Re: [asterisk-users] CDR extract call numbers on interval on unique callers

2019-11-12 Thread Andre Gronwald 

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 "2019-10-14 08:00:00", 
"2019-10-14 08:15:00" etc...

any quick idea? i will search for that anyway.

regards,
andre



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Re: [asterisk-users] CDR extract call numbers on interval on unique callers

2019-11-12 Thread Andre Gronwald 


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 in the same 
manner. currently the output is like:


MariaDB [asteriskcdrdb]> select calldate, count(distinct(clid)), 
count(clid) from cdr where calldate > '2019-10-12' group by 
unix_timestamp(calldate) DIV 900 ;

+-+---+-+
| calldate    | count(distinct(clid)) | count(clid) |
+-+---+-+
| 2019-10-14 08:04:36 | 5 |  24 |
| 2019-10-14 08:16:42 | 6 |  14 |
| 2019-10-14 08:30:55 | 7 |  29 |
| 2019-10-14 08:45:10 | 3 |   6 |
| 2019-10-14 09:00:46 | 6 |  19 |
| 2019-10-14 09:35:57 | 4 |   5 |
| 2019-10-14 09:45:05 | 4 |  19 |
| 2019-10-14 10:01:12 | 6 |  45 |

[...]

would be better to have dates starting with "2019-10-14 08:00:00", 
"2019-10-14 08:15:00" etc...

any quick idea? i will search for that anyway.

regards,
andre

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[asterisk-users] CDR extract call numbers on interval on unique callers

2019-11-12 Thread Andre Gronwald 

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 having ;


Unfortunately we have lots of callers who calls multiple times when they 
are forwarded to a queue instead of being answered by a human 
immediately. But to know when we need more people I want to count same 
caller-ids within an interval as one call.


Any ideas how to do this?

kind regards,
andre

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Re: [asterisk-users] where to set fax header field ( unknown, field) in app_fax

2019-08-04 Thread Andre Gronwald 

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 + (UTC)
From: Min Wang 
To: "asterisk-users@lists.digium.com"

Subject: [asterisk-users] Fw: where to set fax header field ( unknown
field)  in app_fax
Message-ID: <473863635.605888.1564784544...@mail.yahoo.com>
Content-Type: text/plain; charset="utf-8"

HI

I used asterisk 13's  app_fax to send and recv fax.
On the received fax/pdf,  I got a fax with header like this:
26-Ju-2019 | 06:03|  my-test | unknown | p.1

The my-test is  set by LOCALHEADERINFO before I call send_fax.
What I am not clear is where does **unknown** come from?
The doc at:Asterisk 13 Application_SendFAX_app_fax - Asterisk Project - 
Asterisk Project Wiki
said:
This application sets the following channel variables:

- LOCALSTATIONID - To identify itself to the remote end

- LOCALHEADERINFO - To generate a header line on each page
- FAXSTATUS
   - SUCCESS
   - FAILED

- FAXERROR - Cause of failure
- REMOTESTATIONID - The CSID of the remote side
- FAXPAGES - Number of pages sent
- FAXBITRATE - Transmission rate
- FAXRESOLUTION - Resolution of sent fax
-


I tried set LOCALSTATIONID, REMOTESTATIONID on before send_fax , but I still 
got unknown in the header.


So my question is where/how to replace the "unknown"?


Thanks

min



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Re: [asterisk-users] Real-time (low latency) monitoring for

2018-12-16 Thread Andre Gronwald 
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
Andre
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Re: [asterisk-users] Real-time (low latency) monitoring for

2018-12-15 Thread Andre Gronwald 
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:

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>1. Real-time (low latency) monitoring for Asterisk (Antony Stone)
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> Message: 1
> Date: Sat, 15 Dec 2018 17:30:37 +0100
> From: Antony Stone 
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Real-time (low latency) monitoring for
> Asterisk
> Message-ID: <201812151730.37546.antony.st...@asterisk.open.source.it>
> Content-Type: text/plain;  charset="us-ascii"
>
> Hi.
>
> Does anyone have any recommendations for a *really* real-time monitoring
> solution for Asterisk?
>
> I'm thinking that something like Grafana (which I've played with for
> another
> purpose, but don't really use yet) can do a good job of displaying the
> data
> with very little latency, but I'm wondering what the best plugin for
> getting
> the data into it might be.
>
> What I'm looking for is a "dashboard" display of Asterisk activity (SIP
> channels currently in use, SIP devices currently registered, duration of
> longest call currently in progress, that sort of thing) but with a very
> low
> latency - so when someone places a call, for example, it shows up on the
> dashboard within a couple of seconds at most.
>
> I'm not looking for anything to show details of individual calls - just
> summaries of all the calls currently going through the system.
>
> I'm currently using Asterisk 13 on Debian Stretch (with, if it matters,
> the
> classic SIP driver, not PJ-SIP), with ODBC real-time CDRs and CEL, I'm
> happy
> with a plugin which connects via AMI if that helps, and the system has no
> queues, no hunt groups, just simple DDI -> SIP account mappings.
>
> Any ideas, people?
>
>
> Thanks,
>
>
> Antony.
>
> --
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Re: [asterisk-users] res_fax_spandsp - information about used protocol t38 or g711?

2018-05-21 Thread Andre Gronwald 

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

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[asterisk-users] setting contact within asterisk -rx 'channel originate local ...'

2018-05-21 Thread Andre Gronwald 

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 tried to set callerid(name) and callerid(all) to something describing, 
the header is set, but the sip-phones (android) seem to display the 
contact header.


can you tell me how to change the contact header? i would like to set up 
some further "ping-services", but for this i need to change the 
displayed name.


regards,
andre


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Re: [asterisk-users] Problem using Android SIP-Client

2017-10-24 Thread Andre Gronwald 

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 format 
gsm since our

native format has changed to 0x8 (alaw)

looks like your android phone uses gsm, but only alaw is supported. just 
try to put gsm as allowed codec.


the sip invite and ok-message would be interesting as well (especially sdp)-

regards,
andre


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Re: [asterisk-users] PJSIP add header not working

2017-10-02 Thread Andre Gronwald 
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, call-id abc123def456)
Goal: Put call-id of call leg a in a separate header of call leg b or 
vice versa (I guess that makes more sense because I have more contorl of 
the headers inside of my pbx, right?).


Following your suggestions I have now call-id b in a separate header of 
the same call-leg, but in call-leg a.


regards,
andre

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[asterisk-users] PJSIP add header not working

2017-10-02 Thread Andre Gronwald 

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", 
"pjsipCallId=313530363933383438363436353930-1gh0bjceo933") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:5] 
Set("PJSIP/10-0006", 
"PJSIP_HEADER(add,X-CID)=313530363933383438363436353930-1gh0bjceo933") 
in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-0006", 
"0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-0006", 
"1?Set(CONNECTEDLINE(num,i)=0xx)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-0006", 
"1?Set(CONNECTEDLINE(name,i)=CID:3x)") in new stack
-- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-0006", 
"0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3x)") in new stack
-- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-0006", 
"0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-0006", 
"PJSIP/0xx@3x,300,T") in new stack

-- Called PJSIP/0xx@3x
<--- Transmitting SIP request (991 bytes) to UDP:217.23.24.100:5060 --->
INVITE sip:0xxx...@sip.provid.er:5060 SIP/2.0
Via: SIP/2.0/UDP 
192.168.253.185:15070;rport;branch=z9hG4bKPj453d15e0-de58-4945-8b95-d05b16b9e4c3
From: 
;tag=080788ac-7c10-4cf3-86b3-359764ffb5a2

To: 
Contact: 
Call-ID: de41b93b-51d8-44b5-9c34-f2c0928192b0
CSeq: 1519 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, MESSAGE, REFER

Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.1.10(14.6.2)
Content-Type: application/sdp
Content-Length:   308

v=0
o=- 1719768133 1719768133 IN IP4 192.168.253.185
s=Asterisk
c=IN IP4 192.168.253.185
t=0 0
m=audio 55112 RTP/AVP 107 9 8 3 101
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Received SIP response (559 bytes) from UDP:217.23.24.100:5060 --->
[...]

##




But I can't see that header anywhere in my call-legs. What am I missing?


kind regards,
andre

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Re: [asterisk-users] double NAT - one way audio

2017-03-20 Thread Andre Gronwald 
> 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 generally messes things up. Also make sure your asterisk
can
> get correct public IP - "externip=" ...
> Martin

Would like to disable it, but this hardware is not supporting it. Neither
by standard GUI nor by hidden menu... :-(

regards,
andre
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Re: [asterisk-users] double NAT - one way audio

2017-03-15 Thread Andre Gronwald 
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

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<andregronwal...@googlemail.com>>
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Re: [asterisk-users] double NAT - one way audio

2017-03-13 Thread Andre Gronwald 
Hi Glenn,
unfortunately there is no bridge mode or any comparable mode available. I
am using the same router (but another type) on my private homenetwork with
another router at the back (=> same architecture as in this failing
scenario), but everything works fine.
There are only two differences:
1. Another Type (w724v Type B instead of w724v Type A)
2. No VoIP services used by w724v (which is on Type A hardware currently
the case, maybe disablling them helps?).

I will check to switch 2., but that is not easily doable because there are
productive numbers used... The asterisk installation is currently in
development...

regards,
andre

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<andregronwal...@googlemail.com>>
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[asterisk-users] double NAT - one way audio

2017-03-11 Thread 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 
RTP-ports 51000-51999.
Those ports are forwarded on DSL-router  to the bintec router and from 
the bintec router to asterisk.


what I see is the Invite from provider goes to 192.168.2.1 and rtp port 
7070. my asterisk responds with audio to be sent to ip address 
80.142.12.12 port 51242.
Afterwards RTP goes from 10.17.46.99:51242 to 192.168.2.1:7070. but the 
RTP back is not coming in.


I would expect the RTP traffic to be sent to 80.142.12.12 (intetnal 
192.168.2.1) port 51242 - then i would have successfully two way audio.

But why is port 7070 used?

The DSL-router is a speedport w724v type A.

regards,
andre

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Re: [asterisk-users] First SIP-registering succeeds, second doesn't

2017-02-13 Thread Andre Gronwald 

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, at which my freepbx installation is located. However, NAT etc. 
seems to work fine.

BUT: Something is not working...:
When registering my sip-trunk towards my provider (3 different 
providers, all behave comparable), everything works at first. Calls 
are possible. But after some time, when the next REGISTER happens, the 
answer of my provider is sent towards the wrong port. My freePBX 
listens on 55060, where the first registration request are answered as 
they should. in the second registration request wrong ports are used. 
besides this, Header "Expires" is set to "0" and no "Allows" are 
listed...

[...]



Any suggestions how to fix this? Or at least any idea what causes this?

regards,
andre



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[asterisk-users] First SIP-registering succeeds, second doesn't

2017-02-13 Thread 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, at which my freepbx installation is located. However, NAT etc. 
seems to work fine.

BUT: Something is not working...:
When registering my sip-trunk towards my provider (3 different 
providers, all behave comparable), everything works at first. Calls are 
possible. But after some time, when the next REGISTER happens, the 
answer of my provider is sent towards the wrong port. My freePBX listens 
on 55060, where the first registration request are answered as they 
should. in the second registration request wrong ports are used. besides 
this, Header "Expires" is set to "0" and no "Allows" are listed...


Good case:
2017/02/10 20:40:59.236563 192.193.194.99:55060 -> 217.10.79.9:5060
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj7d031cfc-7602-4ac1-a264-bcf8d267500b

From: sip:custumoeri...@sipgate.de;tag=b14c0d37-ef26-4dc5-b112-caf0c12a51f1
To: sip:custumoeri...@sipgate.de
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2367 REGISTER
Contact:
Expires: 300
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE

Max-Forwards: 70
User-Agent: FPBX-13.0.190.12(13.13.1)
Content-Length: 0

2017/02/10 20:40:59.300873 217.10.79.9:5060 -> 192.193.194.99:55060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj7d031cfc-7602-4ac1-a264-bcf8d267500b

From: sip:custumoeri...@sipgate.de;tag=b14c0d37-ef26-4dc5-b112-caf0c12a51f1
To: sip:custumoeri...@sipgate.de;tag=86e53dd608d1c001e0b8060625977563.11b6
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2367 REGISTER
WWW-Authenticate: Digest realm="sipgate.de", 
nonce="WJ4Yd1ieF0tFmpj4o617dwQB2X5d9sKR"

Content-Length: 0

2017/02/10 20:40:59.301187 192.193.194.99:55060 -> 217.10.79.9:5060
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj5fb9a513-f49b-4823-8c63-6f0a38314d89

From: sip:custumoeri...@sipgate.de;tag=b14c0d37-ef26-4dc5-b112-caf0c12a51f1
To: sip:custumoeri...@sipgate.de
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2368 REGISTER
Contact:
Expires: 300
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE

Max-Forwards: 70
User-Agent: FPBX-13.0.190.12(13.13.1)
Authorization: Digest username="custumoerIDe0", realm="sipgate.de", 
nonce="WJ4Yd1ieF0tFmpj4o617dwQB2X5d9sKR", uri="sip:sipgate.de:5060", 
response="1067666a187dd3413fcecede9820e87c"

Content-Length: 0

2017/02/10 20:40:59.367186 217.10.79.9:5060 -> 192.193.194.99:55060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj5fb9a513-f49b-4823-8c63-6f0a38314d89

From: sip:custumoeri...@sipgate.de;tag=b14c0d37-ef26-4dc5-b112-caf0c12a51f1
To: sip:custumoeri...@sipgate.de;tag=86e53dd608d1c001e0b8060625977563.262c
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2368 REGISTER
Contact: ;expires=300
Content-Length: 0

Bad case:
2017/02/10 20:43:08.324919 192.193.194.99:55060 -> 217.10.79.9:5060
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj9fb980fb-f4a3-465b-80bd-f25e668b723c

From: sip:custumoeri...@sipgate.de;tag=a5226407-51ae-41c9-845e-4791428aa44f
To: sip:custumoeri...@sipgate.de
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2369 REGISTER
Contact:
Expires: 0
Max-Forwards: 70
User-Agent: FPBX-13.0.190.12(13.13.1)
Content-Length: 0

2017/02/10 20:43:08.387098 217.10.79.9:5060 -> 192.193.194.99:61276
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj9fb980fb-f4a3-465b-80bd-f25e668b723c

From: sip:custumoeri...@sipgate.de;tag=a5226407-51ae-41c9-845e-4791428aa44f
To: sip:custumoeri...@sipgate.de;tag=86e53dd608d1c001e0b8060625977563.1e67
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2369 REGISTER
WWW-Authenticate: Digest realm="sipgate.de", 
nonce="WJ4Y+FieF8wCvpDGuyqz40rHntgzb6xy"

Content-Length: 0

Any suggestions how to fix this? Or at least any idea what causes this?

regards,
andre


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Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

2016-10-17 Thread Andre Gronwald 
> │ │
▒   +3.518524   │200 OK (SDP) │ 
│ │ │
▒ 18:27:06.222761   │ <<< 
│ │ │ │
│   +0.483722   │ │ │ ACK 
│ │
│ 18:27:06.706483   │ │ │ 
──> │ │
│   +3.516816   │200 OK (SDP) │ 
│ │ │
│ 18:27:10.223299   │ <<< 
│ │ │ │
│   +0.484398   │ │ ACK │ 
│ │
│ 18:27:10.707697   │ │ 
<── │ │ │
│  +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 
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 sngrep is the following options-flow:
192.168.3.50:55060 -> 217.10.79.9:5060
217.10.79.9:5060 -> 192.168.3.50:48757 (200 OK)

shouldn't sipgate answer on the same port that the communication 
initiated??? in this case 55060?




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Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

2016-10-15 Thread Andre Gronwald 

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 sngrep is the following options-flow:
192.168.3.50:55060 -> 217.10.79.9:5060
217.10.79.9:5060 -> 192.168.3.50:48757 (200 OK)

shouldn't sipgate answer on the same port that the communication 
initiated??? in this case 55060?


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Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

2016-10-15 Thread Andre Gronwald 

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 packets? that 
doesn't sound logical to me. What do you think?


regards,
andre

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Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

2016-10-15 Thread Andre Gronwald 

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, everything to port 51000-55999 to device 
192.168.3.99 (same ports)


other direction is totally open.

I observed that all sip calls are closed exactly after 32s. call is 
disconnected on calling side as well... seems to be a timeout issue.


here i have some debug logs. I see lot of requests from asterisk to 
sipgate.de, which are not answered. but communication is going fine in 
both directions (otherwise registration would not be possible?):



<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0@80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP 
217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0

Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: 
Record-Route: 
Record-Route: 
From: "02363361779" ;tag=as02fa8fcc
To: 
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
CSeq: 103 INVITE
Contact: 
max-forwards: 66
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Content-Type: application/sdp
Content-Length:   394

v=0
o=root 15363811 15363812 IN IP4 192.168.2.1
s=sipgate VoIP GW
c=IN IP4 192.168.2.1
t=0 0
m=audio 7070 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0

Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: 
Record-Route: 
Record-Route: 
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "09" ;tag=as02fa8fcc
To: 
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Content-Length:  0

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0

Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: 
Record-Route: 
Record-Route: 
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "09" ;tag=as02fa8fcc
To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: 
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE

Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0@80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP 
217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0

Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: 
Record-Route: 
Record-Route: 
From: "09" ;tag=as02fa8fcc
To: 
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
CSeq: 103 INVITE
Contact: 
max-forwards: 66
supported:

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

2016-10-15 Thread Andre Gronwald 

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 11:20:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now Reachable. 
RTT: 433.814 msec
[2016-10-15 11:20:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Reachable
[2016-10-15 11:21:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now Unreachable. 
RTT: 0.000 msec
[2016-10-15 11:21:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Unreachable
[2016-10-15 11:30:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now Reachable. 
RTT: 439.006 msec
[2016-10-15 11:30:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Reachable
[2016-10-15 11:31:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now Unreachable. 
RTT: 0.000 msec
[2016-10-15 11:31:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Unreachable
[2016-10-15 11:40:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now Reachable. 
RTT: 433.426 msec
[2016-10-15 11:40:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Reachable
[2016-10-15 11:41:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now Unreachable. 
RTT: 0.000 msec
[2016-10-15 11:41:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Unreachable


I think that the times are matching exactly the qualify frequency and 
registry expiration - expiration is set to 600s, and qualify frequency 
to 50s. Seems that the qualify requests are not supported (this is the 
case for the other provider as well!). So maybe I should work without 
sip qualify.


Besides this I have another curiousity:
One call:
-- Executing [s@app-announcement-1:3] 
Wait("PJSIP/pjsip_sipgate-0019", "1") in new stack
   > 0x7fabf004bfd0 -- Probation passed - setting RTP source 
address to 217.10.77.109:16248

Another call:
-- Executing [s@app-announcement-1:3] 
Wait("PJSIP/pjsip_sipgate-001a", "1") in new stack
   > 0x7fabf0070bb0 -- Probation passed - setting RTP source 
address to 192.168.2.1:7074


??? 217.10.77.109 is sipgate.de -> ok. 192.168.2.1 is my 
vDSL-access-router ??? Why does the RTP source address changes? that 
must not happen.



And another observation: I am registered to sipgate.de, fine. Incoming 
call is processed,  announcement is played. But when the caller hangs up 
asterisk is not recognizing it. it takes about 16s until the channel is 
closed after hangup?



regards,
andre
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Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

2016-10-15 Thread Andre Gronwald 

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 (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57 time=46.8 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=5 ttl=57 time=47.1 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=6 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=7 ttl=57 time=47.1 ms
^C
--- sipgate.de ping statistics ---
7 packets transmitted, 7 received, 0% packet loss, time 6360ms
rtt min/avg/max/mdev = 46.406/46.809/47.191/0.393 ms
[root@freepbx asterisk]#


this high RTT appears only sometimes.  After removing STUN-server it 
looks better, did two test calls right now, both gone through 
immediately. At the end of the second test call I see:


-- Executing [s@app-announcement-1:5] 
Playback("PJSIP/pjsip_sipgate-0003", 
"custom/araz01/07-polly,noanswer") in new stack
--  Playing 'custom/araz01.alaw' 
(language 'en')
-- Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now 
Reachable.  RTT: 493.094 msec

  == Endpoint pjsip_sipgate is now Reachable
--  Playing 'custom/07-polly.slin' 
(language 'en')
-- Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now 
Unreachable.  RTT: 0.000 msec

*  == Endpoint pjsip_sipgate is now Unreachable*


Why do I have that loss of registrations?

here my pjsip config for sipgate.de:

freepbx*CLI> pjsip show registration pjsip_sipgate

  
  

==

 pjsip_sipgate/sip:sipgate.de:5060 pjsip_sipgate Registered

 ParameterName: ParameterValue
 
 auth_rejection_permanent : true
 client_uri   : sip:263614...@sipgate.de:5060
 contact_user : 2636146e0
 endpoint :
 expiration   : 600
 fatal_retry_interval : 0
 forbidden_retry_interval : 0
 line : false
 max_retries  : 10
 outbound_auth: pjsip_sipgate
 outbound_proxy   :
 retry_interval   : 60
 server_uri   : sip:sipgate.de:5060
 support_path : false
 transport: 0.0.0.0-udp

Remind: Endpoint is currently unreachable, but asterisk shows 
"Registered". Test call fails at this moment.



regards,
andre
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Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

2016-10-15 Thread Andre Gronwald 
Very interesting: I have another provider configured, that was not 
reachable as well. I disabled the STUN-server (external STUN server), 
and now the second registration works fine, BUT with the same "error" 
messages (unreachable etc) as the other provider. But in contrast the 
number 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 
received from 'sip:sipgate.de:5060' on registration attempt to 
'sip:263614...@sipgate.de:5060', retrying in '60'
-- Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now 
Reachable.  RTT: 434.393 msec

  == Endpoint pjsip_sipgate is now Reachable

so it is somewhat clear, why i get a busy, because the endpoint is not 
reachable. But WHY is the endpoint not reachable?


Regarding the architecture: I have two routers cascaded, that is 
unfortunately necessary. On the first router (vDSL-access router) I 
have forwarded nearly everything to the second router (Bintec rj 353), 
where a port forwarding for relevant ports (sip and pjsip (udp and 
tcp), rtp (udp)) is configured. IF a call goes through, nearly 
everything is working (audio only incoming, but that is another issue).


STUN is configured. FreePBX Firewall is disabled.

Kind regards,
andre




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[asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

2016-10-15 Thread Andre Gronwald 

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:


   


==

 pjsip_sipgate/sip:sipgate.de:5060 pjsip_sipgate Registered


Calling the registered number is even successfully shown in asterisk (it 
is a freepbx installation).
But when doing a second call the number is busy ("provider" busy, I 
don't see anything in asterisk verbose mode).

Sending a pjsip unregister results in the following messages:

[2016-10-15 10:03:22] WARNING[10162]: 
res_pjsip_outbound_registration.c:761 schedule_retry: No response 
received from 'sip:sipgate.de:5060' on registration attempt to 
'sip:263614...@sipgate.de:5060', retrying in '60'
-- Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now 
Reachable.  RTT: 434.393 msec

  == Endpoint pjsip_sipgate is now Reachable

so it is somewhat clear, why i get a busy, because the endpoint is not 
reachable. But WHY is the endpoint not reachable?


Regarding the architecture: I have two routers cascaded, that is 
unfortunately necessary. On the first router (vDSL-access router) I have 
forwarded nearly everything to the second router (Bintec rj 353), where 
a port forwarding for relevant ports (sip and pjsip (udp and tcp), rtp 
(udp)) is configured. IF a call goes through, nearly everything is 
working (audio only incoming, but that is another issue).


STUN is configured. FreePBX Firewall is disabled.

Kind regards,
andre


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Re: [asterisk-users] Handle a call if one phone of a ring, group is busy

2016-02-28 Thread Andre Gronwald 

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 17 = SIPcode 486=user busy here
exten => general,200,Return()
...


that works as well and you can specify how many calls are allowed.
regards,
andre

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Re: [asterisk-users] asterisk-users Digest, Vol 106, Issue 41

2013-05-30 Thread Andre Gronwald 
hi,
try

exten = .,n,System(wget -P /var/log/asterisk/wgets
'http://theUrlYouWantToCall' )

kind regards,
andre

Am 30.05.2013 19:00, schrieb asterisk-users-requ...@lists.digium.com:
 Message: 9
 Date: Thu, 30 May 2013 15:06:59 +
 From: Salaheddine Elharit salah.elharit...@gmail.com
 Subject: [asterisk-users] how to launch a URl when dialing a number
 To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
 Message-ID:
   CAHexamt4=evwlj7pn90mgqvirn-_mnyh6p2tdhtpq4d89e2...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Hello

 i want to luanch an URL in my PC when i call a number  like below

 exten = 066104,1,Set(CALLERID(number)=52xxx)
 exten = 066104,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten
 = 
 066104,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
 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


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[asterisk-users] addressing peers dynamically

2012-11-19 Thread Andre Gronwald 
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 type 'SIP' (cause 20 - Subscriber absent)
Is there a way to avoid those messages? I think about something like a
virtual queue (please excuse if the wording is incorrect, i am not in too
deep to asterisk, i am more firm with genesys) that is addressed and peers
are registering to that queue.
is that the right path, or am i barking the wrong tree?

regards,
andre
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Re: [asterisk-users] addressing peers dynamically

2012-11-19 Thread Andre Gronwald 

Am 19.11.2012 19:00, schrieb asterisk-users-requ...@lists.digium.com:
 Subject: Re: [asterisk-users] addressing peers dynamically To:
 Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com Message-ID:
 50aa2586.80...@digium.com Content-Type: text/plain;
 charset=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 always, resulting in error messages:
  Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  Is there a way to avoid those messages? I think about something like a
  virtual queue (please excuse if the wording is incorrect, i am not in
  too deep to asterisk, i am more firm with genesys) that is addressed and
  peers are registering to that queue.
  is that the right path, or am i barking the wrong tree?
 Is there any particular reason you don't want to do this or is it just 
 because you get the Unable to create channel message? There's nothing 
 really *wrong* with that message in your case.
it is just because i think that something is not wrong (which is
correct, because i address a currently not existing peer). and if there
is a way to handle it better, then i would like to know it (virtual
queues is just oversized, but maybe there is a simple usage of
addressing only registered peers...

regards,
andre

-- 
Andre Gronwald
andregronwal...@gmail.com
andre.gronw...@gmx.de PGP-0x9CDEE439


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