[asterisk-users] 180 Ringing missing

[marek]

hi,

after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know, 
its old. customer is very conservative...)


i have problem with missing 180 Ringing

flow is easy (PBX -> Asterisk -> SIP SBC)

Asterisk 11
PBX - Asterisk
-> INVITE
<- 100 Trying
<- 183 Session Progress
( <- RTP -> )
<- 180 Ringing
<- 200 OK

Asterisk 13
-> INVITE
<- 100 Trying
<- 183 Session Progress
( <- RTP -> )

__MISSING RINGING___

<- 200 OK

temporarily i solved problem with using "R" param

R: Default: Indicate ringing to the calling party, even if the called party
    isn't actually ringing. Allow interruption of the ringback if early 
media

    is received on the channel.

it changed to

Asterisk 13 (Dial(${ARG1},300,R)
-> INVITE
<- 100 Trying
<- 180 Ringing
<- 183 Session Progress
( <- RTP -> )
<- 200 OK

any ideas why Ringing is missing? any solutions?

Marek


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Re: [asterisk-users] 180 Ringing missing

[Joshua C. Colp]

On Tue, Dec 1, 2020 at 7:20 AM marek  wrote:

> hi,
>
> after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know,
> its old. customer is very conservative...)
>
> i have problem with missing 180 Ringing
>
> flow is easy (PBX -> Asterisk -> SIP SBC)
>
> Asterisk 11
> PBX - Asterisk
> -> INVITE
> <- 100 Trying
> <- 183 Session Progress
> ( <- RTP -> )
> <- 180 Ringing
> <- 200 OK
>
> Asterisk 13
> -> INVITE
> <- 100 Trying
> <- 183 Session Progress
> ( <- RTP -> )
>
> __MISSING RINGING___
>
> <- 200 OK
>
> temporarily i solved problem with using "R" param
>
> R: Default: Indicate ringing to the calling party, even if the called party
>  isn't actually ringing. Allow interruption of the ringback if early
> media
>  is received on the channel.
>
> it changed to
>
> Asterisk 13 (Dial(${ARG1},300,R)
> -> INVITE
> <- 100 Trying
> <- 180 Ringing
> <- 183 Session Progress
> ( <- RTP -> )
> <- 200 OK
>
> any ideas why Ringing is missing? any solutions?
>

Have you compared the signaling in both directions between the two versions
to see if there is a difference?

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] [External] 180 Ringing missing

[Floimair Florian]

If you have 183 Session progress, there is no need to send 180 Ringing 
(especially not AFTER 183 Session progress), as you already have early media 
instead. Having both is actually a bit misleading IMHO.

So this is actually correct. One should not rely on any of these 1xx 
"Provisional" messages.
They may or may not be sent, without violating SIP standards.

Am 01.12.20, 12:20 schrieb "asterisk-users im Auftrag von marek" 
:


hi,

after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know,
its old. customer is very conservative...)

i have problem with missing 180 Ringing

flow is easy (PBX -> Asterisk -> SIP SBC)

Asterisk 11
PBX - Asterisk
-> INVITE
<- 100 Trying
<- 183 Session Progress
( <- RTP -> )
<- 180 Ringing
<- 200 OK

Asterisk 13
-> INVITE
<- 100 Trying
<- 183 Session Progress
( <- RTP -> )

__MISSING RINGING___

<- 200 OK

temporarily i solved problem with using "R" param

R: Default: Indicate ringing to the calling party, even if the called party
 isn't actually ringing. Allow interruption of the ringback if early
media
 is received on the channel.

it changed to

Asterisk 13 (Dial(${ARG1},300,R)
-> INVITE
<- 100 Trying
<- 180 Ringing
<- 183 Session Progress
( <- RTP -> )
<- 200 OK

any ideas why Ringing is missing? any solutions?

Marek



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[asterisk-users] How to DIY/Setup An Open Source IP PBX Appliance/Server?

[Turritopsis Dohrnii Teo En Ming]

Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server?

Good day from Singapore,

After reading recent reviews, I gather that Asterisk is the gold 
standard when it comes to open source VoIP systems and it is the most 
famous open source PBX out there.


Article: Compare the Top 10 Best Open Source PBX Software of 2020
Link: 
https://www.voipreview.org/business-voip/best-open-source-pbx-software


Article: Top 10 Free Open Source PBX Software Solutions
Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/

The following is an excerpt from Wikipedia:

"Asterisk is a core component in many commercial products and 
open-source projects. Some of the commercial products are hardware and 
software bundles, for which the manufacturer supports and releases the 
software with an open-source distribution model.


AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software 
to realize all telephony functions.


AstLinux is a "Network Appliance for Communications" open-source 
software distribution.[15]


FreePBX, an open-source graphical user interface, bundles Asterisk as 
the core of its FreePBX Distro[16]


LinuxMCE bundles Asterisk to provide telephony; there is also an 
embedded version of Asterisk for OpenWrt routers.


PBX in a Flash/Incredible PBX and trixbox are software PBXes based on 
Asterisk.


Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer 
PBX, fax, instant messaging and email functions, respectively, before 
switching to 3CX.


Issabel is an open-source Unified Communications software which uses 
Asterisk for telephony functions. It was forked from the open-source 
versions of Elastix when 3CX acquired it.


*astTECS uses Asterisk in its VoIP and mobile gateways."

Link: https://en.wikipedia.org/wiki/Asterisk_(PBX)

I would like to DIY/setup an IP PBX appliance/server using free open 
source projects.
Which free open source project, mentioned in the list and links above, 
would you recommend to DIY my IP PBX appliance/server?


Should I buy a desktop computer or get one of those appliances listed in 
the link below to serve as my IP PBX appliance/server?


Link: 
https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celeron-j1800-dual-core-mini-pc-firewall-soft-router-with-ddr3l-msata-ssd-4-gigabit-lan-rj45-com-port-i449270007-s1196780479.html?spm=a2o42.searchlist.list.89.100857d22PjCYx&search=1


Please also refer me to very good, detailed and well explained 
guides/tutorials/manuals on setting up open source IP PBX 
appliances/servers.


Lastly, please recommend a cheap and affordable IP phone (suggest brand 
and model) to go along with my DIY open source IP PBX appliance/server.


Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 
Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore.


Thank you very much.








-BEGIN EMAIL SIGNATURE-

The Gospel for all Targeted Individuals (TIs):

[The New York Times] Microwave Weapons Are Prime Suspect in Ills of
U.S. Embassy Workers

Link: 
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html




Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's 
Academic
Qualifications as at 14 Feb 2019 and refugee seeking attempts at the 
United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 
2019) and Australia (25 Dec 2019 to 9 Jan 2020):


[1] https://tdtemcerts.wordpress.com/

[2] https://tdtemcerts.blogspot.sg/

[3] https://www.scribd.com/user/270125049/Teo-En-Ming

-END EMAIL SIGNATURE-

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Re: [asterisk-users] How to DIY/Setup An Open Source IP PBX Appliance/Server?

[John Novack]

JMO

AstLinux installed on an HP Thin Client is a good choice for someone with 
limited knowledge of Linux who wants a less steep learning curve.

YMMV


John Novack


Turritopsis Dohrnii Teo En Ming wrote:

Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server?

Good day from Singapore,

After reading recent reviews, I gather that Asterisk is the gold standard when 
it comes to open source VoIP systems and it is the most famous open source PBX 
out there.

Article: Compare the Top 10 Best Open Source PBX Software of 2020
Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software

Article: Top 10 Free Open Source PBX Software Solutions
Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/

The following is an excerpt from Wikipedia:

"Asterisk is a core component in many commercial products and open-source 
projects. Some of the commercial products are hardware and software bundles, for 
which the manufacturer supports and releases the software with an open-source 
distribution model.

AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to 
realize all telephony functions.

AstLinux is a "Network Appliance for Communications" open-source software 
distribution.[15]

FreePBX, an open-source graphical user interface, bundles Asterisk as the core 
of its FreePBX Distro[16]

LinuxMCE bundles Asterisk to provide telephony; there is also an embedded 
version of Asterisk for OpenWrt routers.

PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk.

Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, 
fax, instant messaging and email functions, respectively, before switching to 
3CX.

Issabel is an open-source Unified Communications software which uses Asterisk 
for telephony functions. It was forked from the open-source versions of Elastix 
when 3CX acquired it.

*astTECS uses Asterisk in its VoIP and mobile gateways."

Link: https://en.wikipedia.org/wiki/Asterisk_(PBX)

I would like to DIY/setup an IP PBX appliance/server using free open source 
projects.
Which free open source project, mentioned in the list and links above, would 
you recommend to DIY my IP PBX appliance/server?

Should I buy a desktop computer or get one of those appliances listed in the 
link below to serve as my IP PBX appliance/server?

Link: 
https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celeron-j1800-dual-core-mini-pc-firewall-soft-router-with-ddr3l-msata-ssd-4-gigabit-lan-rj45-com-port-i449270007-s1196780479.html?spm=a2o42.searchlist.list.89.100857d22PjCYx&search=1

Please also refer me to very good, detailed and well explained 
guides/tutorials/manuals on setting up open source IP PBX appliances/servers.

Lastly, please recommend a cheap and affordable IP phone (suggest brand and 
model) to go along with my DIY open source IP PBX appliance/server.

Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, 
is a TARGETED INDIVIDUAL (TI) living in Singapore.

Thank you very much.








-BEGIN EMAIL SIGNATURE-

The Gospel for all Targeted Individuals (TIs):

[The New York Times] Microwave Weapons Are Prime Suspect in Ills of
U.S. Embassy Workers

Link: 
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html



Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic
Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United 
Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and 
Australia (25 Dec 2019 to 9 Jan 2020):

[1] https://tdtemcerts.wordpress.com/

[2] https://tdtemcerts.blogspot.sg/

[3] https://www.scribd.com/user/270125049/Teo-En-Ming

-END EMAIL SIGNATURE-



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Re: [asterisk-users] [External] 180 Ringing missing

[marek]

i know

but there is some existing integration based on AMI event NewExten

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_NewExten 
ChannelStateDesc = Ringing


and if 180 Ringing is missing, there is no event

as you may have guessed, its hard to convice "Integrator" to "change" 
the code




Dne 01/12/2020 v 13:22 Floimair Florian napsal(a):

If you have 183 Session progress, there is no need to send 180 Ringing 
(especially not AFTER 183 Session progress), as you already have early media 
instead. Having both is actually a bit misleading IMHO.

So this is actually correct. One should not rely on any of these 1xx 
"Provisional" messages.
They may or may not be sent, without violating SIP standards.

Am 01.12.20, 12:20 schrieb "asterisk-users im Auftrag von marek" 
:


 hi,

 after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know,
 its old. customer is very conservative...)

 i have problem with missing 180 Ringing

 flow is easy (PBX -> Asterisk -> SIP SBC)

 Asterisk 11
 PBX - Asterisk
 -> INVITE
 <- 100 Trying
 <- 183 Session Progress
 ( <- RTP -> )
 <- 180 Ringing
 <- 200 OK

 Asterisk 13
 -> INVITE
 <- 100 Trying
 <- 183 Session Progress
 ( <- RTP -> )

 __MISSING RINGING___

 <- 200 OK

 temporarily i solved problem with using "R" param

 R: Default: Indicate ringing to the calling party, even if the called party
  isn't actually ringing. Allow interruption of the ringback if early
 media
  is received on the channel.

 it changed to

 Asterisk 13 (Dial(${ARG1},300,R)
 -> INVITE
 <- 100 Trying
 <- 180 Ringing
 <- 183 Session Progress
 ( <- RTP -> )
 <- 200 OK

 any ideas why Ringing is missing? any solutions?

 Marek





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Re: [asterisk-users] 180 Ringing missing

[marek]

Dne 01/12/2020 v 12:58 Joshua C. Colp napsal(a):
On Tue, Dec 1, 2020 at 7:20 AM marek > wrote:


hi,

after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know,
its old. customer is very conservative...)

i have problem with missing 180 Ringing

flow is easy (PBX -> Asterisk -> SIP SBC)

Asterisk 11
PBX - Asterisk
-> INVITE
<- 100 Trying
<- 183 Session Progress
( <- RTP -> )
<- 180 Ringing
<- 200 OK

Asterisk 13
-> INVITE
<- 100 Trying
<- 183 Session Progress
( <- RTP -> )

__MISSING RINGING___

<- 200 OK

temporarily i solved problem with using "R" param

R: Default: Indicate ringing to the calling party, even if the
called party
 isn't actually ringing. Allow interruption of the ringback if
early
media
 is received on the channel.

it changed to

Asterisk 13 (Dial(${ARG1},300,R)
-> INVITE
<- 100 Trying
<- 180 Ringing
<- 183 Session Progress
( <- RTP -> )
<- 200 OK

any ideas why Ringing is missing? any solutions?


Have you compared the signaling in both directions between the two 
versions to see if there is a difference?



whats your goal with this question?

asking if there are some side effects in incoming call ? (SBC -> 
Asterisk -> PBX)




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Re: [asterisk-users] 180 Ringing missing

[Joshua C. Colp]

On Tue, Dec 1, 2020 at 11:24 AM marek  wrote:

>
> Dne 01/12/2020 v 12:58 Joshua C. Colp napsal(a):
>
> On Tue, Dec 1, 2020 at 7:20 AM marek  wrote:
>
>> hi,
>>
>> after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know,
>> its old. customer is very conservative...)
>>
>> i have problem with missing 180 Ringing
>>
>> flow is easy (PBX -> Asterisk -> SIP SBC)
>>
>> Asterisk 11
>> PBX - Asterisk
>> -> INVITE
>> <- 100 Trying
>> <- 183 Session Progress
>> ( <- RTP -> )
>> <- 180 Ringing
>> <- 200 OK
>>
>> Asterisk 13
>> -> INVITE
>> <- 100 Trying
>> <- 183 Session Progress
>> ( <- RTP -> )
>>
>> __MISSING RINGING___
>>
>> <- 200 OK
>>
>> temporarily i solved problem with using "R" param
>>
>> R: Default: Indicate ringing to the calling party, even if the called
>> party
>>  isn't actually ringing. Allow interruption of the ringback if early
>> media
>>  is received on the channel.
>>
>> it changed to
>>
>> Asterisk 13 (Dial(${ARG1},300,R)
>> -> INVITE
>> <- 100 Trying
>> <- 180 Ringing
>> <- 183 Session Progress
>> ( <- RTP -> )
>> <- 200 OK
>>
>> any ideas why Ringing is missing? any solutions?
>>
>
> Have you compared the signaling in both directions between the two
> versions to see if there is a difference?
>
> whats your goal with this question?
>
> asking if there are some side effects in incoming call ? (SBC -> Asterisk
> -> PBX)
>

 No side effects, but looking at the actual SIP signaling (sip set debug
on) and see what the remote side is sending for SIP responses as well.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] [External] 180 Ringing missing

[Alex Hermann]

On dinsdag 1 december 2020 13:22:17 CET Floimair Florian wrote:
> If you have 183 Session progress, there is no need to send 180 Ringing
> (especially not AFTER 183 Session progress), as you already have
> early media instead. Having both is actually a bit misleading IMHO.

I disagree (and I think rfc3261 agrees).  How would the caller know if 
the callee has been alerted if he doesn't receive the 180?

183 and 180 have different meaning. 180 indicates the callee is being 
alerted. An 183 has no such meaning but is often used to setup early 
media (although any 1xx can do that). 180 and 183 are not mutually 
exclusive and in fact form a full matrix of possible and useful states. 
There can be early media without any phone ringing (announcement), 
ringing without early media (caller is expected to generate a ringback 
tone itself), no early media nor ringing and finally early media and 
ringing simultaneous.

Especially automata calling will want to know the difference.

Of course, a generated ringback tone by the caller should be stopped 
when media is received.

Asterisk should indicate to the caller the same state it received from 
the callee.
-- 
Alex Hermann



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[asterisk-users] Fwd: Legacy TDM400

[Roy Kidder]

Hello all,

It's been quite some number of years since I played around with Asterisk
and I'm just now getting back into it. I think the last version I worked
with was 1.8.

I have a legacy Digium TDM400 PCI card and am wondering if that will still
work on newer versions of Asterisk. My initial attempts on Debian 10 and
the Debian repository version of Asterisk didn't get me very far.

Any pointers would be appreciated.

-Roy
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Re: [asterisk-users] Fwd: Legacy TDM400

[John Novack SCII_U]

AFAIK it requires DAHDI version 2
For unknown reasons, this and many other card drivers were removed in DAHDI 
version 3

Suggest you compile from source, rather than any repository, selecting the last DAHDI version 2 and at least Asterisk 13, though it is EOL or nearly, it still is a good version to 
work with

For learning there isn't any good reason to have the latest of anything

I have a working version of Asterisk 13 with DAHDI and a 4 port T1 card on 
CentOS 6, and support a buddy with a TDM 400 or 410 - no issues

YMMV

John Novack

Roy Kidder wrote:


Hello all,

It's been quite some number of years since I played around with Asterisk and 
I'm just now getting back into it. I think the last version I worked with was 
1.8.

I have a legacy Digium TDM400 PCI card and am wondering if that will still work on newer versions of Asterisk. My initial attempts on Debian 10 and the Debian repository version 
of Asterisk didn't get me very far.


Any pointers would be appreciated.

-Roy




--
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