client having access to all 3 nodes made no difference to performance. Perhaps that's because the 3rd node that wasn't accessible from the client before was the arbiter node? It makes sense, as no data is being generated towards

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, Voisonics is hiring a VoIP support engineer to assist our customers running Asterisk based hosted PBX platforms. This is a part-time contract work-from-home position. For communication reasons we're looking for someone in a timezone encompassing Far East Asia, Australia, New Zealand, Canada, the USA, and Mexico. If you are not physically located in that area please do not apply -

Hello, Voisonics is hiring a VoIP support engineer to assist our customers running Asterisk based hosted PBX platforms. This is a part-time contract work-from-home position. For communication reasons we're looking for someone in a timezone encompassing New Zealand, Canada, the USA, and Mexico. If you are not physically located in that area please do not apply - being "flexible"