Displaying 20 results from an estimated 80000 matches similar to: "res_motv: Request for Comment"
2004 Jan 24
4
retrans_pkt: Maximum retries exceeded on call
Hey,
I'm getting an odd message in my logs, and have'nt been able to find much information on it:
Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call 6010532c6fedf9be383872e07e4be70c@192.168.1.2 for seqno 102 (Request)
I'm running asterisk with a Cisco 7960G
If anyone know's why i'd get this.....Any help would be appreciated!
2003 Apr 16
5
SIP Proxy
Hi,
Is Asterisk (or can it be set up as) a SIP proxy?
Thanks
--
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2005 Oct 06
14
www.openpbx.org
Hello,
What do you think of this project www.openpbx.org ?
Something like ser and openser !
Kinds Regards
Harry
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
T?l?chargez cette version sur http://fr.messenger.yahoo.com
2004 Apr 05
2
Disambiguating incoming IAXTel calls
I have two 1-700 numbers from IAXTel. Both get registered from the same
Asterisk server. I can make and receive calls on each without any
difficulty. What I can't figure out how to do is route the incoming calls
differently based on which 1-700 number is dialed. I must be missing
something obvious.
Thanks
-brian
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2005 Feb 02
2
different IAX ports for different contexts
I have a problem with my asterisk@home installation (configured with
AMP)
My question is this, can you have different ports for different contexts
within IAX?
[Faktortel]
port = 5036 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
allow=all ; Allow all codecs
register => XXXXX:XXXXX@iax.faktotel.com/EXTEN
2006 Feb 15
4
SPA-941 stutter tone
I dont recall the SPA-941 playing a stutter tone in the previous firmware
but it is driving me nuts, anyone know where to turn it off?
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - <mailto:kerryg@techdatapros.com>
kerryg@techdatapros.com
<http://www.techdatapros.com/> http://www.techdatapros.com
2004 Jan 21
4
What technology could my phone company be using?
I live in New Brunswick Canada. The phone company is Aliant. When
you set up business service here, you can go with either analog or
digital lines. This isn't a T1 or ISDN. They are talking individual
lines direct to handsets that they provide. They offer the digital
option with even very small ( 2 - 4) number of lines.
What technology could this be? Is there any way to connect such a
2004 Jan 20
2
How to diagnose "pops" and "clicks"?
My setup is as follows:
Handset -> Sipura SPA 2000 -> Asterisk -> VoicePulse
and
Handset -> Sipura SPA 2000 -> Asterisk -> Digium X100P -> POTS
I notice when making VoicePulse calls (but *not* POTS calls through the
X100P) that there is significant "popping" and "clicking" on the line.
This isn't enough to interfere seriously with the call, and
2004 Mar 27
1
AGI crashes asterisk
I configured agi-test.agi on extension 111 when i dial into asterisk
extension 111 using a IAX softphone and hangup while the AGI is playing
asterisk crashes. Does anyone have any idea why this happens.
--
regards
Vikram (http://www.vicramresearch.com)
2004 Mar 17
4
can't logon to voice mail - bad password
I have one SIP extension that can't logon to voicemail. The log file says
-- Incorrect password '3213' for user '4035' (context=other)
even though the context in voicemail.cnf says
4035 => 3213,Bill Smith
Thanks!
Paul Mahler
mail:pmahler@signate.com
phone: 650.207.9855
fax: 877.408.0105
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2005 Feb 07
2
Pro biz Asterisk
Dear All,
.
After installing, testing and like many other I found that Asterisk is reliable and a great Open source telephony solutions for a professional use. I would like to create a business with *, offering it as a IP PBX solution to customer, as a server, whatever with some Digium HW and SIP phone, .. etc.
Asking for something that might have been asked.
What are the implications for
2004 Jan 21
11
Digium X100P for $43
Digium X100P / new cards are is available on ebay for $43.
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309
<http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309
>
Hope this helps to who want to play with X100P! Are these being sold by
Digium ? I don't know ??
- SamW
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2020 Apr 06
1
Zoom?
On 4/6/20 9:33 AM, mark wrote:
> On 2020-04-06 03:57, J Martin Rushton via CentOS wrote:
>> If you have any choice in the matter I would suggest a read of
>> https://en.wikipedia.org/wiki/Zoom_Video_Communications#Criticism
>> before using or installing Zoom.
>>
> I can argue against zoom, and have, but organizations I belong to or
> join, like this past
2004 Jan 24
3
Grandstream 100 sidetone
For people who are using GS 101, what do you think the sidetone
generated by the phone.
I find mind a bit annoying. It has a delay and you notice it as an echo.
The volume of the sidetone is also quite hight. I am distracted when
both caller and called party talking over each other occasssionally.
The volume of the sidetone can be turned down using the volume button
but it also control the
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
2010 Apr 14
1
Exporting nfs share with glusterfs?
Hi all,
Is it possible to use a nfs share to serve as a glusterfs server process?? For
example: I have a host that mounts a nfs share from a solaris server under /mnt. Can
I use this /mnt directory for a glusterfs server??
Thanks.
--
CL Martinez
carlopmart {at} gmail {d0t} com
2005 Mar 09
3
voicepulse "silence" during conversations
Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter )
of voicepulse. For me, it works perfectly, but one of my customers
noticed a small problem: During a conversation, when the otherside
isn't talking, it's almost like the mic turns off.
Not that big of a deal I know, and the more I think about it, the more
this seems a voicepulse issue. But in the off
2007 Jul 24
4
Possible Bug
I''m trying to test some code that has validates each and I''ve got a very
strange failure
Mock ''Book_1027'' expected :store_with_privacy? with (#<Clip:0x1a99b96
@name="Clip_1025">) but received it with (#<Clip:0x1a99b96
@name="Clip_1025">)
The Spec
it "should check that a book can save a clip" do
@user =
2005 Oct 12
8
parameters documentation
Another trivial question:
Is there a "place" where all the parameters are documented ?
In example (my example!) I would like to know the meaning of a lot of
parameter that can be used in sip.conf,
A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060;
context=xxxxx) but other are not (at least for me)
i.e.:
type = peer, friend
insecure=very
host=dynamic
and so on.
2015 May 25
4
Name based SSH proxy
On 25/05/15 09.51, Damien Miller wrote:
> I'm not sure it should be part of the banner exchange, though there is
> no other trivial way to do it and maintain backwards compatibility.
Even if backwards compatibility wasn't a requirement,
I don't see any better way it could be done.
> I don't much like it because it reveals host identity information
> in the clear.
So