On Wed, 2005-01-26 at 15:59 -0500, Paul Dugas wrote:
I've got a single IAXy installed in a little office nearby and got a call
from someone on site a finew mintues ago. Apparently they couldn't make a
call on that extension. They'd pick up the phone and get nothing; no
dial-tone.
Has
We're using Asterisk 1.6.1. When our SIP clients have silence
suppression turned on, it's a problem for many apps. Is there a
workaround for this in Asterisk? Other than turning silence
suppression off in the SIP client, is there anything I can do on the
Asterisk side to make things work
We are trying to get Asterisk to behave correctly when our SIP clients
have Silence Suppression turn on, but are not having any luck.
Basically, there are several apps in Asterisk which won't send any
audio to the SIP client, unless the SIP client itself sends audio to
Asterisk (which it
We have a customer trying to dial through our server, and our server is throwing tons of these log messages:
Jul 27 14:21:02 NOTICE[29210]: rtp.c:431 ast_rtp_read: RTP: Received packet with bad UDP checksum
Is it pretty certain, that these are caused by a bad or misconfigured router along
We are using Asterisk 1.0.7. We have this scenario:
IAX2 user comes in to Asterisk, dials an extension, and transfers to a SIP user.
The dial command is simple, looks like this:
exten = 300,1,Dial(SIP/300)
Extension 300 is a legacy PBX device operated by touchtones. The user (coming in
If anybody here is a sixTel customer, can you share any tips tricks for getting in touch with anybody there? They are absurdly hard to get a hold of, particularly when you have a technical issue needing to be resolved. If anyone has any phone numbers other than their main 800 line, I'd sure
We are tracking the following situation:
SIP client connects to our Asterisk server, and then connects to another SIP user. Re-invite is OFF, so Asterisk is in the middle of the whole conversation.
When one SIP client sends DTMF tones, the SIP client uses RFC2833 to send the tones to the
), the RFC2833 packets (length 16) never arrive.
Weird?
On Thu, 2005-06-02 at 13:01 +1200, Matt Riddell wrote:
Bryan Field-Elliot wrote:
The problem is, the other SIP client is never receiving the RFC2833
packets. An ethereal trace on the same local network shows that the
regular conversation UDP
I'm trying to gauge the amount of overhead for idle users (NOT in the middle of a phone call) per user, per server. These are a combination of SIP and IAX2 clients, with qualify=yes.
On, for example, a dual 2.4 Ghz Pentium server (with plenty of RAM), how many hundreds, or thousands (rough
We are using TDM40B's to connect some devices to Asterisk which depend on caller information arriving as ANI, rather than as Caller ID. I am unsure if the TDM40B supports this in the first place, and if so, I am unsure how to configure it so. I've searched the wiki but couldn't find anything.
We have a script which executes asterisk -n -r -x . periodically against the running server, to check the status of a few things, and pipe the output to a file.
With prior versions of Asterisk this worked fine, but having just upgraded to 1.2.9, we are finding that if the output is
We've been seeing a pattern over the last couple of weeks with our Asterisk servers (1.0.9). The number of ports in use (RTP) seems to grow by two every minute or so. Eventually the server will run out of allowable files open and crash. We are resetting the server once per day to prevent this
For the typical home user who has a SIP ATA behind (usually) a Linksys home router/firewall, what's the best practice qualify= time we should be running on the server, to keep the home user's NAT happy?
The default, 2 seconds, is way too short (generates too much net traffic).
I am wondering
We have several Asterisk servers (1.6.2.15) all configured for Realtime, all
backed by the same database. The Asterisk servers are all listed under DNS SRV
records, and SIP ATAs find us this way.
Normally, no matter which Asterisk server an ATA connects to, we get our
database fields filled
2011 kl. 00.26 skrev Bryan Field-Elliot:
Normally, no matter which Asterisk server an ATA connects to, we get our
database fields filled out correctly, such as regseconds, lastms,
ipadr, etc. However, with some ATA's we are experiencing a problem as
follows:
1. ATA reaches its re
On Jan 4, 2011, at 12:26 PM, Tilghman Lesher wrote:
It wasn't designed to do this. While you can have the same sippeers table
for multiple servers, you really should have a separate sipregs table for
each backend server. The reason why is that some mappings depend
implicitly on the host to
We are using PostgreSQL real-time connector (res_config_pgsql) with Asterisk
1.6.2.15.
From time to time, we need to reset our PostgreSQL server, causing all active
DB connections to close. While other packages in our system re-connect
gracefully when this happens, Asterisk appears to not
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