.
In:
ITSP Profile A
DigitMap: (xx.|*xx.|#xx.)
In: PHONE1 Port
DigitMap: (xx.|*xx.|#xx.)
When I call Line1 on OBi202 it rings but I can not make a call out
neither local no outbound.
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Thelma
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is working OK.
It seems to me Audiocodes MP-112 is trimming anything that is longer than
3-digits.
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Thelma
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Check out the new Asterisk community forum
as set to 3 in:
Configuration->VoIP->GW and IP to IP->DTMF and Supplementary ->DTMF & Dialing
I don't know when did they change it to 3 as my MP-114 came with default
32-digits
Thelma
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sterisk server connected/registered over IAX and that error:
"...exited non-zero on..."
eg.
-- SIP/54-0006 is ringing
== Spawn extension (extensions, 4, 3) exited non-zero on
'IAX2/home_server-424'
I'm not the only one with this problem, this guy has the same problem as me:
n 'IAX2/home_server-424'
Does it have something to do with "transfer=no" in iax.conf?
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Thelma
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Does asterisk listen on port 4569 by default?
I'm running version Asterisk 11.25.1 and have a problem registering
Zoiper (IAX) to Asterisk.
I'm getting an error:
Registration refused
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Thelma
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I'm getting:
netstat -a |grep 4569
udp0 0 0.0.0.0:45690.0.0.0:*
Should I be getting localhost IP?
Thelma
On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> Does asterisk listen on port 4569 by default?
>
> I'm running version Asterisk 11.25.1 and have
k
== Using SIP RTP CoS mark 5
-- Called SIP/54
-- SIP/54-0289 is ringing
== Spawn extension (extensions, 4, 3) exited non-zero on
'IAX2/home_server-6364'
-- Hungup 'IAX2/home_server-6364'
Extension 4 is not BUSY (just nobody pickup the call) so why isn't call going
to "Voicemail&
transferred?
Even when I put:
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
exten => 4,n(line2),Dial(${FD_L2},20,trw)
exten => 4,n(line2),Voicemail(4)
The call (line2) would dial "FD_L2" but would not jump to next line
"Voicemail"
--
Thelm
;SIP/54,20,rw")
in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/54
-- SIP/54-0307 is ringing
== Spawn extension (extensions, 4, 3) exited non-zero on
'IAX2/home_server-435'
-- Hungup 'IAX2/home_server-435'
So FD_L1 (exten: 4) is ringing for 25sec.; nobody pickup the p
No, I don't think it is IP table issue, I've not upgraded dd-wrt for a
while and it was zoiper was working OK with my previous version of asterisk.
After upgrade to 11.25.1 it stop working.
I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
443 open.
Thelma
On 06/05/2017 07
Doesn't matter how much I increase the verbose output
asterisk -vvr
asterisk will not even print a single line.
How to find out if my firewall has this port open?
https://www.grc.com
is reporting that my port is 4569 is in Stealth mode (so it is closed) :-/
Thelma
On 06/05/2017 02:19 PM
ength 40
14:20:45.357331 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, length 65
14:20:45.376559 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, length 53
14:20:45.376630 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, length 12
^C
12 packets captured
12 packets received by filter
0 packets
yes it does.
netstat -nap | grep 4569
udp0 0 0.0.0.0:45690.0.0.0:*
17375/asterisk
Thelma
On 06/05/2017 03:10 PM, Helvio Junior wrote:
> Use the command bellow to check if is Asterisk opening the port.
>
> netstat -nap | grep 4569
n sip.conf only:
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
Is perhaps the name effected by the special character "-" (dash) that is
why it only matches "pstn" and take the first one it found. Will it
make a difference if I rename the port to pstn_ in
o asterisk reported as such.
And I think asterisk suppose to lookup this label in sip.conf to the
registered entry but instead selected pstn-9998 entry; I don't know why.
If the call came IN on pstn-
and sip.conf has two entries:
[pstn-]
[pstn-9998]
Why it can not distinguish between the two of t
Thelma
On 02/15/2018 07:16 PM, the...@sys-concept.com wrote:
>
> On 02/15/2018 04:49 PM, Joshua Colp wrote:
>> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>>
>>
>>
>>>
>>> Thanks again for the hint.
>>> Her
this one)
disallow=all
allow=gsm
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=1
qualify=yes
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Thelma
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Chec
69 ; just for audiocodes error complain
> host=dynamic
> insecure=port,invite
> canreinvite=no ; (dtmf not working correctly without this one)
> disallow=all
> allow=gsm
> allow=ulaw
> allow=alaw
> nat=no
> context=incoming
> callg
d I think asterisk suppose to lookup this label in sip.conf to the
>> registered entry but instead selected pstn-9998 entry; I don't know why.
>>
>> If the call came IN on pstn-
>> and sip.conf has two entries:
>> [pstn-]
>> [pstn-9998]
>>
>> Why i
hts up but
asterisk is showing that the call is coming on "pstn-9998"
-- Executing . Answer("SIP/pstn-9998
Asterisk should be showing "pstn-" (not pstn-9998)
Where is this label coming from?
--
Thelma
--
___
54 incoming
No No
Caller display ID from PSTN on FXO ports are working OK.
The [pstn-] is channel: 4
The [pstn-9998] is channel: 3
If the call on Audocode is lighting UP "channel:3" the sip.conf should
associate that call with [pstn-] (and no
NAT)
Sending to 10.10.0.8:5060 (no NAT)
Using INVITE request as basis request - 7668022781522018162620@10.10.0.8
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio descriptio
I'm planning to upgrade my asterisk-11.25 to ver. 13
or should I go to 11 to 16
Is there any official documentation how to upgrade, what to watch for
during upgrade?
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On 12/06/2020 01:44 PM, Jöran Vinzens wrote:
> Hi,
>
> I did a talk on Astricon 2019 on this topic. Unfortunately there are no
> videos of that year but you can find my slides here covering some pitfalls.
> https://www.slideshare.net/JranVinzens/asterisk-11to16-what-could-go-wrong
>
> Good luck
Sound reasonable. I know it take time to debug the dial-plan after upgrade.
Can I use sipp, from command line to call my local asterisk specific
extension and to observe in another terminal via "asterisk -vvr"
what it is doing?
On 12/07/2020 11:50 AM, Eric Wieling wrote:
> Read UPGRADE.TXT
On 12/07/2020 05:06 AM, Jöran Vinzens wrote:
> Hi,
>
> I guess describing how SIPp works here on a mailliste might be too much.
> But if you do not want to prove your setup automatically, you do not need
> to know SIPp.
>
> But there was a talk in 2014 Astricon giving an overview about SIP
On 12/23/2020 09:54 AM, Doug Lytle wrote:
> Review your features.conf file in /etc/asterisk
>
> Doug
I found id. Thanks.
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I just upgraded to asterisk-13 (from 11) and I get some errors:
1.)
Unknown DYNAMIC_FEATURES item 'automon' on channel SIP
Unknown DYNAMIC_FEATURES item 'automon' on channel IAX2/voip
Does anybody know how to get rid of them?
--
In: /var/lib/asterisk
-rw-r--r-- 1 asterisk asterisk 12288 Dec 23 10:52 astdb.sqlite3
ast_db_put: Couldn't execute statement: SQL logic error
ast_db_put: Couldn't execute statement: attempt to write a readonly database
db_execute_sql: Error executing SQL (COMMIT): database is locked
The
In astersik-11 MWI light was cleared as soon as I checked the message.
In asterink-13 it takes about 20min to set the light ON and the light
takes over an hour to clear.
What had changed?
In sip.cong
[400]
...
mailbox=400
voicemail.conf
[default]
400 => ,user, email
--
On asterisk-11 MWI was working correctly, phone message light was
blinking (standard phone).
With asterisk-13, this feature is not working.
Who to trouble shoot?
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I just tired/upgraded to asterisk-16.13.0 and the problem still persist.
Looking at some post on digium.support web-page, they don't have a clear
solution or know what causing it.
On 12/24/2020 05:03 AM, Julian Beach wrote:
> Hello Thelma,
>
> Thursday, December 24, 2020, 9:26:53 AM, t
I'm disappointed with Sangoma!
I have one of those Digium S101i (iaxy) adapters that is still working (in
production), doesn't need any drivers.
After, I purchased Sangoma USBfxo (U100) adapter, never had a chance to use it
(still brand new in a box) that require some extra driver to run it,
On 1/4/21 10:09 AM, Doug Lytle wrote:
app.c:280 ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is
not available.
>
> Macros are no longer built by default in Asterisk 16. This was documented in
> the UPGRADE.txt file
>
> app_macro:
> - The app_macro module is now
On 1/4/21 9:04 AM, Joshua C. Colp wrote:
> On Mon, Jan 4, 2021 at 11:57 AM wrote:
>
>> Did execution of macro changed in Astersik-16.15 ?
>>
>> When I try to dial an extension that call macro I get an error:
>>
>> app.c:280 ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is
>> not
Did execution of macro changed in Astersik-16.15 ?
When I try to dial an extension that call macro I get an error:
app.c:280 ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not
available.
Dial(SIP/718xx@pstn-5665,20,m(default)M(atb))
--
Thelma
On 1/4/21 12:01 PM, Doug Lytle wrote:
Gosub(check-number-forwarding,Dial(SIP/718xxpstn-5665,20,U(atb-sub))
>
Is ARG1 = atb-sub ?
>
> No.
>
> My complete line
>
> exten => _45XX,1,Set(_ARG1=${EXTEN}
> same => n,Gosub(check-number-forwarding,s,1(${ARG1}))
>
> So, if
On 1/4/21 10:44 AM, Doug Lytle wrote:
>
How do you enable the phone speaker on the Gosub?
>
I had:
Dial(SIP/718x@pstn-5665,20,m(default)M(atb))
>
> You can provide variables to your gosub routine, for an example
>
> Gosub(check-number-forwarding,s,1(${ARG1}))
>
> Doug
I'm using asterisk-16.30.1
When I try to call another asterisk server over IAX I get a busy signal,
chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
-- IAX2/192.168.143.1:4569-656 is circuit-busy
Asterisk-16.16 is working normally, no congestion error.
--
Thelma
On 1/2/24 15:13, aster...@phreaknet.org wrote:
On 1/2/2024 4:03 PM, the...@sys-concept.com wrote:
I'm using asterisk-16.30.1
When I try to call another asterisk server over IAX I get a busy signal,
chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
--
On 1/3/24 04:53, Henning Follmann wrote:
On Jan 2, 2024, at 23:17, the...@sys-concept.com wrote:
On 1/2/24 15:13, aster...@phreaknet.org wrote:
On 1/2/2024 4:03 PM, the...@sys-concept.com wrote:
I'm using asterisk-16.30.1
When I try to call another asterisk server over IAX I get a busy
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