hi,
in my small setup (just for home usage) i have 5 phones configured. but
only 2 of them are permanent connected to asterisk.
nevertheless i want to address beside those two phones other peers if
available. nowadays i address them always, resulting in error messages:
Unable to create channel of
=ISO-8859-1; format=flowed Andre Gronwald wrote:
hi,
Hola,
in my small setup (just for home usage) i have 5 phones configured. but
only 2 of them are permanent connected to asterisk.
nevertheless i want to address beside those two phones other peers if
available. nowadays i address them
))
exten = 066104,n,http://192.168.5.109/interface2/interface2.php ( here
i want to launch this url in my pc )
exten = 066104,n,Hangup()
thanks and regards
--
Andre Gronwald
andregronwal...@gmail.com
andre.gronw...@gmx.de PGP-0x9CDEE439
I do it via a group count:
main call handling:
exten => sub123,n,Set(GROUP()=11122345)
...
the main routine calls subroutine:
exten => general,1,GotoIf($["${busyonbusy}"="YES"]?100:200)
exten => general,100,GotoIf($[ ${GROUP_COUNT()} > 1 ]?110:200)
exten => general,110,Hangup(17) ; fehlercode
Some further information:
asterisk version: 13.13.1, pjsip (pjproject) 2.5.5
regards,
andre
Am 13.02.2017 um 17:32 schrieb Andre Gronwald:
Hi all,
I have a strange issue, with a some kind complicate architecture...
A router of our internet provider is in front of another bintec rs353j
router
Hi all,
I have a strange issue, with a some kind complicate architecture...
A router of our internet provider is in front of another bintec rs353j
router, at which my freepbx installation is located. However, NAT etc.
seems to work fine.
BUT: Something is not working...:
When registering my
Hi all,
I have an issue with asterisk 13 and pjsip. I guess it is somehow
Firewall related, but I'm unsure.
A registration to Sipgate is established successfully:
==
is always reachable!!!
Is there any explanation for this? I just want to understand... ;-) ...
and solve it.
regards,
andre
Am 15.10.2016 um 10:11 schrieb Andre Gronwald:
[2016-10-15 10:03:22] WARNING[10162]:
res_pjsip_outbound_registration.c:761 schedule_retry: No response
rec
ping times are fine as well:
[root@freepbx asterisk]# ping sipgate.de
PING sipgate.de (217.10.79.9) 56(84) bytes of data.
64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
64 bytes from sipgate.de
Thanks Jonathan for your support.
I would like to avoid TLS at the moment (in general I am a fan of
secured communication!) because the other provider is not supporting
TLS. And sipgate is just used for testing.
However I can see the following which is quite interesting:
[2016-10-15
hi,
let me explain in detail, what i have configured and what is happening now:
1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to
device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
- configured NAT,
│
│ +59.790625 │ │ │
BYE│
│ 18:28:10.498322 │ │ │
> │
Am 15.10.2016 um 15:39 schrieb Andre Gronwald:
ok, now it is getting weird...
actually i don't see any firewall issues, but i am not able to get a
ok, solved the firewall issue.
A first test call worked fine. Another one now still gets disconnected
after 32s.
But in FW there are no blocked packets anymore?!
And I don't understand why the registration to the same IP and same Port
is working, but not later transmission of further SIP
ok, now it is getting weird...
actually i don't see any firewall issues, but i am not able to get a
call from outside to my sipgate account. in asterisk nothing is visible,
core set verbose is activated.
sngrep (on my asterisk server) shows me indeed the invite from sipgate!?
What I see via
> Can you get your own modem? (double) NAT is ugly hack.
Unfortunately not. The provider is only supporting this hardware.
> Not sure what is VoIP in the router here, but looks like some sort of SIP
ALG
> or VoIP passthrough - disable it! It rewrites ip addresses inside of the
> packets ang it
ISP won't change, but will check.
in the hidden menus it isn't changeable either.
However, it is working after i deactivated VoIP in the router. And even
after reenabling VoIP it is still working. I don't understand why...
However, it works. :-D
thanks a lot.
regards,
andre
--
Andre Gronwald
Hi all,
I have a setup which is not working right now:
Provider - DSL-Router (192.168.2.1) - Bintec-Router (10.17.46.66) -
Asterisk (10.17.46.99)
My issue: Everything works, but RTP is only going from my Asterisk
towards the provider. Asterisk is configured to use SIP-ports 55060 and
ion is currently in
development...
regards,
andre
--
Andre Gronwald
andregronwal...@gmail.com <andregronwal...@gmail.com
<andregronwal...@googlemail.com>>
--
_
-- Bandwidth and Colocation Provided by http://www.
Hi,
I am trying to add a custom header to my calls to map several call-legs
into a global call for viewing.
For this to work I read the call-id from pjsip-channel and write it into
X-CID:
##
-- Executing [s@macro-dialout-trunk-predial-hook:4]
Set("PJSIP/10-0006",
Thanks all for the help, I got a step ahead. But in this scenario I am
not able to deliver call-id of call-leg a to call-leg b.
Extension A is going to make an outbound trunk call:
1. extension calls asterisk (call leg a, call-id 1234567890)
2. asterisk makes outbound trunk call (call leg b,
the issue is quiet sure codec based:
[Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping
incompatible voice frame on SIP/messagenet-028e of format gsm since our
native format has changed to 0x8 (alaw)
shorter:
Dropping incompatible voice frame on SIP/messagenet-028e of
hi,
i am using a script to initiate calls to some sip stations simply for
notifying some people.
that is working fine and people like this simple way of getting an
information (just by being pinged this way).
my problem is, that in this case the calling number is always
"asterisk@".
i
after completion you find ${FAXMODE} filled with audio or T38, depending on
what has been used.
hope that is what you are looking for.
regards,
andre
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Look at Homer 7, which is using time series databases and you can do a lot
more than sip. But it is not an out of the box solution.
And it is not real time, but you can minimize intervals to some seconds.
Look at the several docker containers:
https://github.com/sipcapture/homer7-docker
Regards
You might have a look into Homer . It is really great, the community is
great, but it won't give you all the metrics you want. But it might be a
good start.
http://sipcapture.org
Regards,
Andre
Am Sa., 15. Dez. 2018, 19:01 hat
geschrieben:
> Send asterisk-users mailing list submissions to
>
Hi,
I am using these variables in my callfiles:
CallerID: "My Fax-ID" <+1234567890123>
setvar:FAXOPT(headerinfo)=My Fax-ID
setvar:FAXOPT(localstationid)=001234567890123
regards,
andre
Am 03.08.19 um 19:00 schrieb asterisk-users-requ...@lists.digium.com:
Date: Fri, 2 Aug 2019 22:22:24 +
hi,
we want to extract the information when the most callers are entering
our phone system based on an interval of 15 minutes. this is quite
simple (although not perfect) with
select calldate, count(*) as anzahl from cdr where calldate >
'2019-10-12' group by unix_timestamp(calldate) DIV 900
thanks john,
that is a good idea and really easy.
I selected both values to have a good comparison:
select calldate, count(distinct(clid)), count(clid) from cdr where
calldate > '2019-10-12' group by unix_timestamp(calldate) DIV 900 ;
now it would be nice to have intervals starting always
i've got it:
select from_unixtime(round((ceiling(unix_timestamp(calldate)/ 900)
*900))) as intervall, count(distinct(clid)), count(clid) from cdr where
calldate > '2019-09-01' group by intervall;
Am 12.11.19 um 15:16 schrieb Andre Gronwald:
would be better to have dates starting with &q
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