Displaying 20 results from an estimated 1000 matches similar to: "SIP Calls on Asterisk fails after 25000 calls"
2009 Apr 26
1
Error, Clue to what?
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer
'3516533812' is now UNREACHABLE! Last qualify: 86
[Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke:
Peer '3516533812' is now Reachable. (98ms / 2000ms)
[Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 -
2010 Jan 29
1
Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
Hi,
I have a tdm22b (2 fxs / 2 fxo)
When Asterisk is just started, outbound calls routing to fxo port, do not working with error:
Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
Inbound calls to fxo port work fine.
After first inbound call, the outbound calls starts working.
CentOS 5.4
asterisk 1.6.0.21-1
dahdi 2.2.1.-1
Can anybody help me to identify what is the
2006 Mar 20
4
simple perl-agi - where's the error?
Hello!
I'm trying to setup a perl-deadagi, but my perl skills lack. can
someone tell me why the following code doesn't work:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$dialstring = $AGI->get_variable("DIALSTRING");
$res = $AGI->exec("DIAL $dialstring");
the asterisk output says:
AGI Rx << GET VARIABLE DIALSTRING
AGI Tx >> 200
2007 Dec 06
1
Dial() Macro option error in 1.4.15
After updating to 1.4.15, I have the following issue:
When I try to use the "M" macro option in the Dial() option, I get the
following in the console:
-- Executing Dial("Zap/1-1", "Zap/g2/w5051234|60|M(set-userfield^local)KT")
-- Called g2/w5051234
-- Zap/3-1 answered Zap/1-1
[Dec 6 12:10:58] ERROR[19496]: app_dial.c:1541 dial_exec_full: Unable to
start
2005 May 18
2
Call forwarding...
Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based -
2015 Jan 19
1
tg3 network link unstable
Full updated EL6: Does someone have the same behaviour, unstable eth interface (tg3/no dhcp)?
Jan 19 18:01:46 ane kernel: tg3 0000:04:00.0: eth0: Link is up at 1000 Mbps, full duplex
Jan 19 18:01:46 ane kernel: tg3 0000:04:00.0: eth0: Flow control is on for TX and on for RX
Jan 19 18:01:46 ane kernel: tg3 0000:04:00.0: eth0: EEE is enabled
Jan 19 18:01:46 ane NetworkManager[1735]: <info>
2006 Dec 21
2
asterisk crashed
our * crashed twice in a month with segmentation fault &
a core dump. here's the stack trace:
#0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6
#1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6
#2 0xb7e17090 in strdup () from /lib/tls/libc.so.6
#3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879
#4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at
2013 Jun 10
3
no silk translation ?
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:
[Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c
[Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032
dial_exec_full: Had to drop call because I couldn't make
2013 Feb 04
0
Calculating Weights for Variable Groups
Hello and Thanks in advance for your suggestions.
As a new member, I do not know exactly if such problem has ever been discussed in this forum.
I need a small help using FactoMineR
and RcmdrPlugin.FactoMineR package to calculate weights for the
individual observational units. My data looks like: You can treat the data spaced instead of tabbed.
2007 Oct 09
1
Error: 603 declined
I have Asterisk 1.2.13 installed as a Debian package and I edit only
sip.conf and extensions.conf in this way:
sip.conf:
[general]
realm=work.com.ar ; Realm for digest
authentication
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes
2009 Jun 02
2
error with dial timeout
Hello,
I am trying to do :
Exten =>_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:10000))
But it return that error:
[Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid
timeout specified: 'L(10208400:61000:10000)'
Why?
I forgot something ?
Thank you
Cordialement,
BERGANZ Fran?ois
P Pensez ? l'Environnement, n'imprimez ce mail que
2015 Nov 24
2
subscriber state before dial
Hi All
After a Dial() I get:
WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create
channel of type 'SIP' (cause 20 - Subscriber absent)
if the subscriber is not registered.
Is there a way from dialplan to know, *before* Dial(), if a destination
Subscriber is
a) not registered or
b) busy ?
I need to redirect a call to some other Subscriber if (s)he is not there
2006 May 01
6
Problems with zaptel and TE210P
Hello,
I'm just starting out with asterisk and I'm playing around with the
system. Currently I have a Digium TE210P connected to a PRI on the
Asterisk server. I have a SIP soft phone on my laptop for testing that
is working fine. When I try to place a call from my soft phone I get
this from Asterisk:
May 1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to
create
2016 Mar 15
2
Fwd: Unable to place outbound calls
Hi I need help
This is the error:
Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.'
Method: SUBSCRIBE
-- Executing [00919885497796 at internal:1] Set("SIP/1001-0000000b",
"CALLERID(num)=8790771141") in new stack
-- Executing [00919885497796 at internal:2] Dial("SIP/1001-0000000b",
"SIP/00919885497796 at sonetel")
2019 Feb 15
2
Set qualify = yes on trunk can't do outgoing call
Hello when I set qualify = yes on trunk I can't do outgoing call.
Incoming is always working.
[Feb 15 23:01:41] WARNING[12909][C-00000012]: app_dial.c:2525
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
but my linphone is registered all the time.
when set qualify = no outgoing call is working
(but i have problems when WAN IP is changed after
2008 Jan 28
2
Dial agent channel - busy
Hi,
when I'm trying to call the following extension
exten => 6002,1,Verbose(1|Extension 6002)
exten => 6002,n,Dial(Agent/6002)
exten => 6002,n,Hangup()
the call is terminated and I get the following warning from asterisk:
app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent'
(cause 17 - User busy)
When calling the agent with Dial(SIP/6002) no problem
2005 Oct 13
2
Enum parse errors
I'm running into errors when using Enum lately. I can't figure out what
the problem might be as I've had Enum up and running in the past. I'm
running the latest CVS-Head compiled version. I've also tried using the
new Enum function with the same results. When doing a lookup on a number
that exists in the enum server I get the following results:
-- Executing
2006 Apr 04
1
Too many open files
Dear all,
we have encounter problem when starting asterisk in the foreground,
"asterisk -vvvvgc" with more 100 SIP calls concurrently. we have set
ulimit to the highest value. still has this problem. Is this the
problem keeping asterisk in the foreground or this is a bug in SVN 1.2
16771?
Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel
allocation
2010 Aug 17
1
dial_exec_full problems with TDM400
Hi,
I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support)
at the same
time as moving from Ubuntu hardy to
I have a single TDM400P rev I with two fxo and two fxs modules, these were
working perfectly for years
on Asterisk 1.4 using Zaptel drivers with Oslec.
Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu
package.
After several hours (perhaps 24
2004 Jul 27
2
Enum
You can play also with www.enum2go.com <http://www.enum2go.com/> or
wap.enum2go.com
Regards
Alex
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