Moderators: enjay, williamconley, Staydog, mflorell, MJCoate, mcargile, Kumba
Kumba wrote:What SVN revision are you on? If you aren't on at least SVN revision 2832 then you do not have Asterisk 13 support.
Kumba wrote:If you look roughly 8 lines down from when you connected to the Asterisk console you'll see the relevant warning/error message. That error message is "No path to translate from SIP/xyz". This means that the codecs that asterisk has available to it are not compatible with the codecs the phone/carrier is trying to connect with or with the codecs the phone/carrier entry is configured to use.
So basically you have a codec issue. Almost 99% of the time this relates to the G729 codec. Either you need to install one, or the previous codec you installed was for Asterisk 11 and not for Asterisk 13.
You can double check this by making sure the ulaw codec is enabled in the carrier/phone entry you are testing by adding 'allow=ulaw' to it's configuration and removing the 'allow=g729' section.
Kumba wrote:The timeout on non-critical SIP messages also indicate some sort of firewall issue, likely preventing 2-way SIP messages.
Kumba wrote:I'm already working on ViciBox v.8.1 which ships with Asterisk 13 by default. Probably be done in a week or so.
williamconley wrote:Kumba wrote:I'm already working on ViciBox v.8.1 which ships with Asterisk 13 by default. Probably be done in a week or so.
Could I infer that it will have ViciPhone in it since it has Ast 13?
dspaan wrote:I'm confused, with AMD you refer to answering machine detection?
I'm very interested in the status of virtualization as well. I would really love to clone a server for instance and simply the fallback options that it provides. A lot of clients also are worried that the software is run on dedicated servers.
vkad wrote:We have for the VMs
-a single DB server (24gb ddr4, 2.2ghz 8 cores) (handles on average 3000-6000 queries per second when the shift is active)
-a single web (running PHP7 with opcache) + telephony server (agent only server; no dialing allowed; asterisk 13.19.0-vici) (8gb ddr4, 2.2ghz 4 cores)
-5 x telephony servers for balancing calls (the dialer connects to these servers through IAX and places calls) (4gb ddr4, 2.2ghz 2 cores) We use a mix of Asterisk 13 and asterisk 11 here, since these are not connecting to the agents they don't need to be asterisk 11. However, we need the AMD from either ASTERISK 11 or the sandbox version ASTERiSK 13.19.0 so we use a mix for now.
All of the above servers are kept as virtual machines a single E5 v5 + 64gb ddr4. This bare metal has a clone running in High Availability so everything is replicated across the bare metal (we use proxmox so that handles the HA)
So even if the physical machine goes down, we are safe.
frequency wrote:Are you recording calls as well? If yes, are you using an archive server or are they recorded on web server?
williamconley wrote:frequency wrote:Are you recording calls as well? If yes, are you using an archive server or are they recorded on web server?
I can't say how vkad does it, but I can say that the recordings are MADE on the dialers and then pushed to an archive server that has NO requirements of even being part of the Vicidial cluster. Any FTP/Web server will do.
An FTP server that also has Web is ideal so the files can be pushed via FTP, served via Web and the links can be updated in the Vicidial DB to point to the new location and as of that moment they are no longer a burden on the Vicidial cluster.
But they are made on the dialer unless they are made on an intercept SIP server that's passing the audio through (which is overly complex, but doable).
Kumba wrote:I'm already working on ViciBox v.8.1 which ships with Asterisk 13 by default. Probably be done in a week or so.
WARNING[3192][C-00000004]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
alo wrote:pattern match timed-out at /usr/share/astguiclient/AST_update.pl line 470
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