Andrew Martin wrote:
- Original Message -
From: "John Novack SCII_U"
To: "Asterisk Users Mailing List, Non-Commercial Discussion"
, "Andrew Martin"
Sent: Monday, October 8, 2018 4:29:41 PM
Subject: Re: [asterisk-users] Dropped calls when all DAHDI l
You could use GROUP & GROUP_COUNT to track how many channels you are using
before you attempt to dial out and send back a Busy/Congestion/Whatever to
your endpoint when you are at your limit.
On Mon, Oct 8, 2018 at 4:33 PM Andrew Martin wrote:
> Hello,
>
> I am running Asterisk 11.17 with DAHDI
- Original Message -
> From: "John Novack SCII_U"
> To: "Asterisk Users Mailing List, Non-Commercial Discussion"
> , "Andrew Martin"
>
> Sent: Monday, October 8, 2018 4:29:41 PM
> Subject: Re: [asterisk-users] Dropped calls when all DAH
Have you given any thought to moving to at least a current supported version 13?
Asterisk 11 has been EOL for some time now
I doubt you will get a resolution to a version no longer supported.
Moving to the latest version 13 should be relatively quick and painless, and if
the issue persists you mi
Hello,
I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x
analog
POTS lines coming into my Asterisk server from the phone company. Internally, I
have about 180 SIP clients defined in sip.conf. What appears to be happening is
that if existing calls are consuming all 8 exter
Hi
When using an extension to my android gingerbread nexus one,
calls drop after a n minutes of call due as per the following
[Jun 21 09:34:37] == Begin MixMonitor Recording SIP/nexusone-0a39
[Jun 21 09:34:37] -- Executing [00310@default:4] Dial(" ...
[Jun 21 09:34:37] -- Cal
On 4/7/2010 2:45 AM, asterisk card support wrote:
> hi:
> how about the codecs?
>
>
> Best wishes!
> Asterisk Support group(sangoma, digium...), providing asterisk conf,
> pri, ss7, elastix, trixbox support.
> website:www.cnasterisk.com, www.voip88.com
>
>
>
I have the phones and asterisk limited t
>> On 3/31/2010 12:16 PM, Philipp von Klitzing wrote:
>> >
>> > Maybe rtptimeout in sip.conf is involved (and not behaving as expected)?
>> >
>> I was suspecting something with either rtptimeout or sip registration
>> timeout, but I'm not sure what.
Hi,
I have had similar issue. I have downgrade
isk-users@lists.digium.com
> Subject: Re: [asterisk-users] Dropped Calls
>
> On 3/31/2010 12:16 PM, Philipp von Klitzing wrote:
> >
> > Maybe rtptimeout in sip.conf is involved (and not behaving as expected)?
> >
> I was suspecting something with either rtptimeout or s
On 3/31/2010 12:16 PM, Philipp von Klitzing wrote:
>
> Maybe rtptimeout in sip.conf is involved (and not behaving as expected)?
>
I was suspecting something with either rtptimeout or sip registration
timeout, but I'm not sure what.
--
_
On 3/31/2010 12:06 PM, Danny Nicholas wrote:
> Just to get a 100% correct response to last question, are you using the flat
> CDR or mysql/some other DB?
>
All sip clients/peers are defined in sip.conf, dial-plan is entirely in
extensions.ael. We have one office that uses an Asterisk native
Hi!
> > I am just curious because I was having problems with dropped calls as
> > well
Maybe rtptimeout in sip.conf is involved (and not behaving as expected)?
> All extensions are hard-coded. We only have a handful of
> phones that don't change.
This last sentence is a wounderful example of a
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dropped Calls
On 3/31/2010 10:38 AM, Michael L. Young wrote:
>
> Is there a chance that you are using Realtime at all?
>
> I am just curious because I was having problems with dropped calls as wel
On 3/31/2010 10:38 AM, Michael L. Young wrote:
>
> Is there a chance that you are using Realtime at all?
>
> I am just curious because I was having problems with dropped calls as well
> and just discovered that it appears to be related to the database server.
> If for some reason on the database se
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of JR Richardson
> Sent: Tuesday, March 30, 2010 6:55 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Dropped
> I've written about this issue several times, but have not yet found any
> solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones
> are primarily Snom 300's but I also have a couple of headset phones
> connected to Grandstream HT286 SIP adapters. I have 8 offices, each has
> it'
On 3/30/2010 3:14 PM, Danny Nicholas wrote:
> A few thoughts;
> 1. I assume that the * servers aren't on dedicated networks; Do the dropped
> or one-way calls occur during high-traffic times or are they concurrent with
> large downloads? In my shop, we had to get a router that would prioritize
>
:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson
Sent: Tuesday, March 30, 2010 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped Calls
I've written about this issue several times, but have not yet found any
solution to
I've written about this issue several times, but have not yet found any
solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones
are primarily Snom 300's but I also have a couple of headset phones
connected to Grandstream HT286 SIP adapters. I have 8 offices, each has
it's own
Benny and Mark,
Thank you for your replies.
I tried adding t1min=500 to sip.conf per the suggestion below and since doing
that haven't been able to reproduce the issue.
If it comes back, I'll do the SIP debug per Mark's suggestion and post the
results here. (Mark, per your question the Auto At
Maybe you have a Codec issue?
On Tue, Oct 28, 2008 at 11:20 PM, Benny Amorsen
<[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> wrote:
> Lincoln King-Cliby <[EMAIL PROTECTED]> writes:
>
> > Periodically I'm seeing calls placed from the 7961s through anything
> > on the PBX that requires digit entry (the Au
Lincoln King-Cliby <[EMAIL PROTECTED]> writes:
> Periodically I'm seeing calls placed from the 7961s through anything
> on the PBX that requires digit entry (the Auto Attendant, Voicemail,
> etc.) 'randomly' drop; extension-to-extension calls
> extension-to-PSTN, and PSTN-to-extension calls never
Lincoln King-Cliby wrote:
> Hi All,
>
> I've looked through the archives and tried several variations in Google, and
> I haven't found anything on-point... So I'm hoping someone here may be able
> to help this relative Asterisk neophyte shed some light on an issue:
>
> I have a box running Aste
Hi All,
I've looked through the archives and tried several variations in Google, and I
haven't found anything on-point... So I'm hoping someone here may be able to
help this relative Asterisk neophyte shed some light on an issue:
I have a box running Asterisk 1.4.22 in our lab with several Cisc
On Jan 31, 2008 6:45 AM, mccoy silva <[EMAIL PROTECTED]> wrote:
> I have a very serious problem with calls between PAP2-NA and a TDM2400 (8
> FXO). Almost every call dropped after between 20 and 30 seconds with
> conversation.
> I disable the sound card, serial and other things on my server, but th
I have a very serious problem with calls between PAP2-NA and a TDM2400 (8
FXO). Almost every call dropped after between 20 and 30 seconds with
conversation.
I disable the sound card, serial and other things on my server, but the
problem still continues. I've changed the RPT Packet Size to .20 on PA
Jared Smith wrote:
> On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org
> wrote:
>
>> Randomly I have dropped calls during communication. No absolutetimeout or
>> other
>> calling limitation options.
>>
>> Any ideas on how to solve this problem?
>>
>
> The first place
On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org
wrote:
> Randomly I have dropped calls during communication. No absolutetimeout or
> other
> calling limitation options.
>
> Any ideas on how to solve this problem?
The first place I'd look would be the Asterisk CLI. Make su
Hi all,
I have a problem with some asterisk boxes.
I have a standard installation with 1.4.14 (I also test with 1.4.4) in core duo
Mac Mini on Debian Etch. I use SJphone softphone, Linksys SPA921 or Thomson 2030
for phones. All my phones are in a LAN with good status of 2ms max.
Randomly I have
h a Digium card is
a Dell 1750.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J. Oquendo
Sent: Thursday, March 15, 2007 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped calls in Asterisk - A general q
Hey all, I have a question for those administrating/building out
systems with over 30 users on them. How often do you experience
the dropped call phenomena. Would you care to share your
experiences including what versions of * you were using, what
kind of connectivity was present (T1, Fractional
I've been getting a number of dropped calls today with the following
visible in the logs:
Feb 5 12:06:29 DEBUG[8340] channel.c: Bridge stops because we're zombie
or need a soft hangup: c0=SIP/peachnet-213243-b7830fd8,
c1=SIP/2142-08f4b6a0, flags: No,No,No,Yes
Feb 5 12:06:29 DEBUG[8340] chan
On Fri, Jan 12, 2007 at 08:23:27PM -0600, Carlos Chavez wrote:
> I have an Asterisk server with 3 TDM400P cards. 9 FXO and 3 FXS ports.
> It also has 2 Astribank-8 units connected. The customer is having calls
> dropped at random intervals but several times a day. Could this be an issue
>
I have an Asterisk server with 3 TDM400P cards. 9 FXO and 3 FXS ports.
It also has 2 Astribank-8 units connected. The customer is having calls
dropped at random intervals but several times a day. Could this be an issue
with Interrupts with the 3 cards?
I am also having a problem se
>> Thanks,>
> Steve
>>>> >> *From:*
[EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]] *On Behalf Of> *Jonathan Barratt
> *Sent:* Wednesday, September 13, 2006 1:02 PM> *To:* Asterisk Users M
*From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]] *On Behalf Of> *Jonathan Barratt> *Sent:* Wednesday, September 13, 2006 1:02 PM> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Dropped Calls on TDM400p>>>> These are just P
EMAIL PROTECTED] *On Behalf Of
*Jonathan Barratt
*Sent:* Wednesday, September 13, 2006 1:02 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Dropped Calls on TDM400p
These are just PSTN calls, and I have set busydetect=no and
callprogress=
] On Behalf Of Jonathan Barratt
Sent: Wednesday, September 13,
2006 1:02 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped
Calls on TDM400p
These are just PSTN calls, and I have set busydetect=no and
callprogress=no in zapata.conf as per voip
These are just PSTN calls, and I have set busydetect=no and
callprogress=no in zapata.conf as per voip-info guidance, but problem
persists.
CPU load never breaks 20, so that doesn't seem to be the problem, but it's a 1.2Ghz Athlon with 768MB RAM.
Power supply to system is clean, there's no heavy
We are receiving a large amount of dropped calls on our asterisk system.
After debugging I find the following line at the same time the call is
dropped.
(DEBUG[8882] channel.c: Got a FRAME_CONTROL (5) frame on channel)
I was unable to find very much information on this message. Just a quick
back
Hi All... Well, I'm still experiencing LOTS of dropped calls since
installing the new (non pri) T1 here... I keep noticing a few things in the
logs when this happens, namely the "Wink/Flash" statements and the "Didn't
get a frame" messages...
Anyone got any ideas on if this is a telco issue, a wir
Chris Mason (Lists) wrote:
I have been experiencing dropped calls on my iax2 connections between
my Asterisk server and my ITSP providers, I use Teliax and Voxee but
it seems to happen on both so I don't think it is the provider. I
don't see any packet loss at the time so I don't think it is
I have been experiencing dropped calls on my iax2 connections between my
Asterisk server and my ITSP providers, I use Teliax and Voxee but it
seems to happen on both so I don't think it is the provider. I don't see
any packet loss at the time so I don't think it is poor Internet
connectivity, w
This post is exactly my problem:
http://lists.digium.com/pipermail/asterisk-users/2005-July/117988.html
Has anybody encountered this and been able to solve it and use g729
successfully? Are there other g729 implementations available as a codec
for asterisk?
Mojo
Mojo with Horan & Company, L
With verbose and debug both on 255, here's all I get at the CLI. The X
is during the call, at the instant the Zap leg seems to drop, almost
concurrently with the 'Hungup Zap/1-1'.
-- Executing Macro("SIP/112-a88a", "internaldialout|7476011") in new
stack
-- Executing ChanIsAvail("SIP/1
I don't know what to look for in my sip debug logs, can anybody suggest
what sorts of messages my phones might unexpectedly give asterisk
causing it to drop the zap leg?
Mojo with Horan & Company, LLC wrote:
Hello - I have 8 polycom 501s all setup great using ulaw. We have put
them through a
Is it at all possible asterisk is receiving a SIP message from the phone
causing it to drop the zap channel? I've got vad turned off in the
polycom configs. Guess I'll comb the sip debug logs.
I've got callprogress turned off. I'll try verbosity and debug levels
greater than 30 to see if an
Hello - I have 8 polycom 501s all setup great using ulaw. We have put
them through a pretty rigorous torture over the last 4 months, and
they've performed famously. No dropped calls ever.
We invested in some g729 licenses. changed my ipmid.cfg so that g729 is
priority 1 and ulaw is priority
I have an application where calls come into an *box from a DID
provider, and may be transferred to a meetme conference on another
*box (the call is released by the first *box after transfer).
These are ulaw IAX channel calls, and if the source is from a Verizon
or Nextel mobile phone to the
Hey,
I've done some searching for this and never really found a concrete
answer. Is there a specific reason or solution why just in the middle
of a call Asterisk will drop it and I'll get dial tone again? Anyways,
this is the output from /var/log/asterisk/full at the time of disconnection:
Ma
Hi,
I am having trouble with dropped calls in Asterisk. I've done a bunch
of searching but all I could find was setting busydetect and
callprogress to yes in zapata.conf to help combat the problem, but I did
this to no avail. The following is the output from
/var/log/asterisk/full at the time
I have an Asterisk system that seems to be randomly dropping calls. The
system is currently running on a Junghanns octo-bri card and one Digium
TDM400b card. The calls seem to be dropped during peak call hours and I
assume that it probably has something to do with congestion (this box
makes/receive
I am having lots of problems with dropped calls from my asterisk.
My Setup is:
Dell PowerEdge 1750 with 2.4Ghz Single Zeon
1GB Ram
2 x 36 GB SCSI Drives
The server is co-located
I connect to my providers via uLaw and IAX2.
My remote users are mainly are IAX2 softphone users.
I also have a DDI
Hiya Andres,
I stared to have exactly the same problem - very soon after enabling the
busydetect=yes in zapata.conf.
Used to work flawless with it set to 'no'. The only reason I turned it
on was I was trying to busy line detection and auto redials. Ive set it
back off at the mo - just to be sure
hi
Il mer, 2004-11-03 alle 21:09, Andres Maduro ha scritto:
> I am currently using busydetect=yes with busycount=5 and I have tweaked the
raise busycount to at least 6 or even 8.
In my experience 4 is an assurance to see dropped calls,
5 also, but less frequently, 6 never.
but I prefer 8, to be
Hi,
I have successfully configured and built Asterisk and now it is working fine
from the functionality point of view as sometimes we are getting dropped
calls.
The problem I am getting with POTS lines even if I receive/make a call from
a sip or analog phone is that the call may be dropped rando
do a 'sip debug' and make sure all looks good.
TL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bruce Komito
Sent: Monday, May 17, 2004 10:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dropped calls
I'm having a problem with o
I'm having a problem with outgoing dropped calls. They symptom is, when I
place a call from a sip extension to the outside, the call is connected
properly, but then abruptly disconnects anywhere from 10 to 60 seconds
later. This happens when the outgoing call is through a POTS line (TDM)
as well
> Ok, I'll watch for that as well since I upgraded my desk's Grandstream to
> 1.0.4.54 an hour ago (previously I had either 4.26 or 4.17).
It appears the hangup is triggered by a SIP ACK with CSeq set to 0.
Some Grandstream UAs happen to pick 0 as CSeq. chan_sip.c contains
if (!p->lastinvite &&
Hi!
> Only Grandstream phones appear to be affected. All phones affected
> have been behind a coned NAT, running firmware 1.0.4.39 with STUN
> enabled. The hangup only occurs in dialogs with CSeq set to '0'.
Ok, I'll watch for that as well since I upgraded my desk's Grandstream to
1.0.4.54 an h
ailto:[EMAIL PROTECTED] Behalf Of Philipp von
Klitzing
Sent: Thursday, April 15, 2004 11:22 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dropped calls
Hi!
> > I see this very same effect rather often in the following setup:
> >
> > SIP (GS101) --> *1 --> IAX2 --&g
On Thu, 2004-04-15 at 18:21, Philipp von Klitzing wrote:
> So to me it looks like IAX2 is involved as well, not just SIP.
Are you sure?
I did some analysis of my traffic. Here is what I found so far:
Only Grandstream phones appear to be affected. All phones affected have
been behind a coned NAT
Hi!
> > I see this very same effect rather often in the following setup:
> >
> > SIP (GS101) --> *1 --> IAX2 --> *2 --> MGCP (ip10)
> >
> > In fact I think I've seen it also with SIP instead of MGCP at the end.
> > The first client is behind NAT, by the way.
>
> That must be it. I have seen thi
> I see this very same effect rather often in the following setup:
>
> SIP (GS101) --> * --> IAX2 --> * --> MGCP (ip10)
>
> In fact I think I've seen it also with SIP instead of MGCP at the end.
> The first client is behind NAT, by the way.
That must be it. I have seen this happening with sip --
Hi!
> Lately, I have been experiencing unexpected hangups just when the a call
> has been established. This effects a small percentage of all calls
> coming from sip phone which are terminated on a zap pri channel.
I see this very same effect rather often in the following setup:
SIP (GS101) -->
Lately, I have been experiencing unexpected hangups just when the a call
has been established. This effects a small percentage of all calls
coming from sip phone which are terminated on a zap pri channel. I
turned on sip and pri debugging and it almost looks like the ACK message
coming back from th
Hate to reply to my own message again, but I just figured it out.
Nothing wrong with asterisk really, just a bad configuration. Somehow
the queue line in extensions conf got changed by someone to:
exten => 81003,3,Queue(receptionistq|tTH||10)
Thats where the 10 was coming from. :) Could this
I just updated to latest cvs and the problem remains. I did also notice
that when the call coming in on the queue is through a Zap line (from an
adtran 750 to an x100p) it logs the following just before the warnings
below:
pr 7 14:21:21 VERBOSE[60194841]: -- SIP/hrutter-432b answered Zap/
We're having a strange problem with our receptionist. She runs an xpro
softphone and we're using a queue to handle incoming calls. It seems
nearly all of the calls that come in through the queue get dropped. At
first we thought it might have been human error (clicking the wrong
button in xpr
Hello,
We have an * installation that is causing us fits.
The problems we are seeing:
1) In the middle of a call the call gets dumped and the caller hears a
dial tone.
2) While talking on a call the caller hears nothing for 5 to 10 seconds.
The person on the other end of the call hears e
> > > worth a try.
> > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED]
> > > > [mailto:[EMAIL PROTECTED] On Behalf Of
> > > > Paulo Loureiro
> > > > Sent: Friday, March 05, 2004 10:26 AM
> &g
t; -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of
> > > Paulo Loureiro
> > > Sent: Friday, March 05, 2004 10:26 AM
> > > To: [EMAIL PROTECTED]
> > > Subject: [Asterisk-Users] dropped calls
&g
l Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Paulo Loureiro
> > Sent: Friday, March 05, 2004 10:26 AM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] dropped calls
> >
> > Hello list,
> >
&
L PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Paulo Loureiro
> Sent: Friday, March 05, 2004 10:26 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] dropped calls
>
> Hello list,
>
> I'm getting droped calls on an asterisk installation. When on GS phone
>
Hello list,
I'm getting droped calls on an asterisk installation. When on GS phone
dials another one, the call is dropped after some (usually random) time
but most of the tome within 3 to 20 seconds.
I think the cause is stated on the logs, see bellow, and is related with
the message "Didn't get
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