Re: [asterisk-users] Asterisk LTS segment faults

2014-10-08 Thread Ryan Wagoner 
On Wed, Oct 8, 2014 at 9:35 AM, Grant Bagdasarian  wrote:

> Hello,
>
>
>
> Does anyone know how frequent segment faults occur in the current LTS
> release (version 11) and in the future LTS release (version 13)?
>
> We are currently using 1.6, which frequently throws unexplained segment
> faults, that’s why we are considering to upgrade to the latest LTS version.
>
>
>
>
>
I was having crashes at least once a month with Asterisk 1.6. Each time I
would upgrade to fix one issue another would appear. I moved to Asterisk
1.8 LTS when it was released and haven't looked back. I have around 700
endpoints registered and we handle over 10k calls per day. Even with
Asterisk 1.8 I was running into a hung channels every few months when using
a Sangoma card with chan_dahdi. About 6 months ago I switched over to a
Cisco gateway for the PRIs and am only using chan_sip with Asterisk. The
result has been rock solid performance.

Ryan
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Re: [asterisk-users] Asterisk Phone ( Telecom feature )

2014-10-08 Thread Steve Edwards 

On Wed, 8 Oct 2014, Dania Asi wrote:


Dear Mr. Adam,

Thank you for you kind words and for judging me.

I am a system integrator and I have a whale clients in UAE , I will not 
proceed further in dealing with Asterisk because of the lack of support 
and because of the rude emails.


I have no idea what is wrong with you people. And I hope you get well 
soon from whatever is happening to you.


You are misjudging this community.

It is staffed by volunteers. Their range of experience and knowledge range 
from rank novices to experts that you couldn't afford to hire. They do it 
out of the kindness of their hearts and because they have been helped in 
the past.


I've been on this list for about 10 years and continue to be amazed at the 
knowledge and ingenuity exchanged for 'free.'


People are more likely to help if you include sufficient information and 
demonstrate that you have at least tried to resolve your issue. Nobody is 
interested in doing your work (be it homework or work-work) for you.


Be thankful there are no 'Carl J. Lydick's on this list. He was a 
brilliant VMS programmer back in the day. If you asked an intelligent 
question, you would get an amazing answer. If you asked a stupid question 
or indicated you had not invested any effort, you might receive an 
obfuscated command that would erase your disks if you were to ignorantly 
'cut-n-paste' the command.


Asterisk is a fantastic product. It is like a Lego or Erector set. You 
have all the bits. It is up to you to fulfill the vision in your head.


If you are looking for paid support or a canned solution, it is available. 
You just looked in the wrong place.


Maybe it's just a 'cultural misunderstanding.'

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Asterisk Phone ( Telecom feature )

2014-10-08 Thread Don Kelly 
Although you may have found them rude and unhelpful, I think most of the
responses had a common theme-we need more information about what you have
and what you want to do before we can help at all.

 

  --Don

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Wednesday, October 08, 2014 1:40 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 

This is free software supported by volunteers who are helping you because
somewhere in the past someone helped them or perhaps they were just brought
up properly.

No one really cares about how many clients you may have. No one here will
make any money from that.

Have you read the free books on Asterisk?
Have you at least followed the installation instructions to some point where
you are stuck.

I don't see where you have explained what you want to do with the PBX once
you have it set up. There are a lot of different applications possible -
Small office, enterprise with many branches, call center with thousands of
incoming or outgoing calls, etc.

Have you looked at what phones you want to support - brand, features?

Have you selected a type of trunk that your telephone company offers? Do
they have a document describing how to interface an Asterisk PBX to their
trunks?

What were you expecting a total stranger to do for you when you asked your
question?
If you want to hire a consultant to set up your first few clients and train
you, than you should say so.
Free advice comes in many forms but there is a limit about how much free
time each one of us has and you will get different people helping you
depending on the question that you ask. We are not all experts at
everything. Most of us are people like you or end-users who are supporting
their own company's phone system.

Ron


On 08/10/2014 12:34 AM, Dania Asi wrote:

Dear Mr. Adam,

 

Thank you for you kind words and for judging me.

 

I am a system integrator and I have a whale clients in UAE , I will not
proceed further in dealing with Asterisk because of the lack of support and
because of the rude emails. 

 

I have no idea what is wrong with you people. And I hope you get well soon
from whatever is happening to you.

 

 

Best Wishes,

 

Dania Abu Asi

Sales Executive Engineer





Future Trends Establishment

Abu Dhabi - U.A.E.

Mob : +971 50 4948363

Off : +971 2 6730666

Fax : +971 2 6734888

 

From: Adam Goldberg [mailto:a...@agp-llc.com] 
Sent: Tuesday, October 7, 2014 8:51 PM
To: Dania Asi
Subject: FW: [asterisk-users] Asterisk Phone ( Telecom feature )

 





I suggest that your question amounts to "please do my homework for me."
This may be understandable given that you are a recent grad and probably
don't have much experience in business communication and/or Asterisk
complexities. 

 

You cannot expect a mailing list to rush to answer vague, unanswerable
questions -- nor emails that don't show that you've tried to answer the
question first.  

 

Consider, if I asked: 

 

"I don't understand how to set up Asterisk.  Can someone
tell me how to do that?" 

vs. 

 

"I have a Dell R210-II with 32g of memory and two gigabit ethernet
interfaces, I've installed Asterisk from the FreePBX Distro v9.99 and have
an assortment of Polycom and Snom IP phones.  I've configured paging as
described in http://wiki.snom.com/Interoperability/PBX/Asterisk and
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page, and it is working for
the Snom phones but not the Polycom phones.  Can someone point me at what
it's going to take to make the Polycom phones work?"

 

I'd expect to get attempts at an answer to the second one, but would expect
snide and rude comments (at best) to the first one.

 

Adam Goldberg

AGP, LLC

+1-202-507-9900

 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Tuesday, October 07, 2014 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 

JG confirmed that "it" is possible, but "it" has not been defined.

 

Without knowing what kind of instruments you are using, a possible "it"

would be for a party to dial a 4-digit extension number to talk to someone
internally, completing a call without using the PRI trunks.

 

  --Don

 

-Original Message-

From:  
asterisk-users-boun...@lists.digium.com

[ 
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dania Asi

Sent: Tuesday, October 07, 2014 3:41 AM

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Cc: 'Irene Galera'; 'Maysara Orabi';  
moham...@futuretrendsest.com

Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 

Dear Mr. Mit

Re: [asterisk-users] Asterisk Phone ( Telecom feature )

2014-10-08 Thread Ron Wheeler 
This is free software supported by volunteers who are helping you 
because somewhere in the past someone helped them or perhaps they were 
just brought up properly.


No one really cares about how many clients you may have. No one here 
will make any money from that.


Have you read the free books on Asterisk?
Have you at least followed the installation instructions to some point 
where you are stuck.


I don't see where you have explained what you want to do with the PBX 
once you have it set up. There are a lot of different applications 
possible - Small office, enterprise with many branches, call center with 
thousands of incoming or outgoing calls, etc.


Have you looked at what phones you want to support - brand, features?

Have you selected a type of trunk that your telephone company offers? Do 
they have a document describing how to interface an Asterisk PBX to 
their trunks?


What were you expecting a total stranger to do for you when you asked 
your question?
If you want to hire a consultant to set up your first few clients and 
train you, than you should say so.
Free advice comes in many forms but there is a limit about how much free 
time each one of us has and you will get different people helping you 
depending on the question that you ask. We are not all experts at 
everything. Most of us are people like you or end-users who are 
supporting their own company's phone system.


Ron


On 08/10/2014 12:34 AM, Dania Asi wrote:


Dear Mr. Adam,

Thank you for you kind words and for judging me.

I am a system integrator and I have a whale clients in UAE , I will 
not proceed further in dealing with Asterisk because of the lack of 
support and because of the rude emails.


I have no idea what is wrong with you people. And I hope you get well 
soon from whatever is happening to you.


*Best Wishes,*

**

*Dania Abu Asi*

Sales Executive Engineer


*Future Trends Establishment***

Abu Dhabi - U.A.E.

Mob : +971 50 4948363

Off : +971 2 6730666

Fax : +971 2 6734888

*From:* Adam Goldberg [mailto:a...@agp-llc.com]
*Sent:* Tuesday, October 7, 2014 8:51 PM
*To:* Dania Asi
*Subject:* FW: [asterisk-users] Asterisk Phone ( Telecom feature )



I suggest that your question amounts to "please do my homework for 
me."  This may be understandable given that you are a recent grad and 
probably don't have much experience in business communication and/or 
Asterisk complexities.


You cannot expect a mailing list to rush to answer vague, unanswerable 
questions -- nor emails that don't show that you've tried to answer 
the question first.


Consider, if I asked:

"I don't understand how to set up Asterisk.  Can 
someone tell me how to do that?"


vs.

"I have a Dell R210-II with 32g of memory and two gigabit ethernet 
interfaces, I've installed Asterisk from the FreePBX Distro v9.99 and 
have an assortment of Polycom and Snom IP phones. I've configured 
paging as described in 
http://wiki.snom.com/Interoperability/PBX/Asterisk and 
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page, and it is 
working for the Snom phones but not the Polycom phones.  Can someone 
point me at what it's going to take to make the Polycom phones work?"


I'd expect to get attempts at an answer to the second one, but would 
expect snide and rude comments (at best) to the first one.


Adam Goldberg

AGP, LLC

+1-202-507-9900

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly

Sent: Tuesday, October 07, 2014 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

JG confirmed that "it" is possible, but "it" has not been defined.

Without knowing what kind of instruments you are using, a possible "it"

would be for a party to dial a 4-digit extension number to talk to 
someone internally, completing a call without using the PRI trunks.


  --Don

-Original Message-

From: asterisk-users-boun...@lists.digium.com 



[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dania Asi

Sent: Tuesday, October 07, 2014 3:41 AM

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Cc: 'Irene Galera'; 'Maysara Orabi'; moham...@futuretrendsest.com 



Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

Dear Mr. Mitual,

Kindly check the attached mail where Mr. JG confirmed to me that is 
possible and I already informed my client of that.


Dear Mr. Steve,

I am not expecting a mailing list to do any work for me. All I was 
asking is for you to guide me because this is the first time we deal 
with Asterisk phones.


Best Wishes,

Dania Abu Asi

Sales Executive Engineer

Future Trends Establishment

Abu Dhabi - U.A.E.

Mob : +971 50 4948363

Off : +971 2 6730666

Fax : +971 2 6734888

-Or

Re: [asterisk-users] Asterisk Phone ( Telecom feature )

2014-10-08 Thread Dania Asi 
Dear Mr. Adam,

 

Thank you for you kind words and for judging me.

 

I am a system integrator and I have a whale clients in UAE , I will not
proceed further in dealing with Asterisk because of the lack of support and
because of the rude emails. 

 

I have no idea what is wrong with you people. And I hope you get well soon
from whatever is happening to you.

 

 

Best Wishes,

 

Dania Abu Asi

Sales Executive Engineer



Future Trends Establishment

Abu Dhabi - U.A.E.

Mob : +971 50 4948363

Off : +971 2 6730666

Fax : +971 2 6734888

 

From: Adam Goldberg [mailto:a...@agp-llc.com] 
Sent: Tuesday, October 7, 2014 8:51 PM
To: Dania Asi
Subject: FW: [asterisk-users] Asterisk Phone ( Telecom feature )

 

I suggest that your question amounts to "please do my homework for me."
This may be understandable given that you are a recent grad and probably
don't have much experience in business communication and/or Asterisk
complexities.

 

You cannot expect a mailing list to rush to answer vague, unanswerable
questions -- nor emails that don't show that you've tried to answer the
question first.  

 

Consider, if I asked: 

 

"I don't understand how to set up Asterisk.  Can someone
tell me how to do that?" 

vs. 

 

"I have a Dell R210-II with 32g of memory and two gigabit ethernet
interfaces, I've installed Asterisk from the FreePBX Distro v9.99 and have
an assortment of Polycom and Snom IP phones.  I've configured paging as
described in http://wiki.snom.com/Interoperability/PBX/Asterisk and
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page, and it is working for
the Snom phones but not the Polycom phones.  Can someone point me at what
it's going to take to make the Polycom phones work?"

 

I'd expect to get attempts at an answer to the second one, but would expect
snide and rude comments (at best) to the first one.

 

Adam Goldberg

AGP, LLC

+1-202-507-9900

 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Tuesday, October 07, 2014 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 

JG confirmed that "it" is possible, but "it" has not been defined.

 

Without knowing what kind of instruments you are using, a possible "it"

would be for a party to dial a 4-digit extension number to talk to someone
internally, completing a call without using the PRI trunks.

 

  --Don

 

-Original Message-

From:  
asterisk-users-boun...@lists.digium.com

[ 
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dania Asi

Sent: Tuesday, October 07, 2014 3:41 AM

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Cc: 'Irene Galera'; 'Maysara Orabi';  
moham...@futuretrendsest.com

Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 

Dear Mr. Mitual,

 

Kindly check the attached mail where Mr. JG confirmed to me that is possible
and I already informed my client of that.

 

 

Dear Mr. Steve,

 

I am not expecting a mailing list to do any work for me. All I was asking is
for you to guide me because this is the first time we deal with Asterisk
phones.

 

 

Best Wishes,

 

Dania Abu Asi

Sales Executive Engineer

 

Future Trends Establishment

Abu Dhabi - U.A.E.

Mob : +971 50 4948363

Off : +971 2 6730666

Fax : +971 2 6734888

 

-Original Message-

From:  
asterisk-users-boun...@lists.digium.com

[ 
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes

Sent: Tuesday, October 7, 2014 12:34 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Cc: Irene Galera; Maysara Orabi;  
moham...@futuretrendsest.com

Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 

On 7 Oct 2014, at 09:24, Dania Asi < 
da...@futuretrendsest.com> wrote:

> Kindly note that I asked about the capability of the phones and now I 

> am asking about the way I can do it to my client's phones, because he 

> is asking for a demonstration.

 

Yet you've not even told us the phones in use. You can't just expect a
mailing list to do your work for you. You need to look at the handsets and
see what they support, and how they support it. You don't even say if the
handsets are SIP or not, the PSTN connectivity is pretty irrelevant.

 

> Sales Executive Engineer

 

That explains that one.

 

Steve

--

_

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Re: [asterisk-users] Asterisk LTS segment faults

2014-10-08 Thread Kevin Larsen 
> I'm not going to say you will never experience a crash in a supported
> version of Asterisk. The issue tracker would quickly make me out to be
> a liar.
> 
> On the other hand, if you are using a supported version of Asterisk,
> then you can report the issue [1], get a backtrace [2], and we'll do
> our best to fix the bug.

To add on purely anecdotal evidence, I used to be running 1.6 at a couple 
of sites and would have an unexplained crash about once or twice a month. 
Got all my sites moved to Asterisk 11 with pretty much the same dialplans 
and the stability has been much better. Even if a crash is still a 
possibility, the current versions are, in my limited experience, much more 
stable in day to day operations. It is worth it to upgrade.-- 
_
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Re: [asterisk-users] Asterisk LTS segment faults

2014-10-08 Thread Matthew Jordan 
On Wed, Oct 8, 2014 at 8:35 AM, Grant Bagdasarian  wrote:
> Hello,
>
>
>
> Does anyone know how frequent segment faults occur in the current LTS
> release (version 11) and in the future LTS release (version 13)?
>
> We are currently using 1.6, which frequently throws unexplained segment
> faults, that’s why we are considering to upgrade to the latest LTS version.
>

I'm not going to say you will never experience a crash in a supported
version of Asterisk. The issue tracker would quickly make me out to be
a liar.

On the other hand, if you are using a supported version of Asterisk,
then you can report the issue [1], get a backtrace [2], and we'll do
our best to fix the bug.

The question isn't whether or not you will ever experience a problem:
the question is, should you be running a production system on
something the project no longer supports?

[1] https://issues.asterisk.org/jira
[2] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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[asterisk-users] Asterisk LTS segment faults

2014-10-08 Thread Grant Bagdasarian 
Hello,

Does anyone know how frequent segment faults occur in the current LTS release 
(version 11) and in the future LTS release (version 13)?
We are currently using 1.6, which frequently throws unexplained segment faults, 
that's why we are considering to upgrade to the latest LTS version.


-- 
_
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Re: [asterisk-users] Grandstream GXP2160 + SRTP

2014-10-08 Thread Jonas Kellens 

On 07-10-14 12:32, Jonas Kellens wrote:

Hello,

I am trying to setup a Grandstream GXP2160 IP-phone with secure 
calling (SRTP).


Secure signaling SSIP for registration is working great !

I follow this guide : 
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial


But when I try to make a call with SRTP, I get stuck. There is an 
initial INVITE which is anwered with a 401. There should follow a new 
INVITE with a nonce, but this does not happen. Any idea why ? Is it 
the Grandstream IP-phone ??




<--- SIP read from TLS:my.pub.lic.ip:53416 --->
INVITE sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias
From: ;tag=263162018
To: 
Call-ID: 1695864968-506...@bjc.bgi.b.bae
CSeq: 50 INVITE
Contact: 
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.2.9
Privacy: none
P-Preferred-Identity: 
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, 
REFER, UPDATE, MESSAGE

Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 522

v=0
o=testacc77005 8004 8000 IN IP4 192.168.1.104
s=SIP Call
c=IN IP4 192.168.1.104
t=0 0
m=audio 5020 RTP/SAVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:8m7ZfG+0t3KBFGK40IfDO11SZ6D54glKKIwdgo00|2^32
a=crypto:2 AES_CM_128_HMAC_SHA1_32 
inline:nn+id/sSK7OErMfnZZduKNPLejpscxx1vUQB2seO|2^32



<--- Reliably Transmitting (NAT) to my.pub.lic.ip:53416 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 
192.168.1.104:5068;branch=z9hG4bK60724585;alias;received=my.pub.lic.ip;rport=53416

From: ;tag=263162018
To: ;tag=as1e527556
Call-ID: 1695864968-506...@bjc.bgi.b.bae
CSeq: 50 INVITE
Server: mydomain
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="mydomain.be", 
nonce="13b47342"

Content-Length: 0


<--- SIP read from TLS:my.pub.lic.ip:53416 --->
ACK sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias
From: ;tag=263162018
To: ;tag=as1e527556
Call-ID: 1695864968-506...@bjc.bgi.b.bae
CSeq: 50 ACK
Content-Length: 0



Hello,

I seem to have the same problem with Snom 370 IP-phone. Registration 
works fine ! But I can not make calls with encrypted rtp.



<--- SIP read from TLS:my.pub.lic.ip:1068 --->
INVITE sip:0123123...@ast.ser.ver.ip;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.107:1068;branch=z9hG4bK-gxm8w1q7l2co;rport
From: ;tag=zdwiwg10qx
To: 
Call-ID: 3c2679977b67-9j0euqvseh5v
CSeq: 1 INVITE
Max-Forwards: 70
Contact: ;reg-id=1
X-Serialnumber: 0004132E2809
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/8.4.35
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Call-Info: ;appearance-index=1
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 632

v=0
o=root 1052895538 1052895538 IN IP4 192.168.1.107
s=call
c=IN IP4 192.168.1.107
t=0 0
m=audio 65418 RTP/SAVP 8 3 18 99 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:KiXn5H+mKwavoDNa1PfnBqPoODTnxK6hOlWSNJM7

a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:99 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=audio 65418 RTP/AVP 8 3 18 99 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:99 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<->



<--- Reliably Transmitting (NAT) to my.pub.lic.ip:1068 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 
192.168.1.107:1068;branch=z9hG4bK-gxm8w1q7l2co;received=my.pub.lic.ip;rport=1068

From: ;tag=zdwiwg10qx
To: ;tag=as1cd819c5
Call-ID: 3c2679977b67-9j0euqvseh5v
CSeq: 1 INVITE
Server: mydomain
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="mydomain.be", 
nonce="323823f6"

Content-Length: 0


<>

<--- SIP read from TLS:my.pub.lic.ip:1068 --->
ACK sip:0123123...@ast.ser.ver.ip;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.107:1068;branch=z9hG4bK-gxm8w1q7l2co;rport
From: ;tag=zdwiwg10qx
To: ;tag=as1cd819c5
Call-ID: 3c2679977b67-9j0euqvseh5v
CSeq: 1 ACK
Max-Forwards: 70
Contact: ;reg-id=1
Content-Length: 0

<->



Any feedback is welcome.


Jonas



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