Displaying 20 results from an estimated 1000 matches similar to: "Trunked IAX or not"
2005 Feb 06
3
iax2-jitter-trunking?
Two cvs-head asterisk boxes with iax2 working fine (without register
statements).
When two calls are placed simultanously from system A -> B and the packets
are sniffed on the wire, I see the two calls using two different udp
packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes
(at both ends).
I was expecting to see both calls handled within a single udp packet,
but
2005 Jan 21
3
IAX Inbound Sound Quality
I have a couple of DID's through VP Connect and have been having sound
quality issues on incoming calls. During the call, the calling parties
voice sometimes sound like it is crackling, in other words it is not
very crisp. I would liken it to listening to a radio with a blown
speaker. This sound defect comes and goes throughout the call. The
other person is always audible but it just isn't
2005 Feb 14
5
Sipura g729 call quality to PSTN
If this has been covered before - I appologize.
We use some Sipura SPA-2000's with the g711 codec and all seems fine
(except for the occasional failure to register errors in my asterisk
logs - but I will save that for another post).
g711 call quality is on par with our Cisco 7960's. However, when
using the g729 codec, the call quality on the Sipura device goes
downhill on the PSTN side
2005 Jan 24
2
LiveVoip DTMF Issues
I have a couple of DID's with LiveVoip and am having major DTMF issues
on incoming calls. I am connecting to them through IAX using ULAW.
When someone dials one of these DD's (from a landline) they are for
the most part unable to navigate the IVR menu successfuly. I would say
the failure rate is greater than 80%. For example if the caller
presses 5 sometimes * will see the DTMF as 55 or
2005 Feb 15
2
Asterisk, inband DTMF send by a GSM mobile
Hi all,
I use a GSM device to send dtmf on my asterisk system (via SIP).
the codec I use is ulaw (or a-law).
dtmf mode is INBAND.
relaxmode is on.
but most of the case, I 'missed' some DTMF or
I 'double' one.
as anybody as seen this before?
is there any way to prevent this
thanks
2005 Feb 12
2
soho fax suggestions?
Need to replace our older soho fax machine with something more current.
Would like to run the fax line through *, but haven't been able to
make spandsp work correctly with digium TDM04b card. Our fax volume
is very low (maybe a few per week), but we have multiple offices in
three geographic locations and would like to be able to email the
images to the correct location.
For planning purposes,
2005 Feb 04
3
Callerid problems with 1.0.5
Skipped content of type multipart/alternative
2007 Jul 23
2
IAX Encryption
I am playing around with IAX encryption and have had good success.
I read somewhere, that trunked packets are not encrypted. Does
anybody know if this means the trunk packets themselves are not
encrypted but the voice frames in them are encrypted or does this
mean that if you are using trunking then encryption of the voice
frames will not occur. I have used Wireshark to sniff the packets
and it
2005 Jan 25
2
DTMF digit dropping
I run an automated information retrieval system, using Asterisk. Fairly
often the system misses a dialed digit. Our codes are all 4 digits, see
lots of logs with:
4199 - OK
530 - Invalid code
330 - Invalid code
5330 - OK
As callers experience skipped codes. We're using Broadvoice SIP with
inband DTMF (and we've tried every possible setting or option
2005 Jan 26
1
How to make channel busy signal?
When I make a call over the Internet and call myself IN over POTS my
phone rings to outside party but I can not hear it.
Why isn't my channel extension indicating busy status when I'm making
call over Internet? This way I could ring my next extension with n+101
priority.
I'm using Sipura-3K unit.
--
#Joseph
2005 Mar 08
1
CallerID - Broadvoice vs. VoicePulse
Until recently, I was using Broadvoice for my in/out calling thru
Asterisk. I was extremely pleased to see that Broadvoice was actually
passing the callerid info (number and text) that I had set up on each
device in my SIP.CONF file. I had PSTN users tell me that they were
actually seeing name and extension info when I called them from the
Asterisk box.
Last week, due to numerous user quality
2005 Jan 24
1
Short DTMF Tones and Asterisk
I'm having a very annoying problem with access my asterisk system from
work. Our phone system here only produces very very short DTMF tones.
The phones work fine for other IVR systems (Dell Support, HP Support,
etc, etc). However, tones to Asterisk just never make it.
The way I'm calling into my Asterisk server is such:
OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound
The
2005 Jan 24
4
Is Voice Pulse Connect good ?
Hi,
I am thinking of signing up with voice pulse connect to connect to my
asterisk server and using it as a regular line. Is it good? Or should I go
with vonage or others ?
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2006 Nov 07
0
astertest
Hi all!!
I've made some changes to the applications that Astertest was using to
monitor the performance of the server. Now is also possible to track the
bandwidth usage of the server, this has nothing to do with the executable
(astertest.exe) itself but with the events that the Asterisk Manager
generates.
The method described in:
http://www.asteriskguru.com/tutorials/astertest.html
to
2008 Apr 09
3
Interface bonding?
I'm try to bond a few interfaces together with the hopes of getting
increased throughput, and I'm using a cisco Catalyst 2900 as the switch.
I've tried using mode 0, 5, and 6 with nothing special on the switch,
and mode 4 with some ports "trunked" together (I have a feeling that the
"trunking" that the 2900 does is not 802.3ad, as it disabled the ports
it saw as
2005 Jan 24
6
Damn DTMF Beeps on my calls
Can someone give me a clue as to why I keep hearing DTMF type beeps on my
phone calls. It sounds exactly like someone on the other end is pushing a
key on their phone but they are not!
Has anyone ever heard of this before? It use to happen once in a while,
today it's been happening a LOT and it's driving me batty..
--
Start Your Own ISP!
http://www.YourOwnISP.com
2007 Apr 19
1
Help Astertest - Asterisk stressing tool
Hi,
Did someone ever managed to make Astertest
(http://www.asteriskguru.com/tutorials/astertest.html) work ? I followed
all the instructions of this tutorial and corrected the mistakes pointed
by the users but it still doesn't work. I can compile it and load
app_securax_cpuinfo.so. When trying to load app_securax_serverload.so I
have this error :
WARNING[31477] : loader.c: 325
2005 Jan 27
5
iax.cc / sixtel are they legitimate?
Does anyone have any experience with iax.cc/sixtel?
Are they a legitimate company? From their website
it looks like you can get a private incoming 800
number for 30 cents/month plus 2 cents/minute.
Somehow that pricing seems a little cheap for a
DID number. I assume there has to be some minimum
usage or something. Any info as far as actual costs
and/or voice quality would be appreciated.
2005 Jul 07
1
IAX2 Trunking - CVS-Head
Hi
Is anyone successfully using iax2 trunking with CVS head ?
The reason I am asking is that I have heard there may be some audio
problems, which I would like to know about before sending customer's
calls over a iax2 trunked connection.
Thanks in advance.
Clive
2005 Mar 01
2
Important :: Please support the development of a new Jitterbuffer for SIP
Steve Kann has developed a new jitterbuffer for IAX2, that hopefully
will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable
relase.
Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs
support in the form of funding in order to take the time to test this
out and complete it in time.
Please paypal your contribution to sponsor@astertest.com today. Every