Re: [asterisk-users] Recommended sip providers

[Federico]

Please contact billing at chaneste dot com

Route is flat fee 0.0065 with Stir Shaken included.

Nine_ five_ 4  triple 4 se_ven four_ ze_ro _eight 

From: asterisk-users  On Behalf Of 
Tahir Almas Dhesi
Sent: Monday, November 20, 2023 6:14 AM
To: Commercial and Business-Oriented Asterisk Discussion 
; Asterisk Users Mailing List - Non-Commercial 
Discussion 
Subject: [asterisk-users] Recommended sip providers

 

Interested to know a good wholesale sip providers for 15k concurrent calls 

 

regards


Tahir Almas

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT

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Re: [asterisk-users] Recommended sip providers

[Tahir Almas Dhesi]

Only outbound to USA so no DID

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT



On Mon, Nov 20, 2023 at 4:18 PM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote:
>
> > Interested to know good wholesale SIP providers for 15k concurrent calls
>
> You might want to specify a bit more detail, such as:
>
>  - which country are you located in
>  - do you require inbound DDIs (if so, in which region/s)?
>  - which countries' Caller ID/s do you need to present?
>
> Antony.
>
> --
> These clients are often infected by viruses or other malware and need to
> be
> fixed.  If not, the user at that client needs to be fixed...
>
>  - Henrik Nordstrom, on Squid users' mailing list
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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Re: [asterisk-users] Recommended sip providers

[Antony Stone]

On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote:

> Interested to know good wholesale SIP providers for 15k concurrent calls

You might want to specify a bit more detail, such as:

 - which country are you located in
 - do you require inbound DDIs (if so, in which region/s)?
 - which countries' Caller ID/s do you need to present?

Antony.

-- 
These clients are often infected by viruses or other malware and need to be 
fixed.  If not, the user at that client needs to be fixed...

 - Henrik Nordstrom, on Squid users' mailing list

   Please reply to the list;
 please *don't* CC me.

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[asterisk-users] Recommended sip providers

[Tahir Almas Dhesi]

Interested to know a good wholesale sip providers for 15k concurrent calls

regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
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Re: [asterisk-users] Get SIP Call-ID from ARI

[Joshua C. Colp]

On Sat, Jun 17, 2023 at 8:41 PM TTT  wrote:

> I tried
>
> GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id)
>
>
>
> But it responds with
>
> "message": "Channel not in Stasis application"
>
>
>
> Since I want to get the call-id for a channel not in stasis I guess that
> won’t work.  Similarly, I can’t force the channel through my own code in
> the dialplan, so the PJSIP_HEADER function won’t work.  So it looks like
> I’ll have to upgrade my Asterisk test system to get the Call-ID from the
> ARI event.  It looks like it was added in Ast 16.
>
>
>
> Out of curiosity, I see that call-id is returned in the “protocol_id”
> field of channel data structure.  However, since all channels in the same
> call must have the same Call-ID, how can this data be associated with a
> channel?  Wouldn’t it have to be associated with a bridge?  The Call-ID
> should not be available until two legs are bridged (I think).
>

All channels in a call do not have the same Call-ID. Each channel has its
own SIP Call-ID (if it is a PJSIP channel) as they are individual call legs
and individual SIP dialogs.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
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Re: [asterisk-users] Get SIP Call-ID from ARI

[TTT]

I tried 

GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id)

 

But it responds with

"message": "Channel not in Stasis application"

 

Since I want to get the call-id for a channel not in stasis I guess that won’t 
work.  Similarly, I can’t force the channel through my own code in the 
dialplan, so the PJSIP_HEADER function won’t work.  So it looks like I’ll have 
to upgrade my Asterisk test system to get the Call-ID from the ARI event.  It 
looks like it was added in Ast 16.

 

Out of curiosity, I see that call-id is returned in the “protocol_id” field of 
channel data structure.  However, since all channels in the same call must have 
the same Call-ID, how can this data be associated with a channel?  Wouldn’t it 
have to be associated with a bridge?  The Call-ID should not be available until 
two legs are bridged (I think).

 

Brian

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Joshua C. Colp
Sent: Saturday, June 17, 2023 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Get SIP Call-ID from ARI

 

On Sat, Jun 17, 2023 at 2:55 PM TTT mailto:li...@telium.io> > 
wrote:

Based on postings it should be possible to get the SIP Call-ID header value 
from the ARI.  At what point is this value available ?  As well, how do I 
retrieve that value – something like

 

GET /channels/{channelId}/pjsip_header?key=Call-Id

 

But that doesn’t work.

 

'pjsip_header' is not a valid route. All possible routes are documented on the 
wiki, if it's not there then it doesn't exist.

 

Instead you would use variable[1] to execute the PJSIP_HEADER dialplan 
function[2] or a better way would be the CHANNEL dialplan function[3] such as:

 

GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id)

 

Though I haven't tested that.

 

Newer versions also include the protocol identifier (Call-ID) in the channel 
ARI structure[4] which would be in events, or explicitly retrieved[5].

 

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-getChannelVar

[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER

[3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL

[4] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Channel

[5] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-get

 

-- 

Joshua C. Colp

Asterisk Project Lead

Sangoma Technologies

Check us out at www.sangoma.com <http://www.sangoma.com>  and www.asterisk.org 
<http://www.asterisk.org> 

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Re: [asterisk-users] Get SIP Call-ID from ARI

[Joshua C. Colp]

On Sat, Jun 17, 2023 at 2:55 PM TTT  wrote:

> Based on postings it should be possible to get the SIP Call-ID header
> value from the ARI.  At what point is this value available ?  As well, how
> do I retrieve that value – something like
>
>
>
> GET /channels/{channelId}/pjsip_header?key=Call-Id
>
>
>
> But that doesn’t work.
>

'pjsip_header' is not a valid route. All possible routes are documented on
the wiki, if it's not there then it doesn't exist.

Instead you would use variable[1] to execute the PJSIP_HEADER dialplan
function[2] or a better way would be the CHANNEL dialplan function[3] such
as:

GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id)

Though I haven't tested that.

Newer versions also include the protocol identifier (Call-ID) in the
channel ARI structure[4] which would be in events, or explicitly
retrieved[5].

[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-getChannelVar
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER
[3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL
[4]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Channel
[5]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-get

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] Get SIP Call-ID from ARI

[TTT]

Based on postings it should be possible to get the SIP Call-ID header value
from the ARI.  At what point is this value available ?  As well, how do I
retrieve that value - something like

 

GET /channels/{channelId}/pjsip_header?key=Call-Id

 

But that doesn't work.

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[asterisk-users] Handling SIP refers when using a SIP Proxy

[Dovid Bender]

Hi,,

When using a SIP proxy to load balance calls how do you make it that a call
on an attended transfer reaches the same Asterisk box every time? I was
told that in later versions of Asterisk there is some "magic" to make it
work correctly when load balancing.

TIA.

Dovid
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Re: [asterisk-users] Two sip extensions

[Steve Edwards]

On Fri, 19 Jul 2019, Jerry Geis wrote:

I was not aware of the (+) format... basically "add" to the general 
section.


How far back does that go? T o 1.4.X ?


I don't know, but I checked a sip.conf from 1.2 (2012ish?) and I was using 
it then.



Is there a documentation piece on that ?


I'm sure there is, I just don't know where :)

Another cool configuration file feature is templates (an exclamation mark 
instead of a plus sign). It lets you define common 'snippets' once and 
include them in each context as needed.


Here's an example from that same (1.2 based) project:

; templates
[digit-timeout](!)
exten = t,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = t,n,goto(${CONTEXT},s,1)
[h](!)
exten = h,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = h,n,goto(settle-card,s,1)
[i](!)
exten = i,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = i,n,goto(${CONTEXT},s,1)
[s](!)
exten = s,1,verbose(1,[${EXTEN}@${CONTEXT}])
[max-timeout](!)
exten = T,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = T,n,goto(max-time,s,1)
[x](!)
exten = _x.,1,  verbose(1,[${EXTEN}@${CONTEXT}])

; authorized the card
[auth-card](h,i,s,max-timeout,digit-timeout)
exten = s,2,agi(write-cdr)
exten = s,n,set(PRODUCT=${CONTEXT})
exten = s,n,set(PER-MINUTE=0)
exten = s,n,set(PREAMBLE=${CUSTOMER}/menu/m1101)
exten = s,n,
agi(auth-card,${AUTH-FLAGS},${DEBUG-MODE},${VERBOSE-MODE})
exten = s,n,goto(theme,s,1)

The templates are inserted into the auth-card context when the file is 
parsed. I don't have a 1.2 host running anymore, but a 'show dialplan 
auth-card' (1.2) would look something like:


[auth-card](h,i,s,max-timeout,digit-timeout)
exten = T,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = T,n,goto(max-time,s,1)

exten = h,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = h,n,goto(settle-card,s,1)

exten = i,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = i,n,goto(${CONTEXT},s,1)

exten = s,1,verbose(1,[${EXTEN}@${CONTEXT}])

exten = s,2,agi(write-cdr)
exten = s,n,set(PRODUCT=${CONTEXT})
exten = s,n,set(PER-MINUTE=0)
exten = s,n,set(PREAMBLE=${CUSTOMER}/menu/m1101)
exten = s,n,
agi(auth-card,${AUTH-FLAGS},${DEBUG-MODE},${VERBOSE-MODE})
exten = s,n,goto(theme,s,1)

exten = t,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = t,n,goto(${CONTEXT},s,1)

Templates are also useful in other configuration files like sip.conf to 
define 'classes' of parameters like 'dial-in-agent' or 'supervisor' that 
can be included in endpoint definitions to reduce clutter, increase 
consistency, and reduce maintenance.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Two sip extensions

[Joshua C. Colp]

On Fri, Jul 19, 2019, at 11:58 AM, Jerry Geis wrote:
> >> 
> >> 
> ; 4450
> [general](+)
>   register= 4450 at 10.20.1.1 
> /4450
> [4450]
> 
> Thanks Steve,
> 
> I was not aware of the (+) format... basically "add" to the general section.
> 
> How far back does that go? T o 1.4.X ?
> Is there a documentation piece on that ?

There's a section on the wiki[1] which talks about the various aspects of the 
configuration parser and how to do things in the configuration files. Templates 
are also very handy[2].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Configuration+Files
[2] https://wiki.asterisk.org/wiki/display/AST/Using+Templates

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Two sip extensions

[Jerry Geis]

>
>
>>
>> ; 4450
>> [general](+)
>>  register= 4450 at 10.20.1.1 
>> /4450
>> [4450]
>>
>>
>>
Thanks Steve,

I was not aware of the (+) format... basically "add" to the general section.

How far back does that go? T o 1.4.X ?
Is there a documentation piece on that ?

Jerry
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Re: [asterisk-users] Two sip extensions

[Steve Edwards]

On Thu, 18 Jul 2019, Joshua C. Colp wrote:


On Thu, Jul 18, 2019, at 10:10 AM, Jerry Geis wrote:

I have two SIP extensions defined in sip.conf

register => 4450@10.20.1.1/4450
[4450]
type=friend
username=4450
host=10.20.1.1
allow=all
dtmfmode=inband
context=incoming

register => 4451@10.20.1.1/4451
[4451]
type=friend
username=4451
host=10.20.1.1
allow=all
dtmfmode=inband
context=incoming


"register" lines have to be under the general section. They can't be within a 
friend/peer/user.


I format my entries in sip.conf like below to keep everything related to 
the endpoint together.


; 4450
[general](+)
register= 4450@10.20.1.1/4450
[4450]
allow   = all
context = incoming
dtmfmode= inband
host= 10.20.1.1
type= friend
username= 4450

; 4451
[general](+)
register= 4451@10.20.1.1/4451
[4451]
allow   = all
context = incoming
dtmfmode= inband
host= 10.20.1.1
type= friend
username= 4451

I like to keep the parameters in each stanza sorted and 'tabbed out' to 
make it easier to compare stanzas and because I'm just that kind of guy :)


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Two sip extensions

[Jerry Geis]

It looks like moving both to the general section got it working.
Never new that was a requirement. :)

Thanks,

Jerry

>
>
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Re: [asterisk-users] Two sip extensions

[Joshua C. Colp]

On Thu, Jul 18, 2019, at 10:29 AM, Jerry Geis wrote:
> Thanks - Ok I switch to look like what you have. 
> 
> register => 4450 at 10.20.1.1 
> /4450
> register => 4451 at 10.20.1.1 
> /4451
> 
> [4450]
> type=friend
> username=4450
> host=10.20.1.1
> allow=all
> dtmfmode=inband
> context=incoming
> 
> [4451]
> type=friend
> username=4451
> host=10.20.1.1
> allow=all
> dtmfmode=inband
> context=incoming
> Same thing. I can call into 4450 just fine. But I cannot call into 
> 4451. Both are defined in extensions.conf Both show up in the sip show 
> peers.4450/4450                 10.20.1.1                               
>     Auto (No)  No             5060     Unmonitored                     
>              
> 4451/4451                 10.20.1.1                                   
> Auto (No)  No             5060     Unmonitored                     
> Thanks,
> Jerry

Sorry, have to also readjust to chan_sip thinking and wonkiness. The second may 
get matched depending on the INVITE. You'll need to provide more information as 
I mentioned so it can be seen what exactly is happening on the calls that could 
differentiate them.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Two sip extensions

[Joshua C. Colp]

On Thu, Jul 18, 2019, at 10:29 AM, Jerry Geis wrote:
> Thanks - Ok I switch to look like what you have. 
> 
> register => 4450 at 10.20.1.1 
> /4450
> register => 4451 at 10.20.1.1 
> /4451
> 
> [4450]
> type=friend
> username=4450
> host=10.20.1.1
> allow=all
> dtmfmode=inband
> context=incoming
> 
> [4451]
> type=friend
> username=4451
> host=10.20.1.1
> allow=all
> dtmfmode=inband
> context=incoming
> Same thing. I can call into 4450 just fine. But I cannot call into 
> 4451. Both are defined in extensions.conf Both show up in the sip show 
> peers.4450/4450                 10.20.1.1                               
>     Auto (No)  No             5060     Unmonitored                     
>              
> 4451/4451                 10.20.1.1                                   
> Auto (No)  No             5060     Unmonitored                     
> Thanks,
> Jerry

Only the first entry will be matched for incoming. As another individual has 
stated you'll have to also clarify what doesn't exactly work. If you enable SIP 
debug (sip set debug on) do you not see the INVITE coming in for example?

-- 
Joshua C. Colp
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Re: [asterisk-users] Two sip extensions

[Jerry Geis]

Thanks - Ok  I switch to look like what you have.

register => 4450 at 10.20.1.1
/4450
register => 4451 at 10.20.1.1
/4451

[4450]
type=friend
username=4450
host=10.20.1.1
allow=all
dtmfmode=inband
context=incoming

[4451]
type=friend
username=4451
host=10.20.1.1
allow=all
dtmfmode=inband
context=incoming


Same thing. I can call into 4450 just fine. But I cannot call into
4451. Both are defined in extensions.conf

Both show up in the sip show peers.

4450/4450 10.20.1.1
Auto (No)  No 5060 Unmonitored
4451/4451 10.20.1.1
Auto (No)  No 5060 Unmonitored

Thanks,


Jerry
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Re: [asterisk-users] Two sip extensions

[Joshua C. Colp]

On Thu, Jul 18, 2019, at 10:10 AM, Jerry Geis wrote:
> I have two SIP extensions defined in sip.conf
> 
> register => 4450@10.20.1.1/4450
> [4450]
> type=friend
> username=4450
> host=10.20.1.1
> allow=all
> dtmfmode=inband
> context=incoming 
> 
> register => 4451@10.20.1.1/4451
> [4451]
> type=friend
> username=4451
> host=10.20.1.1
> allow=all
> dtmfmode=inband
> context=incoming
> 
> Pretty straight forward... The first one works the second one does not. 
> Then if I switch them again the first one works the second one does not.
> 
> What is happening ? Thanks,

"register" lines have to be under the general section. They can't be within a 
friend/peer/user.

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Re: [asterisk-users] Two sip extensions

[Dovid Bender]

Jerry,

Can you define working? Are you not able to get calls? Make calls? Also the
register statements should both be on top under the general section for
instance:


register => 4450@10.20.1.1/4450
register => 4451@10.20.1.1/4451

[4450]
type=friend
username=4450
host=10.20.1.1
allow=all
dtmfmode=inband
context=incoming

[4451]
type=friend
username=4451
host=10.20.1.1
allow=all
dtmfmode=inband
context=incoming


On Thu, Jul 18, 2019 at 9:09 AM Jerry Geis  wrote:

> I have two SIP extensions defined in sip.conf
>
> register => 4450@10.20.1.1/4450
> [4450]
> type=friend
> username=4450
> host=10.20.1.1
> allow=all
> dtmfmode=inband
> context=incoming
>
> register => 4451@10.20.1.1/4451
> [4451]
> type=friend
> username=4451
> host=10.20.1.1
> allow=all
> dtmfmode=inband
> context=incoming
>
> Pretty straight forward... The first one works the second one does not.
> Then if I switch them again the first one works the second one does not.
>
> What is happening ? Thanks,
>
> Jerry
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[asterisk-users] Two sip extensions

[Jerry Geis]

I have two SIP extensions defined in sip.conf

register => 4450@10.20.1.1/4450
[4450]
type=friend
username=4450
host=10.20.1.1
allow=all
dtmfmode=inband
context=incoming

register => 4451@10.20.1.1/4451
[4451]
type=friend
username=4451
host=10.20.1.1
allow=all
dtmfmode=inband
context=incoming

Pretty straight forward... The first one works the second one does not.
Then if I switch them again the first one works the second one does not.

What is happening ? Thanks,

Jerry
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Re: [asterisk-users] Spectrum SIP trunks

[Doug Lytle]

>>> We've recently replaced an old Meridian phone system (Analog) with Asterisk 
>>> and signed up for Spectrum SIP trunks.

Should have included that we're running Asterisk 13, under chan_sip

Doug

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[asterisk-users] Spectrum SIP trunks

[Doug Lytle]

We've recently replaced an old Meridian phone system (Analog) with Asterisk and 
signed up for Spectrum SIP trunks.

The service gets installed on July 8th and I was hoping somebody that may have 
already gone through the process could give me some hints.  I've only ever 
dealt with PRIs or IAX2 trunks when it came to Asterisk and this will be my 
first SIP trunk.

They installed the Adtran fibre box yesterday.  (We are in Michigan)

Has anybody already setup a Spectrum SIP trunk?  If so, could you provide me 
some input?

Google provided the suggested setup:

;[spectrum]
;host=IP Address of Adtran
;type=peer
;disallow=all
;allow=ulaw
;allow=alaw
;context=spectrum
;trunk=yes
;insecure=port,invite
;qualify=500
;qualifysmoothing=yes
;jitterbuffer=yes
;forcejitterbuffer=yes
;maxjitterbuffer=300
;maxjitterinterps=100
;resyncthreshold=1500

All comments or suggestions are welcome,

Thanks!

Doug

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Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

[Pavel]

23.04.2019 0:27, Joshua C. Colp wrote:

On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote:




Tried already.

"line" is good, but not perfect.

Every time I restart asterisk, it will generate new random string for ";line=".

So, every time I restart asterisk, registrar (Server1) will save one
more contact in it's database.

Some will remove obsolete contacts, but some will not.

For example, FreePBX will not remove obsolete contacts, if max_contacts
specified (FreePBX will set rewrite_contact=no in this case).

So, after a number of Asterisk restarts, FreePBX will reject new
registrations, as max_contacts is reached.

It should specify remove_existing to remove old ones to make room for the new 
ones. That would be a FreePBX thing, though.


FreePBX is an example, where it can be a critical problem.

3cx will work, but if you will restart asterisk 10 times - you will see 
10 times more contacts in 3cx.


When you will make call from 3cx - it will make 10 calls (10 contacts), 
untill they will obsolete...




Unfortunately, "line" does not save random between restarts.

It's also unable to specify "random" value in pjsip.conf.


I'm thinking to patch res_pjsip_outbound_registration to add this feature.

Am I wrong and there is another way ?

I don't see any reason why this couldn't be an option.


For flexibility.

Not to register new fake contacts in peer PBX.


It's also a security hole, as anybody can generate INVITE with
";line=random" from any IP address !

You can use an ACL to limit the endpoint to certain source IP addresses.


5+ !

Thank you, ACL is a good idea !



res_pjsip_outbound_registration will only match "line", but will not
take care about source IP, ...



Is there any more clear way to identify incoming INVITE/OPTIONS packets ?

Not very familliar with SIP, not sure, how should it be done.

There is no real defined mechanism within SIP to do this. Phones employ 
different mechanisms to differentiate. Some may use a similar mechanism to the 
line option. Some run multiple SIP transports on different ports for each 
account so they can differentiate based on where it came in on. Some look at 
the request URI coming in. Some just don't care.

Sniffered some time ago how it's done in phonerlite, jitsi, linksys, ...

Some use different port, some use ";rinstance=", the same like ";line=" 
in asterisk.


Was not sure it's a right way to go.


I will probably extend "line" a bit to specify it's value in pjsip.conf .

It will be less than 10 lines of code.


Thank you very much !

Your help will simplify my life a lot :-)



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Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

[Joshua C. Colp]

On Mon, Apr 22, 2019, at 2:10 PM, Pavel wrote:



> Tried already.
> 
> "line" is good, but not perfect.
> 
> Every time I restart asterisk, it will generate new random string for 
> ";line=".
> 
> So, every time I restart asterisk, registrar (Server1) will save one 
> more contact in it's database.
> 
> Some will remove obsolete contacts, but some will not.
> 
> For example, FreePBX will not remove obsolete contacts, if max_contacts 
> specified (FreePBX will set rewrite_contact=no in this case).
> 
> So, after a number of Asterisk restarts, FreePBX will reject new 
> registrations, as max_contacts is reached.

It should specify remove_existing to remove old ones to make room for the new 
ones. That would be a FreePBX thing, though.
 
> Unfortunately, "line" does not save random between restarts.
> 
> It's also unable to specify "random" value in pjsip.conf.
> 
> 
> I'm thinking to patch res_pjsip_outbound_registration to add this feature.
> 
> Am I wrong and there is another way ?

I don't see any reason why this couldn't be an option.
 
> 
> It's also a security hole, as anybody can generate INVITE with 
> ";line=random" from any IP address !

You can use an ACL to limit the endpoint to certain source IP addresses.

> 
> res_pjsip_outbound_registration will only match "line", but will not 
> take care about source IP, ...
> 
> 
> 
> Is there any more clear way to identify incoming INVITE/OPTIONS packets ?
> 
> Not very familliar with SIP, not sure, how should it be done.

There is no real defined mechanism within SIP to do this. Phones employ 
different mechanisms to differentiate. Some may use a similar mechanism to the 
line option. Some run multiple SIP transports on different ports for each 
account so they can differentiate based on where it came in on. Some look at 
the request URI coming in. Some just don't care.

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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

[Pavel]

Hi,

Thank for your answer.

22.04.2019 23:47, Joshua C. Colp пишет:

On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:

Hi,

Got problems with incoming SIP calls.

Scenario:

Server1: 3cx or any other server

Server2: Asterisk 16.2.1 . PJPROJECT 2.8

Server2 registers on Server1 with SIP ID 1121.

Registration is OK.

Server2 outgoing calls are OK.

INVITE, unauthorized, INVITE with password, OK, RINGING,...

Troubles with incoming calls / incoming INVITE's .

I can not identify endpoint by IP, I have multiple registrations on the
same Server1.

As far as I understood, res_pjsip_endpoint_identifier_user match
endpoint by "From" header, so it will not match also.

match_headers also seems useless (not able to match "INVITE" string,
just headers like "TO:").

Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY,
... packets)

It should be a typical scenario, but it does not work...

Is there any way to make it working ?

Outbound registration provides the line option[1] which can be used to 
differentiate traffic in regards to different outbound registrations. It 
requires the remote server to adhere to the SIP RFC and report back some data 
we give in our Contact, so you have to test it and see if it works.

[1] 
https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/


Tried already.

"line" is good, but not perfect.

Every time I restart asterisk, it will generate new random string for 
";line=".


So, every time I restart asterisk, registrar (Server1) will save one 
more contact in it's database.


Some will remove obsolete contacts, but some will not.

For example, FreePBX will not remove obsolete contacts, if max_contacts 
specified (FreePBX will set rewrite_contact=no in this case).


So, after a number of Asterisk restarts, FreePBX will reject new 
registrations, as max_contacts is reached.


Unfortunately, "line" does not save random between restarts.

It's also unable to specify "random" value in pjsip.conf.


I'm thinking to patch res_pjsip_outbound_registration to add this feature.

Am I wrong and  there is another way ?

It's also a security hole, as anybody can generate INVITE with 
";line=random" from any IP address !


res_pjsip_outbound_registration will only match "line", but will not 
take care about source IP, ...



Is there any more clear way to identify incoming INVITE/OPTIONS packets ?

Not very familliar with SIP, not sure, how should it be done.

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Re: [asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

[Joshua C. Colp]

On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:
> Hi,
> 
> Got problems with incoming SIP calls.
> 
> Scenario:
> 
> Server1: 3cx or any other server
> 
> Server2: Asterisk 16.2.1 . PJPROJECT 2.8
> 
> Server2 registers on Server1 with SIP ID 1121.
> 
> Registration is OK.
> 
> Server2 outgoing calls are OK.
> 
> INVITE, unauthorized, INVITE with password, OK, RINGING,...
> 
> Troubles with incoming calls / incoming INVITE's .
> 
> I can not identify endpoint by IP, I have multiple registrations on the 
> same Server1.
> 
> As far as I understood, res_pjsip_endpoint_identifier_user match 
> endpoint by "From" header, so it will not match also.
> 
> match_headers also seems useless (not able to match "INVITE" string, 
> just headers like "TO:").
> 
> Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, 
> ... packets)
> 
> It should be a typical scenario, but it does not work...
> 
> Is there any way to make it working ?

Outbound registration provides the line option[1] which can be used to 
differentiate traffic in regards to different outbound registrations. It 
requires the remote server to adhere to the SIP RFC and report back some data 
we give in our Contact, so you have to test it and see if it works.

[1] 
https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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[asterisk-users] Incoming SIP call, outgoing SIP registration. PJSIP.

[Pavel]

Hi,

Got problems with incoming SIP calls.

Scenario:

Server1: 3cx or any other server

Server2: Asterisk 16.2.1 . PJPROJECT 2.8

Server2 registers on Server1 with SIP ID 1121.

Registration is OK.

Server2 outgoing calls are OK.

INVITE, unauthorized, INVITE with password, OK, RINGING,...

Troubles with incoming calls / incoming INVITE's .

I can not identify endpoint by IP, I have multiple registrations on the 
same Server1.


As far as I understood, res_pjsip_endpoint_identifier_user match 
endpoint by "From" header, so it will not match also.


match_headers also seems useless (not able to match "INVITE" string, 
just headers like "TO:").


Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, 
... packets)


It should be a typical scenario, but it does not work...

Is there any way to make it working ?


[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

[endpoint0](!)
type=endpoint
transport=0.0.0.0-udp
disallow=all
allow=alaw
allow=ulaw
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
t38_udptl_nat=no
dtmf_mode=auto
direct_media=yes
from_domain=172.16.25.23
timers_sess_expires=1800
tone_zone=ru
language=ru
rewrite_contact=yes
rtp_symmetric=yes
force_rport=yes

[registration0](!)
type=registration
transport=0.0.0.0-udp
retry_interval=60
max_retries=10
expiration=3600
auth_rejection_permanent=yes
server_uri=sip:172.16.25.23


[fxs17](endpoint0)
context=from-sip-fxs
aors=fxs17
outbound_auth=fxs17
from_user=1121
set_var=DAHDICHAN=17

[fxs17]
type=aor
qualify_frequency=60
contact=sip:1121@172.16.25.23

[fxs17]
type=auth
auth_type=userpass
password=11
username=1121

[fxs17](registration0)
outbound_auth=fxs17
client_uri=sip:1121@172.16.25.23
contact_user=fxs17
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[asterisk-users] Can SIP domain help to set multiple SIP trunks between two boxes ?

[Olivier]

Hello,

I've been asked if it is possible or not to set several (10 or so) SIP
trunks between two boxes, one beeing an Avaya IPBX, the other being an
Asterisk 13 or 16 box (with either chan_sip or pjsip).

The reason behind this question come from billing requirements.
I'm not convinced yet setting several trunks is the proper answer to
specific billing requirements but my above question remains.

Before officially answering the root question, I'm planning to set a
demonstration between two Asterisk boxes, leaving trials with an Avaya IPBX
for a later step.

>From previous trials with chan_sip years ago, the main issues was matching
incoming calls to appropriate trunk as calls may come from the same IP/port
combination.

Now, I read very few lines about SIP domains and wondered if this could be
a mean to set several trunks without touching IP/port settings.

What do you think of this ?
Is it possible for Asterisk to send outbound and receive inbound using SIP
domain syntax in SIP messages instead of IP values ?

Best regards
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Re: [asterisk-users] Configure SIP reply timeout (timerb in sip.conf)

[Markus]

Reply to self: Found the problem after reading this post:

http://lists.digium.com/pipermail/asterisk-dev/2010-March/042735.html

You need to set timert1 in the peer config to *something*, otherwise it 
will ignore the timerb setting. Bug? It now looks like this and works fine:


[peer01]
host=1.2.3.4
type=peer
context=nowhere
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
timert1=500
timerb=2000

timert1=500 is the default anyway, according to sip.conf comments...

Regards
Markus


Am 07.01.2019 um 17:23 schrieb Markus:

Dear list,

Asterisk 11.25.0 user here. I'm trying to set up failing over to a 
second SIP peer if the first SIP peer doesn't answer on our SIP INVITE 
within 2 seconds.


In sip.conf I set timerb=2000 for this peer, but it doesn't seem to have 
any effect. The timeout is 6.5 seconds instead, which is in line with 
this description from sip.conf:


"timerb: Call setup timer. If a provisional response is not received in 
this amount of time, the call will autocongest. Defaults to 64*timert1 
(Which is 100 ms = rougly 6.5 seconds)"


Maybe I cannot set timerb on a peer-basis? Here's my peer config:

[peer01]
host=1.2.3.4
type=peer
context=nowhere
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
timerb=2000

Thank you!
Markus




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[asterisk-users] Configure SIP reply timeout (timerb in sip.conf)

[Markus]

Dear list,

Asterisk 11.25.0 user here. I'm trying to set up failing over to a 
second SIP peer if the first SIP peer doesn't answer on our SIP INVITE 
within 2 seconds.


In sip.conf I set timerb=2000 for this peer, but it doesn't seem to have 
any effect. The timeout is 6.5 seconds instead, which is in line with 
this description from sip.conf:


"timerb: Call setup timer. If a provisional response is not received in 
this amount of time, the call will autocongest. Defaults to 64*timert1 
(Which is 100 ms = rougly 6.5 seconds)"


Maybe I cannot set timerb on a peer-basis? Here's my peer config:

[peer01]
host=1.2.3.4
type=peer
context=nowhere
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
timerb=2000

Thank you!
Markus

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Re: [asterisk-users] Capture SIP all the time

[Marcelo Terres]

You can use the voipmonitor sniffer.

www.voipmonitor.org.

Regards,

Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On Thu, 6 Dec 2018 at 00:13, Steve Edwards  wrote:
>
> On Wed, 5 Dec 2018, Saint Michael wrote:
>
> > Is there a way to configure the old SIP channel to stay in sip set debug
> > all the time, without human intervention and also at boot time, by
> > default?
>
> If your goal is capture all SIP traffic, there may be other tools better
> suited.
>
> For example, tcpdump, dumpcap, or pcapsipdump can capture SIP packets.
> pcapsipdump can even capture the RTP along with the SIP so you can listen
> to the call if that doesn't make your bosses and coworkers freak out.
>
> I like to capture all of the SIP traffic in a pool of files that
> expire after 30 days. Then, when somebody says 'hey, my call didn't
> connect yesterday' I have something to work with.
>
> sngrep is a great tool for searching for calls and displaying decoded
> dialogs.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>  https://www.linkedin.com/in/steve-edwards-4244281
>
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>
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Re: [asterisk-users] Capture SIP all the time

[Steve Edwards]

On Wed, 5 Dec 2018, Saint Michael wrote:

Is there a way to configure the old SIP channel to stay in sip set debug 
all the time, without human intervention and also at boot time, by 
default?


If your goal is capture all SIP traffic, there may be other tools better 
suited.


For example, tcpdump, dumpcap, or pcapsipdump can capture SIP packets. 
pcapsipdump can even capture the RTP along with the SIP so you can listen 
to the call if that doesn't make your bosses and coworkers freak out.


I like to capture all of the SIP traffic in a pool of files that
expire after 30 days. Then, when somebody says 'hey, my call didn't 
connect yesterday' I have something to work with.


sngrep is a great tool for searching for calls and displaying decoded 
dialogs.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Capture SIP all the time

[Social Boh]

sipdebug = yes

sip.conf

---
I'm SoCIaL, MayBe

El 05/12/2018 a las 17:11, Saint Michael escribió:
Is there a way to configure the old SIP channel to stay in sip set 
debug all the time, without human intervention and also at boot time, 
by default?






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[asterisk-users] Capture SIP all the time

[Saint Michael]

Is there a way to configure the old SIP channel to stay in sip set debug
all the time, without human intervention and also at boot time, by default?
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Re: [asterisk-users] Convert SIP to PJSIP

[sean darcy]

On 9/24/18 2:57 PM, John T. Bittner wrote:

Hello all,

I am having some trouble converting this setup from SIP to PJSIP. Any 
help is much appreciated.


I used the converter script and get most of it but don’t see a 
registration entry.


How do you convert this entry into PJSIP.

This working sip config.

register => 
17185553...@sip.ringcentral.com:ARi4uYb2Mz:332940285...@sip12.ringcentral.com:5090/17185553321 



[17185553321]

type = peer

host = sip.ringcentral.com

transport=udp

defaultuser=332940285773   ; Authentication username for outbound 
proxies


username = 332940285773

fromuser=17185553321   ; Many SIP providers require this

fromdomain=sip.ringcentral.com

secret = ARi4uYb2Mz

canreinvite = no

disallow = all

allow = ulaw

nat = yes

dtmfmode = auto

rfc2833compensate = yes

trustrpid = yes

usereqphone = yes  ; This provider requires ";user=phone" on URI

callcounter = yes  ; Enable call counter for parallel 
outbound calls


busylevel = 2  ; Signal busy at 2 or more calls (feel 
free to adjust)


outboundproxy=sip12.ringcentral.com:5090

This is what it was converted too: But nothing for the registration ?

[17185553321]

type = aor

contact = sip:332940285...@sip.ringcentral.com

[17185553321]

type = identify

endpoint = 17185553321

match = sip.ringcentral.com

[17185553321]

type = auth

username = 17185553321

password = ARi4uYb2Mz

[17185553321]

type = endpoint

dtmf_mode = none

disallow = all

allow = ulaw

rtp_symmetric = yes

rewrite_contact = yes

outbound_proxy = sip12.ringcentral.com:5090

direct_media = no

from_user = 17185553321

from_domain = sip.ringcentral.com

device_state_busy_at = 2

auth = 17185553321

outbound_auth = 17185553321

aors = 17185553321



I'd try the convert script again and make sure the input file is 
sip.conf. A lot of this pjsip config doesn't make sense.


And I hope these numbers and passwords are fake !



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Re: [asterisk-users] Convert SIP to PJSIP

[John T. Bittner]

Hello all,

I am having some trouble converting this setup from SIP to PJSIP. Any help is 
much appreciated.

I used the converter script and get most of it but don't see a registration 
entry.
How do you convert this entry into PJSIP.
This working sip config.

register => 
17185553...@sip.ringcentral.com:ARi4uYb2Mz:332940285...@sip12.ringcentral.com:5090/17185553321

[17185553321]
type = peer
host = sip.ringcentral.com
transport=udp
defaultuser=332940285773   ; Authentication username for outbound proxies
username = 332940285773
fromuser=17185553321   ; Many SIP providers require this
fromdomain=sip.ringcentral.com
secret = ARi4uYb2Mz
canreinvite = no
disallow = all
allow = ulaw
nat = yes
dtmfmode = auto
rfc2833compensate = yes
trustrpid = yes
usereqphone = yes  ; This provider requires ";user=phone" on URI
callcounter = yes  ; Enable call counter for parallel outbound calls
busylevel = 2  ; Signal busy at 2 or more calls (feel free to 
adjust)
outboundproxy=sip12.ringcentral.com:5090

This is what it was converted too: But nothing for the registration ?

[17185553321]
type = aor
contact = sip:332940285...@sip.ringcentral.com

[17185553321]
type = identify
endpoint = 17185553321
match = sip.ringcentral.com

[17185553321]
type = auth
username = 17185553321
password = ARi4uYb2Mz

[17185553321]
type = endpoint
dtmf_mode = none
disallow = all
allow = ulaw
rtp_symmetric = yes
rewrite_contact = yes
outbound_proxy = sip12.ringcentral.com:5090
direct_media = no
from_user = 17185553321
from_domain = sip.ringcentral.com
device_state_busy_at = 2
auth = 17185553321
outbound_auth = 17185553321
aors = 17185553321

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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Re: [asterisk-users] Convert SIP to PJSIP

[sean darcy]

On 9/24/18 5:04 PM, John T. Bittner wrote:

Hello all,

I am having some trouble getting this to work under pjsip. Any help is 
much appreciated.


I used the converter script and I see it register but can’t receive or 
send to ringcentral.


Anyone get this working with PJSIP?

Works with chan_sip…

This working sip config.

register => 
17185553...@sip.ringcentral.com:ARi4uYb2Mz:332940285...@sip12.ringcentral.com:5090/17185553321 



[17185553321]

type = peer

host = sip.ringcentral.com

transport=udp

defaultuser=332940285773   ; Authentication username for outbound 
proxies


username = 332940285773

fromuser=17185553321   ; Many SIP providers require this

fromdomain=sip.ringcentral.com

secret = ARi4uYb2Mz

canreinvite = no

disallow = all

allow = ulaw

nat = yes

dtmfmode = auto

rfc2833compensate = yes

trustrpid = yes

usereqphone = yes  ; This provider requires ";user=phone" on URI

callcounter = yes  ; Enable call counter for parallel 
outbound calls


busylevel = 2  ; Signal busy at 2 or more calls (feel 
free to adjust)


outboundproxy=sip12.ringcentral.com:5090



What's your pjsip config ?



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Re: [asterisk-users] Convert SIP to PJSIP

[John T. Bittner]

Hello all,

I am having some trouble getting this to work under pjsip. Any help is much 
appreciated.

I used the converter script and I see it register but can't receive or send to 
ringcentral.

Anyone get this working with PJSIP?

Works with chan_sip...

This working sip config.

register => 
17185553...@sip.ringcentral.com:ARi4uYb2Mz:332940285...@sip12.ringcentral.com:5090/17185553321

[17185553321]
type = peer
host = sip.ringcentral.com
transport=udp
defaultuser=332940285773   ; Authentication username for outbound proxies
username = 332940285773
fromuser=17185553321   ; Many SIP providers require this
fromdomain=sip.ringcentral.com
secret = ARi4uYb2Mz
canreinvite = no
disallow = all
allow = ulaw
nat = yes
dtmfmode = auto
rfc2833compensate = yes
trustrpid = yes
usereqphone = yes  ; This provider requires ";user=phone" on URI
callcounter = yes  ; Enable call counter for parallel outbound calls
busylevel = 2  ; Signal busy at 2 or more calls (feel free to 
adjust)
outboundproxy=sip12.ringcentral.com:5090

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

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Re: [asterisk-users] Decoding SIP register hack

[sean darcy]

On 05/17/2018 05:29 PM, sean darcy wrote:

On 05/17/2018 04:47 PM, Daniel Tryba wrote:

On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:

    WARNING.* .*: fail2ban=''

# Option:  ignoreregex
# Notes.:  regex to ignore. If this regex matches, the line is ignored.
# Values:  TEXT
#
ignoreregex =



Thanks. Very useful as a tutorial for fail2ban.

But I don't think it covers this SIP hack. This guy isn't trying to
register.


His filter doesn't only trigger on REGISTERs, see the last line of the
matches and the context for guests (which logs the pattern of the last
line of the filter on an INVITE).



I'm far from a regex expert, but I don't think that last line would 
capture anything in the invite. In fact, asterisk doesn't throw any 
WARNING at all for this INVITE.


I'm not sure, but I don't even see how you can get asterisk to log these 
invites at all. There's no heading such as WARNING( or NOTICE, SECURITY, 
etc).



  That why I find it puzzling. What is he trying to do ?


There are sip servers publicly reachable that will relay INVITEs, make
sure yours aren't. And there are only 2 kinds of operators of sip
server:
-those that have been the victim of toll fraud
-those that will be the victim of toll fraud

You can do nothing to stop this kind of traffic. The only thing you can
do is block it, either using only a whitelist (cumbersome) or generate a
blacklist with for example fail2ban or a more elaborate honeypot setup.
Or setup a proxy that will filter patterns you discover from

BTW this is not a person, this is an automated script, running most
likely on compromised machines and sending spoofed ips. These scripts
care about generating a ring on a phone (again most an abuseable/hacked
account (or purchased with CC fraud)). If they find a server that does,
it will be targetted for all kind of fraud.



Very interesting.

sen






I found these by staring at sip debug, and tying together the SIP 
retransmission id with the INVITE. That was an afternoon! Is there any 
way to automate this ? Specifically, find the INVITE that generates the 
retransmission ?


Otherwise, I can't see how anyone could block these attempts.

> There are sip servers publicly reachable that will relay INVITEs, make
> sure yours aren't.

How do I make sure my server won't relay INVITEs ?

sean


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Re: [asterisk-users] Decoding SIP register hack

[sean darcy]

On 05/17/2018 04:47 PM, Daniel Tryba wrote:

On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:

WARNING.* .*: fail2ban=''

# Option:  ignoreregex
# Notes.:  regex to ignore. If this regex matches, the line is ignored.
# Values:  TEXT
#
ignoreregex =



Thanks. Very useful as a tutorial for fail2ban.

But I don't think it covers this SIP hack. This guy isn't trying to
register.


His filter doesn't only trigger on REGISTERs, see the last line of the
matches and the context for guests (which logs the pattern of the last
line of the filter on an INVITE).



I'm far from a regex expert, but I don't think that last line would 
capture anything in the invite. In fact, asterisk doesn't throw any 
WARNING at all for this INVITE.


I'm not sure, but I don't even see how you can get asterisk to log these 
invites at all. There's no heading such as WARNING( or NOTICE, SECURITY, 
etc).



  That why I find it puzzling. What is he trying to do ?


There are sip servers publicly reachable that will relay INVITEs, make
sure yours aren't. And there are only 2 kinds of operators of sip
server:
-those that have been the victim of toll fraud
-those that will be the victim of toll fraud

You can do nothing to stop this kind of traffic. The only thing you can
do is block it, either using only a whitelist (cumbersome) or generate a
blacklist with for example fail2ban or a more elaborate honeypot setup.
Or setup a proxy that will filter patterns you discover from

BTW this is not a person, this is an automated script, running most
likely on compromised machines and sending spoofed ips. These scripts
care about generating a ring on a phone (again most an abuseable/hacked
account (or purchased with CC fraud)). If they find a server that does,
it will be targetted for all kind of fraud.



Very interesting.

sen



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Re: [asterisk-users] Decoding SIP register hack

[Steve Edwards]

On Thu, 17 May 2018, Daniel Tryba wrote:

You can do nothing to stop this kind of traffic. The only thing you can 
do is block it, either using only a whitelist (cumbersome) or generate a 
blacklist with for example fail2ban or a more elaborate honeypot setup. 
Or setup a proxy that will filter patterns you discover from


Keep in mind that since this is UDP, source addresses can be spoofed so 
any automated solution will need a whitelist so you don't get tricked into 
blocking legitimate traffic.


And since you 'need a whitelist' why not just use that and block 
everything else?


A clever solution to a mobile user base is to use knockd to allow remote 
access.


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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Decoding SIP register hack

[Daniel Tryba]

On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:
> > WARNING.* .*: fail2ban=''
> >
> ># Option:  ignoreregex
> ># Notes.:  regex to ignore. If this regex matches, the line is ignored.
> ># Values:  TEXT
> >#
> >ignoreregex =
> >
> >
> Thanks. Very useful as a tutorial for fail2ban.
> 
> But I don't think it covers this SIP hack. This guy isn't trying to
> register.

His filter doesn't only trigger on REGISTERs, see the last line of the
matches and the context for guests (which logs the pattern of the last
line of the filter on an INVITE).

>  That why I find it puzzling. What is he trying to do ?

There are sip servers publicly reachable that will relay INVITEs, make
sure yours aren't. And there are only 2 kinds of operators of sip
server:
-those that have been the victim of toll fraud
-those that will be the victim of toll fraud

You can do nothing to stop this kind of traffic. The only thing you can
do is block it, either using only a whitelist (cumbersome) or generate a
blacklist with for example fail2ban or a more elaborate honeypot setup.
Or setup a proxy that will filter patterns you discover from 

BTW this is not a person, this is an automated script, running most
likely on compromised machines and sending spoofed ips. These scripts
care about generating a ring on a phone (again most an abuseable/hacked
account (or purchased with CC fraud)). If they find a server that does,
it will be targetted for all kind of fraud.

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Re: [asterisk-users] Decoding SIP register hack

[sean darcy]

On 05/17/2018 11:38 AM, Frank Vanoni wrote:

On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote:


3. How do I set up the server to block these ?

4. Can I stop the retransmitting of the 401 Unauthorized packets ?


I'm happy with Fail2Ban protecting my Asterisk 13. Here is my
configuration:

in /etc/asterisk/logger.conf:

messages => security,notice,warning,error


in /etc/asterisk/sip.conf:

allowguest=yes
context=unauthenticated


in /etc/asterisk/extensions.conf:

[unauthenticated]
;; Incomming calls from unauthenticated caller -> Fail2Ban
exten => _X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _X.,3,HangUp()

exten => _+X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _+X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _+X.,3,HangUp()



in /etc/fail2ban/jail.conf:

[asterisk]
filter   = asterisk
action = iptables-allports[name=ASTERISK]
logpath  = /var/log/asterisk/messages
maxretry = 1
findtime = 86400
bantime  = 518400
enabled = true


in /etc/fail2ban/filter.d

# Fail2Ban configuration file
#
#
# $Revision: 250 $
#

[INCLUDES]

# Read common prefixes. If any customizations available -- read them
from
# common.local
#before = common.conf


[Definition]

#_daemon = asterisk

# Option:  failregex
# Notes.:  regex to match the password failures messages in the
logfile. The
#  host must be matched by a group named "host". The tag
"" can
#  be used for standard IP/hostname matching and is only an
alias for
#  (?:::f{4,6}:)?(?P\S+)
# Values:  TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for
':.*' - Wrong password
NOTICE.* .*: Call from '.*' \((:[0-9]{1,5})?\) to
extension '.*' rejected because extension not found in context
'unauthenticated'
NOTICE.* chan_sip.c: Call from '.*' \((:[0-
9]{1,5})?\) to extension '.*' rejected because extension not found in
context 'unauthenticated'
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Username/auth name mismatch
    NOTICE.* .*: Registration from '.*' failed for
':.*' - No matching peer found
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Not a local domain
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Peer is not supposed to register
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Device does not match ACL
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Device not configured to use this transport type
    NOTICE.* .*: No registration for peer '.*' \(from
\)
    NOTICE.* .*: Host  failed MD5 authentication for
'.*' \(.*\)
    NOTICE.* .*: Host  denied access to register peer
'.*'
    NOTICE.* .*: Host  did not provide proper
plaintext password for '.*'
    NOTICE.* .*: Registration of '.*' rejected: '.*' from:
''
    NOTICE.* .*: Peer '.*' is not dynamic (from )
    NOTICE.* .*: Host  denied access to register peer
'.*'
    SECURITY.* .*:
SecurityEvent="InvalidAccountID".*,Severity="Error",Service="SIP".*,Rem
oteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
    SECURITY.* .*:
SecurityEvent="FailedACL".*,Severity="Error",Service="SIP".*,RemoteAddr
ess="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
    SECURITY.* .*:
SecurityEvent="InvalidPassword".*,Severity="Error",Service="SIP".*,Remo
teAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
    SECURITY.* .*:
SecurityEvent="ChallengeResponseFailed".*,Severity="Error",Service="SIP
".*,RemoteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-
noservice' \(language '.*'\)
SECURITY.* .*:
SecurityEvent="ChallengeSent".*,Severity="Informational",Service="SIP".
*,AccountID="sip:.*@93.94.247.123".*,RemoteAddress="IPV[46]/(UDP|TCP|TL
S)//[0-9]+
WARNING.* .*: fail2ban=''

# Option:  ignoreregex
# Notes.:  regex to ignore. If this regex matches, the line is ignored.
# Values:  TEXT
#
ignoreregex =



Thanks. Very useful as a tutorial for fail2ban.

But I don't think it covers this SIP hack. This guy isn't trying to 
register. That why I find it puzzling. What is he trying to do ?


sean


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Re: [asterisk-users] Decoding SIP register hack

[Frank Vanoni]

On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote:

> 3. How do I set up the server to block these ?
> 
> 4. Can I stop the retransmitting of the 401 Unauthorized packets ?

I'm happy with Fail2Ban protecting my Asterisk 13. Here is my
configuration:

in /etc/asterisk/logger.conf:

messages => security,notice,warning,error


in /etc/asterisk/sip.conf:

allowguest=yes
context=unauthenticated


in /etc/asterisk/extensions.conf:

[unauthenticated]
;; Incomming calls from unauthenticated caller -> Fail2Ban
exten => _X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}') 
exten => _X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _X.,3,HangUp()

exten => _+X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}') 
exten => _+X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _+X.,3,HangUp()



in /etc/fail2ban/jail.conf:

[asterisk]
filter   = asterisk
action = iptables-allports[name=ASTERISK]
logpath  = /var/log/asterisk/messages
maxretry = 1
findtime = 86400
bantime  = 518400
enabled = true


in /etc/fail2ban/filter.d

# Fail2Ban configuration file
#
#
# $Revision: 250 $
#

[INCLUDES]

# Read common prefixes. If any customizations available -- read them
from
# common.local
#before = common.conf


[Definition]

#_daemon = asterisk

# Option:  failregex
# Notes.:  regex to match the password failures messages in the
logfile. The
#  host must be matched by a group named "host". The tag
"" can
#  be used for standard IP/hostname matching and is only an
alias for
#  (?:::f{4,6}:)?(?P\S+)
# Values:  TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for
':.*' - Wrong password
NOTICE.* .*: Call from '.*' \((:[0-9]{1,5})?\) to
extension '.*' rejected because extension not found in context
'unauthenticated'
NOTICE.* chan_sip.c: Call from '.*' \((:[0-
9]{1,5})?\) to extension '.*' rejected because extension not found in
context 'unauthenticated'
NOTICE.* .*: Registration from '.*' failed for
':.*' - Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for
':.*' - No matching peer found
NOTICE.* .*: Registration from '.*' failed for
':.*' - Not a local domain
NOTICE.* .*: Registration from '.*' failed for
':.*' - Peer is not supposed to register
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device does not match ACL
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device not configured to use this transport type
NOTICE.* .*: No registration for peer '.*' \(from
\)
NOTICE.* .*: Host  failed MD5 authentication for
'.*' \(.*\)
NOTICE.* .*: Host  denied access to register peer
'.*'
NOTICE.* .*: Host  did not provide proper
plaintext password for '.*'
NOTICE.* .*: Registration of '.*' rejected: '.*' from:
''
NOTICE.* .*: Peer '.*' is not dynamic (from )
NOTICE.* .*: Host  denied access to register peer
'.*'
SECURITY.* .*:
SecurityEvent="InvalidAccountID".*,Severity="Error",Service="SIP".*,Rem
oteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="FailedACL".*,Severity="Error",Service="SIP".*,RemoteAddr
ess="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="InvalidPassword".*,Severity="Error",Service="SIP".*,Remo
teAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="ChallengeResponseFailed".*,Severity="Error",Service="SIP
".*,RemoteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-
noservice' \(language '.*'\)
SECURITY.* .*:
SecurityEvent="ChallengeSent".*,Severity="Informational",Service="SIP".
*,AccountID="sip:.*@93.94.247.123".*,RemoteAddress="IPV[46]/(UDP|TCP|TL
S)//[0-9]+
WARNING.* .*: fail2ban=''

# Option:  ignoreregex
# Notes.:  regex to ignore. If this regex matches, the line is ignored.
# Values:  TEXT
#
ignoreregex =


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[asterisk-users] Decoding SIP register hack

[sean darcy]

I need some help understanding SIP dialog. Some actor is trying to 
access my server, but I can't figure out what he's trying to do ,or how.


I'm getting a lot of these warnings.

[May 17 10:08:08] WARNING[1532]: chan_sip.c:4068 retrans_pkt: 
Retransmission timeout reached on transmission 
_zIr9tDtBxeTVTY5F7z8kD7R.. for seqno 101


With SIP DEBUG I tracked the Call-ID to this INVITE :

<--- SIP read from UDP:192.111.139.146:29281 --->
INVITE sip:+48223079992@67.80.191.250:5060 SIP/2.0
Via: SIP/2.0/UDP 
100.149.241.68:5060;branch=z4hG4bK-966187-1---q9ft4HdLB4ZeBqs;rport=5060
Contact: 
;+sip.instance=""

Max-Forwards: 70
To: 
From: "Caller";tag=sXPNixD5Ui42V
Call-ID: _zIr9tDtBxeTVTY5F7z8kD7R..
CSeq: 101 INVITE
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
REGISTER, SUBSCRIBE, INFO

Supported: replaces
User-Agent: GSM
Allow-Events: hold, talk, conference
Accept: application/sdp
Content-Length: 771

v=0
o=CiscoSystemsSIP-IPPhone 18338 11953 IN IP4 100.149.241.68
s=SIP Call
c=IN IP4 100.149.241.68
t=0 0
m=audio 2 RTP/AVP 0 8 18 101
a=rtpmap:3 gsm/8000
a=rtpmap:96 speex/8000
a=rtpmap:97 speex/8000
a=fmtp:97 mode=2
a=rtpmap:98 speex/8000
a=fmtp:98 mode=5
a=rtpmap:99 speex/8000
a=fmtp:99 mode=7
a=rtpmap:107 speex/32000
a=fmtp:107 mode=10
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:108 ilbc/8000
a=rtpmap:113 g7231/8000
a=rtpmap:18 g729/8000
a=rtpmap:100 G726-16/8000
a=rtpmap:101 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:103 G726-40/8000
a=rtpmap:4 g723/8000
a=fmtp:18 annexb=no
a=rtpmap:109 ilbc/8000
a=fmtp:109 mode=20
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
a=ptime:20
a=sendrecv
<->
--- (15 headers 34 lines) ---
Sending to 192.111.139.146:29281 (NAT)
Sending to 192.111.139.146:29281 (NAT)
Using INVITE request as basis request - _zIr9tDtBxeTVTY5F7z8kD7R..
No matching peer for '9353' from '192.111.139.146:29281'
..
Which then generates a lot of transmissions showing Unauthorized:
..
Retransmitting #10 (NAT) to 192.111.139.146:29281:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
100.149.241.68:5060;branch=z4hG4bK-966187-1---q9ft4HdLB4ZeBqs;received=192.111.139.146;rport=29281

From: "Caller";tag=sXPNixD5Ui42V
To: ;tag=as1f60e6dd
Call-ID: _zIr9tDtBxeTVTY5F7z8kD7R..
CSeq: 101 INVITE
Server: Asterisk PBX 13.21.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk_home", 
nonce="0794806c"

Content-Length: 0


1. What's this guy trying to do ? It looks like he's trying to generate 
a call from the server to a Polish number. Why bother ?


2. What's the role of the Via and the Contact line ?  The 100.149.241.68 
seems to be a cell phone. 100.128.0.0/9 is T-mobile.


3. How do I set up the server to block these ?

4. Can I stop the retransmitting of the 401 Unauthorized packets ?

Any help appreciated.

sean


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Re: [asterisk-users] 1. SIP trunks going to the wrong context (Ade Vickers)

[Mc GRATH Ricardo]

How about if you set;
 
 exten => _se,1,Dial(IAX2/cloud/1000,30,r)


Mc GRATH Ricardo
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[asterisk-users] Broken? SIP/devicename/extension/IPorHost

[Saint Michael]

I tried many times and this dial model fails for me
SIP/g729-outbound/155/192.168.1.120
The peer g729-outbound does exist but it does not have a host line,
that is why I am supplying the host dynamically for each call.
According Asterisk13 file configs/samples/sip.conf.sample, this is legal.
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Re: [asterisk-users] German sip dial rules

[Binarus]

On 12.06.2017 17:00, Hans-Peter Jansen wrote:
> 
>  * zero prefix for outside calls
>  * zero zero or plus prefix for international calls
>  * handle emergency calls
> 
> With ISDN, one was able to just forward the called number, but with sip, one 
> has to normalize the dialed pattern in order to match SIP (provider) 
> expectations... As always, the devil is in the details.
> 

I am not sure if I have understood what your problem actually is. What I
have done:

1) Download the number plan for Germany from the Bundesnetzagentur
(abbreviated BA in the following text - I don't know the correct English
expression) and work through it carefully to get a feeling where
possible problems are.

In the dial plan:

2) Check if the dial string is exactly one of the well-defined
*internal* phone numbers of other users. If yes, make the connection.

3) Using Asterisk's text / regex functions, check if there are forbidden
chars in the dial string (i.e. any char except 0-9, +, /, -). If yes,
throw an error / deny the connection.

4) Using Asterisk's text replacement / regex functions, do further
checks (e.g. + (if existent) must be the first char of the dial string),
and normalize the dial string (e.g. convert + to 00, throw away / and -).

Now it gets complicated. I have carefully worked through the BA plan to
determine the appropriate regular expressions for the following steps.

5) Determine if the dial string is an emergency number. If yes, make the
connection. For security reasons, I am doing this before the following
checks; these are quite complicated any might contain errors, so I am
checking for the emergency case as early as possible in the dial plan.

6) Determine if the called number is outside Germany (yes, possibly I
had to use the BA plan for that seemingly easy step, but I can't
remember for sure). If yes, make the connection (no further checks then
because I don't have the time to study the communication authorities'
number plans of all the other countries).

7) If we are still here, the called number is in Germany. Again use the
BA number plan to determine what is being called. It is up to you to
decide what happens if somebody (possibly accidentally) calls something
like "Zeitansage" or "Sozialdienste".

I have one rule / filter for every item from the list in the BA plan.
This makes quite a bunch of rules, but this setup is working since
nearly three years now without any issues.

Regards,

Binarus




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Re: [asterisk-users] German sip dial rules

[Daniel Tryba]

On Mon, Jun 12, 2017 at 05:00:31PM +0200, Hans-Peter Jansen wrote:
> is somebody attending, that wants to share his outgoing dial rules of 
> extension.conf, like used in typical(?) german pbx setups?
> 
>  * zero prefix for outside calls
>  * zero zero or plus prefix for international calls
>  * handle emergency calls
> 
> With ISDN, one was able to just forward the called number, but with sip, one 
> has to normalize the dialed pattern in order to match SIP (provider) 
> expectations... As always, the devil is in the details.

Shouldn't you just ask the provider?

But not being German, the only problem I know of is the ISDN 
sub addressing feature widely in use.

Looking at https://en.wikipedia.org/wiki/Telephone_numbers_in_Germany
I'd guess a dialplan would be (assuming the operator wants e164+):

exten => _1.,1,Dial(SIP/${EXTEN}@provider)
exten => _[2-9].,1,Goto(+49xyz${EXTEN:1},1)
exten => _0[1-9].,1,Goto(+49${EXTEN:1},1)
exten => _00[1-9].,1,Goto(+${EXTEN:2},1)

exten => _+.,1,Dial(SIP/${EXTEN}@provider)


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[asterisk-users] German sip dial rules

[Hans-Peter Jansen]

Hi,

is somebody attending, that wants to share his outgoing dial rules of 
extension.conf, like used in typical(?) german pbx setups?

 * zero prefix for outside calls
 * zero zero or plus prefix for international calls
 * handle emergency calls

With ISDN, one was able to just forward the called number, but with sip, one 
has to normalize the dialed pattern in order to match SIP (provider) 
expectations... As always, the devil is in the details.

Thanks,
Pete

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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

[Jonas Kellens]

Hello


function sip_header is read-only.




Kind regards.

J.



On 14-04-17 11:28, registrator wrote:

In this case you will help function SIP_HEADER(from)


Sent from: Lenovo P70-A

On Apr 14, 2017 12:04 PM, Jonas Kellens  wrote:

Hello


this does not set user field in From-header.

I get :

From: "user762" ;tag=as7f44c043

What I want is :

From: "9876543210" ;tag=as7f44c043


I need this part : 

you see the user part ? I need to set the value 'user762'




Kind regards

J.




On 14-04-17 10:46, registrator wrote:

Hello!



May be you help CALLERID(name) function?



exten => _X.,1,Set(CALLERID(name)=$name)



Then you well see INVITE

SIP : FROM "$name" .



Sent from: Lenovo P70-A



On Apr 14, 2017 10:54 AM, Jonas Kellens  wrote:


Hello





any input on this ? How to set user-field in From-header with the 
Dial()-command in dialplan ?







Kind regards



J.





On 03-04-17 10:25, Jonas Kellens wrote:


Hello



how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?



None of the below Dial() command give the correct result :



exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})



The From part of the SIP INVITE always has the EXTEN in it in stead of the user 
(user762) :



From: "the_extension" ;tag=as224453ac



How can I get :



From: "the_extension" ;tag=as224453ac



??







I know about sip.conf. That is not the question. My question is clear : how to 
set this in dialplan ?







Thank you for the feedback.





Kind regards.







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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

[registrator]

In this case you will help function SIP_HEADER(from)


Sent from: Lenovo P70-A

On Apr 14, 2017 12:04 PM, Jonas Kellens  wrote:
>
> Hello
>
>
> this does not set user field in From-header.
>
> I get :
>
> From: "user762" ;tag=as7f44c043
>
> What I want is :
>
> From: "9876543210" ;tag=as7f44c043
>
>
> I need this part : 
>
> you see the user part ? I need to set the value 'user762'
>
>
>
>
> Kind regards
>
> J.
>
>
>
>
> On 14-04-17 10:46, registrator wrote:
>>
>> Hello!
>>
>>
>>
>> May be you help CALLERID(name) function?
>>
>>
>>
>> exten => _X.,1,Set(CALLERID(name)=$name)
>>
>>
>>
>> Then you well see INVITE 
>>
>> SIP : FROM "$name" .
>>
>>
>>
>> Sent from: Lenovo P70-A
>>
>>
>>
>> On Apr 14, 2017 10:54 AM, Jonas Kellens  wrote:
>>
>>> Hello
>>>
>>>
>>>
>>>
>>>
>>> any input on this ? How to set user-field in From-header with the 
>>> Dial()-command in dialplan ?
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Kind regards
>>>
>>>
>>>
>>> J.
>>>
>>>
>>>
>>>
>>>
>>> On 03-04-17 10:25, Jonas Kellens wrote:
>>>
 Hello



 how can I set the fromuser field of the SIP INVITE when using the 
 Dial()-command in the dialplan ?



 None of the below Dial() command give the correct result :



 exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)

 exten => 
 _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})

 exten => 
 _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})

 exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})



 The From part of the SIP INVITE always has the EXTEN in it in stead of the 
 user (user762) :



 From: "the_extension" ;tag=as224453ac



 How can I get :



 From: "the_extension" ;tag=as224453ac



 ??







 I know about sip.conf. That is not the question. My question is clear : 
 how to set this in dialplan ?







 Thank you for the feedback.





 Kind regards.





>
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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

[Jonas Kellens]

Hello


this does not set user field in From-header.

I get :

From: "user762" ;tag=as7f44c043

What I want is :

From: "9876543210" ;tag=as7f44c043


I need this part : 

you see the user part ? I need to set the value 'user762'




Kind regards

J.




On 14-04-17 10:46, registrator wrote:

Hello!

May be you help CALLERID(name) function?

exten => _X.,1,Set(CALLERID(name)=$name)

Then you well see INVITE
SIP : FROM "$name" .

Sent from: Lenovo P70-A

On Apr 14, 2017 10:54 AM, Jonas Kellens  wrote:

Hello


any input on this ? How to set user-field in From-header with the 
Dial()-command in dialplan ?



Kind regards

J.


On 03-04-17 10:25, Jonas Kellens wrote:

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?

None of the below Dial() command give the correct result :

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})

The From part of the SIP INVITE always has the EXTEN in it in stead of the user 
(user762) :

From: "the_extension";tag=as224453ac

How can I get :

From: "the_extension";tag=as224453ac

??



I know about sip.conf. That is not the question. My question is clear : how to 
set this in dialplan ?



Thank you for the feedback.


Kind regards.




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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

[registrator]

Hello!

May be you help CALLERID(name) function?

exten => _X.,1,Set(CALLERID(name)=$name)

Then you well see INVITE 
SIP : FROM "$name" .

Sent from: Lenovo P70-A

On Apr 14, 2017 10:54 AM, Jonas Kellens  wrote:
>
> Hello
>
>
> any input on this ? How to set user-field in From-header with the 
> Dial()-command in dialplan ?
>
>
>
> Kind regards
>
> J.
>
>
> On 03-04-17 10:25, Jonas Kellens wrote:
>>
>> Hello
>>
>> how can I set the fromuser field of the SIP INVITE when using the 
>> Dial()-command in the dialplan ?
>>
>> None of the below Dial() command give the correct result :
>>
>> exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
>> exten => 
>> _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
>> exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})
>> exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})
>>
>> The From part of the SIP INVITE always has the EXTEN in it in stead of the 
>> user (user762) :
>>
>> From: "the_extension" ;tag=as224453ac
>>
>> How can I get :
>>
>> From: "the_extension" ;tag=as224453ac
>>
>> ??
>>
>>
>>
>> I know about sip.conf. That is not the question. My question is clear : how 
>> to set this in dialplan ?
>>
>>
>>
>> Thank you for the feedback.
>>
>>
>> Kind regards.
>>
>>
>
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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

[Jonas Kellens]

Hello


any input on this ? How to set user-field in From-header with the 
Dial()-command in dialplan ?




Kind regards

J.


On 03-04-17 10:25, Jonas Kellens wrote:

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?


None of the below Dial() command give the correct result :

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten => 
_XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => 
_XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})

The From part of the SIP INVITE always has the EXTEN in it in stead of 
the user (user762) :


From: "the_extension" ;tag=as224453ac

How can I get :

From: "the_extension" ;tag=as224453ac

??



I know about sip.conf. That is not the question. My question is clear 
: how to set this in dialplan ?




Thank you for the feedback.


Kind regards.




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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

[Jonas Kellens]

Hello


in what way does this set the 'fromuser' field in the SIP INVITE ?



Kind regards.


J.

On 05-04-17 22:05, Pete Mundy wrote:

Hi Jonas

Does the information at this link help?

http://the-asterisk-book.com/1.6/funktionen-callerid.html

Pete


On 5/04/2017, at 8:11 pm, Jonas Kellens > wrote:


Hello

anyone have some useful input on this ?



Thanks.


On 03-04-17 10:25, Jonas Kellens wrote:

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?





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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

[Pete Mundy]

Hi Jonas 

Does the information at this link help? 

http://the-asterisk-book.com/1.6/funktionen-callerid.html

Pete


> On 5/04/2017, at 8:11 pm, Jonas Kellens  wrote:
> 
> Hello
> 
> anyone have some useful input on this ?
> 
> 
> 
> Thanks.
> 
> 
> On 03-04-17 10:25, Jonas Kellens wrote:
>> Hello
>> 
>> how can I set the fromuser field of the SIP INVITE when using the 
>> Dial()-command in the dialplan ?


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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

[Jonas Kellens]

Hello

anyone have some useful input on this ?



Thanks.


On 03-04-17 10:25, Jonas Kellens wrote:

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?


None of the below Dial() command give the correct result :

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten => 
_XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => 
_XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})

The From part of the SIP INVITE always has the EXTEN in it in stead of 
the user (user762) :


From: "the_extension" ;tag=as224453ac

How can I get :

From: "the_extension" ;tag=as224453ac

??



I know about sip.conf. That is not the question. My question is clear 
: how to set this in dialplan ?




Thank you for the feedback.


Kind regards.




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[asterisk-users] Define SIP fromuser field in Dial()-command

[Jonas Kellens]

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?


None of the below Dial() command give the correct result :

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten => 
_XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => 
_XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})

The From part of the SIP INVITE always has the EXTEN in it in stead of 
the user (user762) :


From: "the_extension" ;tag=as224453ac

How can I get :

From: "the_extension" ;tag=as224453ac

??



I know about sip.conf. That is not the question. My question is clear : 
how to set this in dialplan ?




Thank you for the feedback.


Kind regards.
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Re: [asterisk-users] Some SIP and IAX Asterisk unreachable after server restart

[Gokan Atmaca]

>
> Everything is working again if we restart the customers Asterisk.

Have you looked at the error logs?

On Mon, Feb 27, 2017 at 3:03 PM, Administrator TOOTAI  wrote:
> Hi all,
>
> we have a running Asterisk 11.25.1 in a VM (qemu/kvm) OS being Debian 7.11
> (wheezy), the host OS being the same.
>
> Problem: when we restart the server (eg host + VM), all customers Asterisk
> connecting without a VPN (doesn't matter which Asterisk version) are no more
> reachable. Same for IAX users.
>
> On the host side, after restart, SIP packet are entering the host but NOT to
> the VM (tshark debug). In IAX, POKE packets sended by the VM are reaching
> the client Asterisk but no packet answer.
>
> Everything is working again if we restart the customers Asterisk.
>
> Any clue ?
>
> Regards
>
> --
> Daniel
>
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[asterisk-users] Some SIP and IAX Asterisk unreachable after server restart

[Administrator TOOTAI]

Hi all,

we have a running Asterisk 11.25.1 in a VM (qemu/kvm) OS being Debian 
7.11 (wheezy), the host OS being the same.


Problem: when we restart the server (eg host + VM), all customers 
Asterisk connecting without a VPN (doesn't matter which Asterisk 
version) are no more reachable. Same for IAX users.


On the host side, after restart, SIP packet are entering the host but 
NOT to the VM (tshark debug). In IAX, POKE packets sended by the VM are 
reaching the client Asterisk but no packet answer.


Everything is working again if we restart the customers Asterisk.

Any clue ?

Regards

--
Daniel

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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Am 16.02.2017 um 15:01 schrieb Joshua Colp:

> As for your issues please do file them. I'd also suggest using bundled
> PJSIP, it works the best with Asterisk and we backport applicable fixes
> and include fixes we've created that have not yet made it to a PJSIP
> release.

OK, I'll try again with the bundled version.
If the bugs persist, I'll file some bugs ;-)


Max



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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Joshua Colp]

On Thu, Feb 16, 2017, at 09:51 AM, Max Grobecker wrote:
> Hi,
> 
> Am 16.02.2017 um 14:19 schrieb Annus Fictus:
> > And Microsip using PJSIP SIP stack :)
> 
> Sorry (also, for off-topic), based on my latest experience with PJSIP,
> I'm not sure if this really is a sign of good quality.
> Maybe it's a problem with the implementation in Asterisk (I haven't tried
> PJSIP in other software), but after just five minutes of testing
> I found several bugs regarding PJSIP preventing me to use it in a
> production enviroment :-(
> I'm going to file these bugs at the moment...

Asterisk uses PJSIP at a different layer than a lot of other things. A
lot of the application logic and features are done by Asterisk, while
clients based on PJSIP use pjsua which takes care of that. You can't
compare the two.

As for your issues please do file them. I'd also suggest using bundled
PJSIP, it works the best with Asterisk and we backport applicable fixes
and include fixes we've created that have not yet made it to a PJSIP
release.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Hi,

Am 16.02.2017 um 14:19 schrieb Annus Fictus:
> And Microsip using PJSIP SIP stack :)

Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not 
sure if this really is a sign of good quality.
Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP 
in other software), but after just five minutes of testing
I found several bugs regarding PJSIP preventing me to use it in a production 
enviroment :-(
I'm going to file these bugs at the moment...


Max



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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[A J Stiles]

On Thursday 16 Feb 2017, Max Grobecker wrote:
> I'm a big fan of PhonerLite.
> It's more poplar in Germany, but also available in English language.
> This client supports TLS, SRTP and ZRTP:
> http://phonerlite.de/features_en.htm
> 
> Yes, the GUI is not that much user friendly as Zoiper is - but at least a
> very good and stable client for testing purposes ;-)

It seems to be Windows-only, though, and does not appear to include any Source 
Code  (even although the licence allows you to distribute modified versions).  
Either of those could potentially be a show-stopper.

Linphone  ( http://www.linphone.org/technical-corner/linphone/overview )  
supports TLS -- and is both cross-platform and GPLv2.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Annus Fictus]

And Microsip using PJSIP SIP stack :)


El 16/02/2017 a las 08:15, Jonathan H escribió:

Microsip (Windows) is free and small.
2.5Mb download, 10Mb RAM usage, does everything I need and configuring
TLS is a doddle.
http://www.microsip.org/

On 16 February 2017 at 13:04, Max Grobecker
 wrote:

Hello,

I'm a big fan of PhonerLite.
It's more poplar in Germany, but also available in English language.
This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm

Yes, the GUI is not that much user friendly as Zoiper is - but at least a very 
good and stable client for testing purposes ;-)

Max


Am 15.02.2017 um 19:46 schrieb Motty Cruz:

Hello, I have a user that prefers Soft SIP phone install on his laptop, for 
security reasons I have enable TLS on our Asterisk server to support TLS 
authentication, It works well with hard phones. Has anybody in this forum use 
SIP Soft phones with TLS authentication enabled? Any suggestions?



Thanks,
Motty





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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Jonathan H]

Microsip (Windows) is free and small.
2.5Mb download, 10Mb RAM usage, does everything I need and configuring
TLS is a doddle.
http://www.microsip.org/

On 16 February 2017 at 13:04, Max Grobecker
 wrote:
> Hello,
>
> I'm a big fan of PhonerLite.
> It's more poplar in Germany, but also available in English language.
> This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm
>
> Yes, the GUI is not that much user friendly as Zoiper is - but at least a 
> very good and stable client for testing purposes ;-)
>
> Max
>
>
> Am 15.02.2017 um 19:46 schrieb Motty Cruz:
>> Hello, I have a user that prefers Soft SIP phone install on his laptop, for 
>> security reasons I have enable TLS on our Asterisk server to support TLS 
>> authentication, It works well with hard phones. Has anybody in this forum 
>> use SIP Soft phones with TLS authentication enabled? Any suggestions?
>>
>>
>>
>> Thanks,
>> Motty
>>
>>
>>
>
>
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>
> New to Asterisk? Start here:
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>
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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Hello,

I'm a big fan of PhonerLite.
It's more poplar in Germany, but also available in English language.
This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm

Yes, the GUI is not that much user friendly as Zoiper is - but at least a very 
good and stable client for testing purposes ;-)

Max


Am 15.02.2017 um 19:46 schrieb Motty Cruz:
> Hello, I have a user that prefers Soft SIP phone install on his laptop, for 
> security reasons I have enable TLS on our Asterisk server to support TLS 
> authentication, It works well with hard phones. Has anybody in this forum use 
> SIP Soft phones with TLS authentication enabled? Any suggestions?
> 
>  
> 
> Thanks,
> Motty
> 
> 
> 



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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Glenn Geller (VDOPh)]

As far as I know, Zoiper does support TLS across all platforms.

We've tested and deployed this on their "paid" software on Android and
Windows, and it works well.

Thanks,

*Glenn @ VDOTel*

On Wed, Feb 15, 2017 at 2:54 PM, Marcelo Terres  wrote:

> Zoiper?
>
> On 15 Feb 2017 6:46 p.m., "Motty Cruz"  wrote:
>
>> Hello, I have a user that prefers Soft SIP phone install on his laptop,
>> for security reasons I have enable TLS on our Asterisk server to support
>> TLS authentication, It works well with hard phones. Has anybody in this
>> forum use SIP Soft phones with TLS authentication enabled? Any suggestions?
>>
>>
>>
>> Thanks,
>> Motty
>>
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>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
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>
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> org/
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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Marcelo Terres]

Zoiper?

On 15 Feb 2017 6:46 p.m., "Motty Cruz"  wrote:

> Hello, I have a user that prefers Soft SIP phone install on his laptop,
> for security reasons I have enable TLS on our Asterisk server to support
> TLS authentication, It works well with hard phones. Has anybody in this
> forum use SIP Soft phones with TLS authentication enabled? Any suggestions?
>
>
>
> Thanks,
> Motty
>
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> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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[asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Motty Cruz]

Hello, I have a user that prefers Soft SIP phone install on his laptop, for
security reasons I have enable TLS on our Asterisk server to support TLS
authentication, It works well with hard phones. Has anybody in this forum
use SIP Soft phones with TLS authentication enabled? Any suggestions? 

 

Thanks, 
Motty

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

[Pete Mundy]

+1! This sounds an awful lot like an ALG doing it best to 'help'...


> On 14/02/2017, at 6:38 am, Israel Gottlieb  wrote:
> 
> Disable all sip alg/helpers in the router



smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] First SIP-registering succeeds, second doesn't

[Israel Gottlieb]

 Disable all sip alg/helpers in the router


  Original Message  
From: andregronwal...@gmail.com
Sent: February 13, 2017 6:40 PM
To: asterisk-users@lists.digium.com
Reply-to: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] First SIP-registering succeeds, second doesn't

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

[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

[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] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Jonas Kellens]

Remove yourself !

Don't hijack my thread !



On 17-08-16 14:53, Dario Estupinan wrote:

REMOVE ME please.

2016-08-15 15:16 GMT-05:00 Jonas Kellens >:


Hello

after I have upgraded from Asterisk 12 to
asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved
in MySQL DB) register anymore.


[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5076 ' - Wrong password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5072 ' - Wrong password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5062 ' - Wrong password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5060 ' - Wrong password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5060 ' - Wrong password


Is this a known problem ??


Second question I have : can I get the complete list of columns
that can be used in realtime database for sip peers somewhere
(update for Ast 13) ? Are columns like dtlsenable, dtlsverify,
dtlscertfile, dtlscafile, dtlssetup possible ??




Thanks for the help.


Kind regards.

Jonas.

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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[John Novack]

Remove yourself

READ - Included with every message -

asterisk-users mailing list
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Dario Estupinan wrote:


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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Dario Estupinan]

REMOVE ME please.

2016-08-15 15:16 GMT-05:00 Jonas Kellens :

> Hello
>
> after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1
> none of my realtime SIP peers (saved in MySQL DB) register anymore.
>
>
> [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5076' - Wrong password
> [Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5072' - Wrong password
> [Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5062' - Wrong password
> [Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5060' - Wrong password
> [Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5060' - Wrong password
>
>
> Is this a known problem ??
>
>
> Second question I have : can I get the complete list of columns that can
> be used in realtime database for sip peers somewhere (update for Ast 13) ?
> Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile,
> dtlssetup possible ??
>
>
>
>
> Thanks for the help.
>
>
> Kind regards.
>
> Jonas.
>
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> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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destinatario. Si recibe este correo por error, por favor elimínelo y avise
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aprovechamiento o divulgación con cualquier propósito. La Corporación
Politécnica Nacional de Colombia no asume ninguna responsabilidad por
eventuales daños generados por el recibo y el uso de este material, siendo
responsabilidad del destinatario verificar con sus propios medios la
existencia de virus u otros defectos. El presente correo electrónico solo
refleja la opinión de su Remitente y no representa necesariamente la
opinión oficial de la Corporación o de sus Directivos.
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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Jonas Kellens]

On 15-08-16 23:00, Carlos Chavez wrote:



I highly recommend that you use alembic to set up your database as 
this will make sure you are always using the correct database schema.  
You should be able to find the "official" structure in the 
contrib/realtime/mysql directory of the Asterisk source.




Hello

in contrib/realtime/mysql I see a table 'sippeers' with a column 
"transport ENUM('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp') " but 
I see no columns dtlsenable, dtlsverify, dtlscertfile, dtlscafile, 
dtlssetup ?


So if we can define a sip peer with transport 'ws' or 'wss', then why 
are there no columns for the 'dtls'-part (which is kinda mandatory) ?




Kind regards.



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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Carlos Chavez]

On 8/15/16 3:16 PM, Jonas Kellens wrote:


Hello

after I have upgraded from Asterisk 12 to 
asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in 
MySQL DB) register anymore.



[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5076' - 
Wrong password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5072' - 
Wrong password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5062' - 
Wrong password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - 
Wrong password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - 
Wrong password



Is this a known problem ??


Second question I have : can I get the complete list of columns that 
can be used in realtime database for sip peers somewhere (update for 
Ast 13) ? Are columns like dtlsenable, dtlsverify, dtlscertfile, 
dtlscafile, dtlssetup possible ??



The first thing you need to test is if you are properly loading the 
realtime data.  The best way would be to enable "rtcachefriends=yes" and 
then "sip show peer XXX load".  If you are not getting anything then 
there is a problem with your realtime setup.  I used realtime sip until 
13.7 before switching to PJSIP so it should work.


I highly recommend that you use alembic to set up your database as 
this will make sure you are always using the correct database schema.  
You should be able to find the "official" structure in the 
contrib/realtime/mysql directory of the Asterisk source.


--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161


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[asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Jonas Kellens]

Hello

after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 
none of my realtime SIP peers (saved in MySQL DB) register anymore.



[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5076' - Wrong 
password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5072' - Wrong 
password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5062' - Wrong 
password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - Wrong 
password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - Wrong 
password



Is this a known problem ??


Second question I have : can I get the complete list of columns that can 
be used in realtime database for sip peers somewhere (update for Ast 13) 
? Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile, 
dtlssetup possible ??





Thanks for the help.


Kind regards.

Jonas.

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[asterisk-users] Some SIP OPTIONS packages seem to be ignored by the peer

[Michael Maier]

Hello!

Sometimes, I can see here the following scene:

Asterisk sends 11 SIP OPTIONS-packages (qualify=120) and they are all
ignored by the peers - but the 12. package is answered immediately as
expected (I'm sure there is no network related problem).

I can see this on trunks via Internet and locally with extensions - no
matter if it is via pjsip or traditional sip module.

What's the difference between the 11 packages and the 12. package?
It's the branch, tag and Call-ID.
The 11 packages before all do have the same branch, tag and Call-ID. The
12. package changes them.

Looks like some of these values are ignored by the peers. But why? Are
they broken?



Thanks,
kind regards,
Michael

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Re: [asterisk-users] Strange SIP debug

[Aqs Younas]

ACK is being forwarded to contact that you have in 200 ok. You need to
check 200 contact header.

On 17 May 2016 at 17:59, Антон Сацкий  wrote:

> Hi list need your advice
> i dont understand why reply ACK goes to diferrent ip (192.168.88.32)
> SCREEN below
>
> http://tinypic.com/view.php?pic=s6m7me=9#.VzsVhvl96Ik
>
> THANK U ALL
>
>
>
> --
> Best regards
> Antony
> tel.   +380669197533
> tel2. +380636564340
> Paypal http://paypal.me/Satskiy
> 
> satski...@gmail.com 
>
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[asterisk-users] Strange SIP debug

[Антон Сацкий]

Hi list need your advice
i dont understand why reply ACK goes to diferrent ip (192.168.88.32)
SCREEN below

http://tinypic.com/view.php?pic=s6m7me=9#.VzsVhvl96Ik

THANK U ALL



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tel2. +380636564340
Paypal http://paypal.me/Satskiy

satski...@gmail.com 
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[asterisk-users] Basic SIP NAT traversal question

[Gabriel Ortiz Lour]

Hi all,

  Checking on the asterisk source code I've seen that SIP will always use
the IP address in the "c=" field of SDP to send media. Is that correct?

  Is there a case where asterisk would send media to the received source IP
address instead of the one he got on the SDP?

  I know the externip and localnet config, but it seams that this only
changes the media IP address that asterisk informs on his SDP.

  The only way to solve it would be the endpoint be using STUN/ICE or I'm
missing something?

  scenario: endpoint inform his LAN address on his SDP and asterisk send
media to it, no audio.
asterisk=DMZ
endpoint behind NAT
[general]
externhost setted
localnet setted

[endpoint]
nat=yes

Thanks,
Gabriel
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Re: [asterisk-users] Windstream SIP Trunk settings

[James Cass]

Here's what I ultimately got to work (in case it helps someone):

Name your trunk
Enter your outgoing CID

Under Outgoing settings-
Trunk name - whatever you choose to name it
PEER Details-
host=IP address of SIP gateway
type=friend
context=from-trunk
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite

Incoming settings -

none

Registration string -
username:passw...@xxx.xxx.xxx.xxx

James Cass 
jcas...@gmail.com


On Tue, Feb 23, 2016 at 12:20 PM, Rodrigo Ramírez Norambuena <
decipher...@gmail.com> wrote:

> February 23 2016 9:37 AM, "James Cass"  wrote:
> > Thanks everyone, all sound advice. Still can't even get the calls to
> show up on the console at all
> > - I suspect the issue is on the WS side, as I'm not having any issues
> with other carriers with
> > similar settings.
>
> You can debug SIP to detect the problem. May be exists some cause tell you
> more information in the
> trace SIP.
> --
> Rodrigo Ramírez Norambuena
> http://www.rodrigoramirez.com
>
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Re: [asterisk-users] Abandoned SIP-TCP connection causes Asterisk to crash

[Joshua Colp]

Nasir Iqbal wrote:

Hi All,

We are using SIP over TCP transport but often we got an Asterisk crash
with following error.

[Mar  1 11:23:13] WARNING[1509]: chan_sip.c:3755 __sip_xmit: sip_xmit of
0x7f294000cac0 (len 680) to Soft.Phone.IP.Address:56780 returned -2:
Interrupted system call

Asterisk uncleanly ending (0).
Executing last minute cleanups
   == Destroying musiconhold processes
   == Manager unregistered action DBGet
   == Manager unregistered action DBPut
   == Manager unregistered action DBDel
   == Manager unregistered action DBDelTree


Interestingly this doesn't appear as though it's a crash. Whatever is 
happening is causing Asterisk to then shutdown normally.




We have tested this issue with Asterisk 11.20 and it can be reproduced
as following

1. From client workstation / labptop register a softphone with Asterisk
over TCP
2. Dial into some local asterisk extension. (i.e play voice message in
loop )
3. From client side kill the softphone i.e ( killall zoiper )
4. Wait some time, Asterisk will crash


Please file an issue[1] for this so we can narrow down the problem. I'm 
not aware of any currently open and I don't think I've seen it.


[1] https://issues.asterisk.org/jira

--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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[asterisk-users] Abandoned SIP-TCP connection causes Asterisk to crash

[Nasir Iqbal]

Hi All,

We are using SIP over TCP transport but often we got an Asterisk crash with
following error.

[Mar  1 11:23:13] WARNING[1509]: chan_sip.c:3755 __sip_xmit: sip_xmit of
0x7f294000cac0 (len 680) to Soft.Phone.IP.Address:56780 returned -2:
Interrupted system call

Asterisk uncleanly ending (0).
Executing last minute cleanups
  == Destroying musiconhold processes
  == Manager unregistered action DBGet
  == Manager unregistered action DBPut
  == Manager unregistered action DBDel
  == Manager unregistered action DBDelTree

We have tested this issue with Asterisk 11.20 and it can be reproduced as
following

1. From client workstation / labptop register a softphone with Asterisk
over TCP
2. Dial into some local asterisk extension. (i.e play voice message in loop
)
3. From client side kill the softphone i.e ( killall zoiper )
4. Wait some time, Asterisk will crash

Further we are unable to get any coredump ! (even running with -g) Any help
will be appreciated.

Thanks in advance

Nasir Iqbal

ICTBroadcast - an Auto Dialer software for ITSP

SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
http://www.ictbroadcast.com/
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Re: [asterisk-users] Windstream SIP Trunk settings

[Rodrigo Ramírez Norambuena]

February 23 2016 9:37 AM, "James Cass"  wrote:
> Thanks everyone, all sound advice. Still can't even get the calls to show up 
> on the console at all
> - I suspect the issue is on the WS side, as I'm not having any issues with 
> other carriers with
> similar settings.

You can debug SIP to detect the problem. May be exists some cause tell you more 
information in the
trace SIP.
--
Rodrigo Ramírez Norambuena
http://www.rodrigoramirez.com

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Re: [asterisk-users] Windstream SIP Trunk settings

[James Cass]

Thanks everyone, all sound advice.  Still can't even get the calls to show
up on the console at all - I suspect the issue is on the WS side, as I'm
not having any issues with other carriers with similar settings.

Thanks again.

James Cass 
jcas...@gmail.com


On Mon, Feb 22, 2016 at 9:06 AM, Mark Wiater 
wrote:

> In my case, username is the BTN. I also set the fromdomain to be the sbc
> that I'm registering with. Externip might help also?
>
> [paetec]
> host=10.250.0.5
> username=btn
> fromdomain=10.250.0.5
> dtmfmode=rfc2833
> externip=10.255.0.2
>
> I've used these settings on both registering and non-registering trunks,
> connecting to both the Broadworks and Plexus platforms in Windstream.
> Though all of my asterisk versions have been 1.8.x
>
> Mark
>
>
> On 2/22/2016 8:20 AM, James Cass wrote:
>
> Does anyone on this list use Windstream as a SIP trunk provider?
>
> If so, would you mind sharing your peer settings?
>
> I'm using asterisk 13.7.2 and can't seem to get the inbound working
> correctly (using registration).  Outbound is fine, but they are seeing an
> authentication error on their end.
>
> Here are my inbound peer settings:
>
> username=
> secret=
> host=
> type=peer
> fromuser=
> context=from-trunk
> dtmfmode=auto
> canreinvite=no
> qualify=yes
> insecure=port,invite
>
> register string:   :@:5060
>
> Thanks in advance,
>
>
>
>
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Re: [asterisk-users] Windstream SIP Trunk settings

[Mark Wiater]

In my case, username is the BTN. I also set the fromdomain to be the sbc
that I'm registering with. Externip might help also?

[paetec]
host=10.250.0.5
username=btn
fromdomain=10.250.0.5
dtmfmode=rfc2833
externip=10.255.0.2

I've used these settings on both registering and non-registering trunks,
connecting to both the Broadworks and Plexus platforms in Windstream.
Though all of my asterisk versions have been 1.8.x

Mark

On 2/22/2016 8:20 AM, James Cass wrote:
> Does anyone on this list use Windstream as a SIP trunk provider?
>
> If so, would you mind sharing your peer settings?
>
> I'm using asterisk 13.7.2 and can't seem to get the inbound working
> correctly (using registration).  Outbound is fine, but they are seeing
> an authentication error on their end.
>
> Here are my inbound peer settings:
>
> username=
> secret=
> host=
> type=peer
> fromuser=
> context=from-trunk
> dtmfmode=auto
> canreinvite=no
> qualify=yes
> insecure=port,invite
>
> register string:   :@:5060
>
> Thanks in advance,
>
>

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Re: [asterisk-users] Windstream SIP Trunk settings

[Frank]

On Mon, 2016-02-22 at 08:20 -0500, James Cass wrote:

> register string:   :@:5060

Try:

register => 5551231234:sec...@sipdomain.com/5551231234 


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[asterisk-users] Windstream SIP Trunk settings

[James Cass]

Does anyone on this list use Windstream as a SIP trunk provider?

If so, would you mind sharing your peer settings?

I'm using asterisk 13.7.2 and can't seem to get the inbound working
correctly (using registration).  Outbound is fine, but they are seeing an
authentication error on their end.

Here are my inbound peer settings:

username=
secret=
host=
type=peer
fromuser=
context=from-trunk
dtmfmode=auto
canreinvite=no
qualify=yes
insecure=port,invite

register string:   :@:5060

Thanks in advance,
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Re: [asterisk-users] force sip URI call through PBX

[D'Arcy J.M. Cain]

On Mon, 30 Nov 2015 16:42:02 +0100
Julien Sansonnens  wrote:
> When I do a SIP URI call from my softphone, the call is made directly
> to the destination host (p2p), bypassing the PBX. So I lose the
> possibility of recording, making statistics, etc ...
> 
> Is there a way to force URI calls through the PBX? I have found no
> configuration at the client or at the server level. Do you know any
> softphone that will allows me to do this ?

If two phones are calling each other directly then there is no server
setting that will reach across the Internet and grab the call.  You
need to insure that the call is proxied through your PBX.  That's just
a setup in your softphone.  You will need to ask about that on a
mailing list for that software.

One thing that I do so that I can call SIP phones from my regular phone
through an ATA is set up extensions for them in Asterisk like this.

exten => 6135553638,1,Dial(SIP/my.fri...@example.com)

Looks like a regular call to my users but bypasses the PSTN.  This
might work for you as well.

Whatever solution you use, you may want to look at directmedia
settings.  If you can talk directly to another SIP client Asterisk may
step out of the picture anyway not allowing you to record the call.
Turning that off forces all the calls to be proxied through you even if
they could talk directly.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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[asterisk-users] force sip URI call through PBX

[Julien Sansonnens]

Hello,

When I do a SIP URI call from my softphone, the call is made directly
to the destination host (p2p), bypassing the PBX. So I lose the
possibility of recording, making statistics, etc ...

Is there a way to force URI calls through the PBX? I have found no
configuration at the client or at the server level. Do you know any
softphone that will allows me to do this ?

Thank you and have a nice day, Julien


--
Julien Sansonnens

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Re: [asterisk-users] same sip username with realms and chan_sip

[Scott Griepentrog]

Just as a reminder: absolutely anytime that you succeed in crashing
Asterisk (no matter the validity of your input), please make sure that
either an issue covering the situation already exists, or please take the
time to create a new one.

When creating an issue (or if one is not already attached), please follow
these [1] instructions for obtaining a backtrace and attach the file to the
issue.  Very often a backtrace on an issue is sufficient for us to identify
and eliminate the bug that caused it.  And if you can, please replicate
using a currently supported version (11, 13, master) of Asterisk compiled
from the latest git head -- this helps us to be confident that it's not
something already fixed, and we can skip that step and get to fixing it
faster.

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

On Tue, Oct 13, 2015 at 5:22 AM, Ludovic Gasc  wrote:

> pjsip crashes only with my realm experiments.
> I'll test with the latest Asterisk 13 stable version to verify.
>
> However, even if I've found a solution for realm, I've the feeling that
> realm in Asterisk isn't well tested/supported.
>
> For now, since September, I use a simpler solution in production:
> integrate the account name as a prefix in the username: enough mainstream
> to be sure is supported ;-)
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
> On 11 Oct 2015 22:22, "Joshua Colp"  wrote:
>
>> Ludovic Gasc wrote:
>>
>>> Hello,
>>>
>>> same sip username with realms is possible with Asterisk ?
>>> I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and
>>> now, Asterisk crashes.
>>>
>>
>> Did PJSIP crash in general (it's usually a build problem if that happens)
>> or was it when you were experimenting with different realms and such?
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
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>>
>
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direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
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Re: [asterisk-users] same sip username with realms and chan_sip

[Ludovic Gasc]

pjsip crashes only with my realm experiments.
I'll test with the latest Asterisk 13 stable version to verify.

However, even if I've found a solution for realm, I've the feeling that
realm in Asterisk isn't well tested/supported.

For now, since September, I use a simpler solution in production: integrate
the account name as a prefix in the username: enough mainstream to be sure
is supported ;-)

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
On 11 Oct 2015 22:22, "Joshua Colp"  wrote:

> Ludovic Gasc wrote:
>
>> Hello,
>>
>> same sip username with realms is possible with Asterisk ?
>> I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and
>> now, Asterisk crashes.
>>
>
> Did PJSIP crash in general (it's usually a build problem if that happens)
> or was it when you were experimenting with different realms and such?
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] same sip username with realms and chan_sip

[Joshua Colp]

Ludovic Gasc wrote:

Hello,

same sip username with realms is possible with Asterisk ?
I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and
now, Asterisk crashes.


Did PJSIP crash in general (it's usually a build problem if that 
happens) or was it when you were experimenting with different realms and 
such?


--
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Re: [asterisk-users] same sip username with realms and chan_sip

[Ludovic Gasc]

Hello,

same sip username with realms is possible with Asterisk ?
I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and
now, Asterisk crashes.

Any clue ?

Have a nice week-end.

--
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/

2015-09-24 22:59 GMT+02:00 Ludovic Gasc :

> Hi,
>
> How have the same sip username in several realms ?
> For now, I must add the realm prefix in the sip username of chan_sip.
>
> For example:
> [lg_2540]
> amaflags = default
> call-limit = 10
> host = dynamic
> language = en_US
> context = lg_default
> callerid = "LG" <2540>
> secret = XX
> type = friend
> subscribemwi = no
> mohsuggest = default
> qualify = yes
> fromdomain=lg.allocloud.com
> fromuser=2540
>
> If I use only [2540] as section name, I'll have a clash on the same
> Asterisk.
>
> Thanks for your answers.
> --
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
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[asterisk-users] same sip username with realms and chan_sip

[Ludovic Gasc]

Hi,

How have the same sip username in several realms ?
For now, I must add the realm prefix in the sip username of chan_sip.

For example:
[lg_2540]
amaflags = default
call-limit = 10
host = dynamic
language = en_US
context = lg_default
callerid = "LG" <2540>
secret = XX
type = friend
subscribemwi = no
mohsuggest = default
qualify = yes
fromdomain=lg.allocloud.com
fromuser=2540

If I use only [2540] as section name, I'll have a clash on the same
Asterisk.

Thanks for your answers.
--
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
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