Displaying 20 results from an estimated 212 matches for "dial_exec_full".
2009 Apr 26
1
Error, Clue to what?
...OTICE[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 - Unknown)
[Apr 26 12:51:20] WARNING[32281]: app_dial.c:1242 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
[Apr 26 12:52:56] WARNING[32284]: app_dial.c:1242 dial_exec_full: Unable to
create channel of ty...
2005 Mar 02
1
Dial application invoked again and again
...92.168.0.203>
Content-Length: 0
to 192.168.0.117:5060
Mar 3 10:44:01 WARNING[6311]: app_dial.c:618
dial_exec_full: hello i am from app_dial
We're at 192.168.0.203 port 15344
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:2000@192.168.0.117 SIP/2.0
Via: SIP/2.0/UDP
192.168.0....
2010 Aug 17
1
dial_exec_full problems with TDM400
...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 or so, not nailed it down precisely)
incoming
calls are not answered and outgoing calls get dial_exec_full.
Incoming calls are reported to either A:just ring and ring, or B:get an
engaged tone.
Strangely when this happens asterisk DOES see the incoming call in situation
A, but fails
to answer.
What tests can I do to resolve this as it is very inconvenient as we are
missing a lot of calls?
At the mom...
2006 Mar 21
0
Queue and busy/congested ZAP channels
.../1000
-- Executing Macro("Local/1003@public-phones-548f,2", "call-user|
1003") in new stack
-- Executing Dial("Local/1003@public-phones-548f,2",
"Zap/g4/1003&Sip/S1003&Sip/1003|45|tr") in new stack
Mar 21 09:57:38 NOTICE[4354]: app_dial.c:1030 dial_exec_full: Unable to
create channel of type 'Zap' (cause 34 - Circu
it/channel congestion)
-- Executing Macro("Local/1000@public-phones-c4bb,2", "call-user|
1000") in new stack
-- Executing Dial("Local/1000@public-phones-c4bb,2",
"Zap/g4/1000&Sip/S1000&a...
2007 Dec 06
1
Dial() Macro option error in 1.4.15
...n 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 autoservice on calling channel
[Dec 6 12:10:58] ERROR[19496]: app_dial.c:1553 dial_exec_full: Could not find
application Macro
-- Hungup 'Zap/3-1'
The requested Macro does exist in extensions.conf as:
[macro-set-userfield]
; Set CDR userfield to value defined in Dia...
2009 Nov 19
1
SIP Calls on Asterisk fails after 25000 calls
...asterisk-1.6.0.5) with
MySQL (using res_odbc)support for extensions and users list.
The call rate is 7 calls / second and each call stays for 120 seconds.
after making 25000 successful calls , calls started
failing with following message on CLI.
[Nov 11 08:50:04] WARNING[2258]: app_dial.c:1502 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
[Nov 11 08:50:04] WARNING[2259]: app_dial.c:1502 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
Is there any configuration parameters I missed out ? please provide
your valuable suggestion...
2005 May 18
2
Call forwarding...
...c/ in the TRUNK line I get...
-- Executing Dial("IAX2/08700688nnn@217.14.132.nnn:4569-1",
"c/07961106nnn|20|r") in new stack
May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for 'c'
May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type 'c' (cause 66)
Asterisk shows this from the moment the sip channel is considered not to
have answered (1 sec)...
-- Nobody picked up in 1000 ms
-- Executing Playback("IAX2/08700688nnn@217.14.132.nnn:4569-1",
"pls-wait-connect-call&...
2006 Dec 21
2
asterisk crashed
..., tv_usec = 0}) at channel.c:3260
#9 0x080655fd in ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0, config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c)
at channel.c:3524
#10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0, config=0xb6c4feb0) at res_features.c:1319
#11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0, peerflags=0xb6c50568) at app_dial.c:1577
#12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at app_dial.c:1619
#13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900, context=0xb659fe20 "op05_x", exten=0xb659ff14 "00116&quo...
2006 Mar 20
4
simple perl-agi - where's the error?
...>exec("DIAL $dialstring");
the asterisk output says:
AGI Rx << GET VARIABLE DIALSTRING
AGI Tx >> 200 result=1 (089324154332)
AGI Rx << EXEC DIAL ""
-- AGI Script Executing Application: (DIAL) Options: ()
Mar 20 11:46:02 WARNING[21970]: app_dial.c:773 dial_exec_full: Dial
requires an argument (technology/number) AGI Tx >> 200 result=-1
-- AGI Script agirouter/dialscript.pl completed, returning 0
so the get_variable-command seems to work, also the exec(with
"$dialstring = 089324154332" the call goes out), but not setting the
variable. shou...
2007 Oct 09
1
Error: 603 declined
......so I can make a call.
In Asterisk I debug the channel and I get this log:
voip*CLI> debug channel 1
No such channel 1
Debugging on new channels is enabled
-- Executing Dial("SIP/user1-08148450", "SIP|user2") in new stack
Oct 9 12:52:41 WARNING[3478]: app_dial.c:1024 dial_exec_full: Dial
argument takes format (technology/[device:]number1)
== Spawn extension (sintys, 1112, 1) exited non-zero on
'SIP/user1-08148450'
Oct 9 12:52:41 WARNING[3453]: channel.c:787 channel_find_locked:
Avoided initial deadlock for '0x81508e8', 10 retries!
What is the problem ??? A...
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 si n?cessaire.
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2013 Jun 10
3
no silk translation ?
...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 SIP/ng-00000000
compatible with Motif/+12025551212 at voice.google.com-da3c
== Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-00000000'
core show translations doesn't include any SILK.
SILK is installed:
core show codec 1...
2006 Apr 04
1
Too many open files
...is a bug in SVN 1.2
16771?
Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel
allocation failed: Can't create alert pipe!
Apr 5 00:48:36 WARNING[14887]: chan_local.c:523 local_new: Unable to
allocate channel structure(s)
Apr 5 08:48:36 NOTICE[14887]: app_dial.c:1042 dial_exec_full: Unable to
create channel of type 'LOCAL' (cause 0 - Unknown)
Apr 5 08:48:36 WARNING[14893]: res_agi.c:246 launch_script: unable to
create fromast pipe: Too many open files
Apr 5 08:48:37 WARNING[14894]: res_agi.c:246 launch_script: unable to
create fromast pipe: Too many open files
Ap...
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 i...
2006 May 03
1
my asterisk crashed
...meone help out?
#0 ast_var_name (var=0x1) at chanvars.c:71
71 if (var->name[0] == '_') {
(gdb) bt
#0 ast_var_name (var=0x1) at chanvars.c:71
#1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46
"OUTBOUND_GROUP") at pbx.c:5904
#2 0xf5bbe1e4 in dial_exec_full (chan=0xa281820, data=0x0,
peerflags=0xf469fee8) at app_dial.c:964
#3 0xf5bc23ed in dial_exec (chan=0x0, data=0x1) at app_dial.c:1601
#4 0x08090aed in pbx_extension_helper (c=0xa281820, con=0x0,
context=0xa281970 "default", exten=0xa281a64 "2348053004990",
priority=2, labe...
2006 May 24
1
DUNDi in 1.2.7.1
...0", "dundi-priv|201") in new stack
-- Executing Goto("SIP/214-9fe0", "201|1") in new stack
-- Goto (macro-dundi-priv,201,1)
-- Executing Dial("SIP/214-9fe0", "SIP/201|10|tr") in new stack
May 24 17:23:47 NOTICE[561]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Dial("SIP/214-9fe0", "SIP/201|10|tr") in new stack
May 24 17:23:47 NOTICE[561]: app_dial.c:1029 dial_exec_full: Unable to c...
2016 Mar 15
2
Fwd: Unable to place outbound calls
...xecuting [00919885497796 at internal:2] Dial("SIP/1001-0000000b",
"SIP/00919885497796 at sonetel") in new stack
Really destroying SIP dialog '
018143ec6cda0ebf5667665837f49196 at 127.0.0.1:5060' Method: INVITE
[Mar 14 19:55:15] WARNING[20595][C-0000000b]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/1001-0000000b' status is 'CHANUNAVAIL'
------------------------------------------------------------------------...
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 reconnect internet
connection)
how can i solve this?
best regards
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 occurs.
What could be wrong?
Some additional information about the configuration:
The asterisk version is 1.4.10
---------------------------------------------------...
2006 Apr 19
1
Codec problem from SIP to H323
...ne's
default codec I get:
-- Executing Dial("SIP/amejia-8be1", "H323/######@H323gw") in new stack
Apr 19 15:02:14 WARNING[68595]: channel.c:2504 ast_request: No translator
path exists for channel type H323 (native 4) to 256
Apr 19 15:02:14 NOTICE[68595]: app_dial.c:1010 dial_exec_full: Unable to
create channel of type 'H323' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
I don't get it why is it trying to "translate" anything. There's nothing to
translate, cause I'm using g729 in both ends.
Well, to make it more interes...