Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Naveen Albert]

On 3/11/2021 2:50 PM, Mike wrote:

Thank you for taking the time.  I believe you misunderstood my question.
Callerid presence is passed perfectly already, as shown through Verbose
commands on both sides of the SIP call.
I should have mentioned before that the scenario I mentioned was with 
IAX2 trunking. I don't use SIP for inter-switch trunking but I do for 
terminating lines. Maybe IAX2 and SIP handle Caller ID differently.

The CALLERID name and numbers
aren't passed properly ONLY when presence is "hidden".
Sorry, by this you mean that Asterisk does not see the number or that 
the telephone set does see it? It sounded like the former.

One thing to note is that prohib and unavailable function differently.
I looked at my code, and if a subscriber does not subscriber to Caller 
ID, I do something like the following:
same => 
n,ExecIf($["${GOSUB_RETVAL}"="1"|"${GOSUB_RETVAL}"="2"]?NoOp():Set(CALLERID(pres)=unavailable))


unavailable and prohib options are interpreted differently by different 
SIP clients (e.g. ATAs or softphones). I did some testing and found that 
there was inconsistent behavior with one option vs. the other.

As if Asterisk decided that since this is a hidden number, to replace the
number with "Anynomous " as opposed to letting the receiving
Asterisk process it as desired with whatever logic I choose.
It may not be Asterisk but the telephone itself. I tested with a soft 
phone and it shows Anonymous . However, an actual telephone 
set shows "PRIVATE", regardless of which option I choose. But, this is 
just a hunch, and I'm not certain.

I just tested without any u() or f() or s() functions - same result. No
improvement or degradation with my issue. (Not sure why I had these
options)


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Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[phreak]

I've been able to pass presentation status between tandems without 
needing to do anything explicitly. This seems to be part of the Caller 
ID that is transmitted without explicit intervention. Have you tested 
without using the u option? I've never used the u option and not had 
issues with presentation transmitting as supposed to. This is with 
manual Set(CALLERID(pres)=something) and then seeing if it gets honored 
on the other end. The remote Asterisk gets the full number, of course, 
but the called telephone does not display it. Perhaps the u option is 
intended for lines, not trunks, and so the number never gets sent?


The only time I've found I (may) need to explicitly account for 
presentation is if I am regenerating the call, and then this needs to be 
accounted for in the call file (ignore that if that makes no sense).


The only inconsistency I've encountered has to do with presentation 
mismatches between the name and the number. If, for instance, I want the 
number to display but not the same, setting the presentations as the 
documentation would suggest does not work. The behavior is inconsistent 
between different SIP clients and it didn't work for me in any logical 
way. I didn't bother to a file a bug report about it, as I worked around 
this by simply doing Set(CALLERID(name)=) to empty the name and write 
the original name back into the variable after the call. Your mileage 
may vary.


NA

On 3/11/2021 2:22 PM, Mike wrote:


Hi,

Using Asterisk 13.36.0

I have a bit of a technical issue with hidden caller IDs.  My setup, 
at the moment, is composed of two Asterisk boxes. In some instance, 
calls arrive on Asterisk A, and are then sent to Asterisk B for 
further processing. The link between them is SIP (both on the same 
switch/LAN). Asterisk A has a Digium PRI card (recent one) and a PRI link.


When I receive a hidden number (i.e. “presentation prohibited”) call 
on Asterisk A through PRI, I get the following Caller ID information 
(using 444-555- as example):


“ <444555>”

And

CallerID presence is received as “prohibited_not_screened”.

Which is fine – I know the incoming number BUT I am told not to show 
it to the end user. All good.


The problem is when calls are not processed on Asterisk A, but sent to 
Asterisk B for further processing. The dial command I used on Asterisk 
A to send calls to AsterisB is the following:


exten => s,n,Dial(SIP/AsteriskB/123,,f("" 
<444555>)u(prohib_not_screened))


Again, so far so good. But, on Asterisk B in the appropriate context, 
on extension 123, my first command is a Verbose to show Callerid(all) 
and the received called id is shown as “Anonymous ” with 
CALLERID presence still “prohib_not_screened”. I would like Asterisk B 
to receive the actual callerid (“ <444555>”) along with the 
appropriate CallerID presence value (which is correct already).


Basically I want to “pass forward” both CALLERID and CALLERIDPRES 
exactly as received on AteriskA to AsteriskB so that AsteriskB gets 
the exact same info AsteriskA had in the first place.


How do I accomplish this?

Michael



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Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Mike]

THANK YOU! Case closed, that was indeed the problem.

Michael





From: asterisk-users  On Behalf Of 
Joshua C. Colp
Sent: March 11, 2021 15:52
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] CallerID presentation - presentation 
prohibited but still passing number



On Thu, Mar 11, 2021 at 4:50 PM Mike mailto:mich...@virtutel.ca> > wrote:

Thank you for taking the time.  I believe you misunderstood my question.
Callerid presence is passed perfectly already, as shown through Verbose
commands on both sides of the SIP call. The CALLERID name and numbers
aren't passed properly ONLY when presence is "hidden".

As if Asterisk decided that since this is a hidden number, to replace the
number with "Anynomous " as opposed to letting the receiving
Asterisk process it as desired with whatever logic I choose.

I just tested without any u() or f() or s() functions - same result. No
improvement or degradation with my issue. (Not sure why I had these
options)



You probably want to set the "trust_id_outbound" option[1] to "yes".



[1] 
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L377



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Joshua C. Colp

Asterisk Technical Lead

Sangoma Technologies

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www.asterisk.org <http://www.asterisk.org/>

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Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Joshua C. Colp]

On Thu, Mar 11, 2021 at 4:50 PM Mike  wrote:

> Thank you for taking the time.  I believe you misunderstood my question.
> Callerid presence is passed perfectly already, as shown through Verbose
> commands on both sides of the SIP call. The CALLERID name and numbers
> aren't passed properly ONLY when presence is "hidden".
>
> As if Asterisk decided that since this is a hidden number, to replace the
> number with "Anynomous " as opposed to letting the receiving
> Asterisk process it as desired with whatever logic I choose.
>
> I just tested without any u() or f() or s() functions - same result. No
> improvement or degradation with my issue. (Not sure why I had these
> options)
>

You probably want to set the "trust_id_outbound" option[1] to "yes".

[1]
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L377

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Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Mike]

Thank you for taking the time.  I believe you misunderstood my question.
Callerid presence is passed perfectly already, as shown through Verbose
commands on both sides of the SIP call. The CALLERID name and numbers
aren't passed properly ONLY when presence is "hidden".

As if Asterisk decided that since this is a hidden number, to replace the
number with "Anynomous " as opposed to letting the receiving
Asterisk process it as desired with whatever logic I choose.

I just tested without any u() or f() or s() functions - same result. No
improvement or degradation with my issue. (Not sure why I had these
options)




-Original Message-
From: phr...@phreaknet.org 
Sent: March 11, 2021 15:33
To: Mike ; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CallerID presentation - presentation
prohibited but still passing number

I've been able to pass presentation status between tandems without needing
to do anything explicitly. This seems to be part of the Caller ID that is
transmitted without explicit intervention. Have you tested without using
the u option? I've never used the u option and not had issues with
presentation transmitting as supposed to. This is with manual
Set(CALLERID(pres)=something) and then seeing if it gets honored on the
other end. The remote Asterisk gets the full number, of course, but the
called telephone does not display it. Perhaps the u option is intended for
lines, not trunks, and so the number never gets sent?

The only time I've found I (may) need to explicitly account for
presentation is if I am regenerating the call, and then this needs to be
accounted for in the call file (ignore that if that makes no sense).

The only inconsistency I've encountered has to do with presentation
mismatches between the name and the number. If, for instance, I want the
number to display but not the same, setting the presentations as the
documentation would suggest does not work. The behavior is inconsistent
between different SIP clients and it didn't work for me in any logical
way. I didn't bother to a file a bug report about it, as I worked around
this by simply doing Set(CALLERID(name)=) to empty the name and write the
original name back into the variable after the call. Your mileage may
vary.

NA

On 3/11/2021 2:22 PM, Mike wrote:
>
> Hi,
>
> Using Asterisk 13.36.0
>
> I have a bit of a technical issue with hidden caller IDs.  My setup,
> at the moment, is composed of two Asterisk boxes. In some instance,
> calls arrive on Asterisk A, and are then sent to Asterisk B for
> further processing. The link between them is SIP (both on the same
> switch/LAN). Asterisk A has a Digium PRI card (recent one) and a PRI
link.
>
> When I receive a hidden number (i.e. “presentation prohibited”) call
> on Asterisk A through PRI, I get the following Caller ID information
> (using 444-555- as example):
>
> “ <444555>”
>
> And
>
> CallerID presence is received as “prohibited_not_screened”.
>
> Which is fine – I know the incoming number BUT I am told not to show
> it to the end user. All good.
>
> The problem is when calls are not processed on Asterisk A, but sent to
> Asterisk B for further processing. The dial command I used on Asterisk
> A to send calls to AsterisB is the following:
>
> exten => s,n,Dial(SIP/AsteriskB/123,,f(""
> <444555>)u(prohib_not_screened))
>
> Again, so far so good. But, on Asterisk B in the appropriate context,
> on extension 123, my first command is a Verbose to show Callerid(all)
> and the received called id is shown as “Anonymous ” with
> CALLERID presence still “prohib_not_screened”. I would like Asterisk B
> to receive the actual callerid (“ <444555>”) along with the
> appropriate CallerID presence value (which is correct already).
>
> Basically I want to “pass forward” both CALLERID and CALLERIDPRES
> exactly as received on AteriskA to AsteriskB so that AsteriskB gets
> the exact same info AsteriskA had in the first place.
>
> How do I accomplish this?
>
> Michael
>

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[asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Mike]

Hi,

 

Using Asterisk 13.36.0

 

I have a bit of a technical issue with hidden caller IDs.  My setup, at
the moment, is composed of two Asterisk boxes. In some instance, calls
arrive on Asterisk A, and are then sent to Asterisk B for further
processing. The link between them is SIP (both on the same switch/LAN).
Asterisk A has a Digium PRI card (recent one) and a PRI link.

 

When I receive a hidden number (i.e. "presentation prohibited") call on
Asterisk A through PRI, I get the following Caller ID information (using
444-555- as example):

" <444555>" 

And 

CallerID presence is received as "prohibited_not_screened".

 

Which is fine - I know the incoming number BUT I am told not to show it to
the end user. All good.

 

The problem is when calls are not processed on Asterisk A, but sent to
Asterisk B for further processing.  The dial command I used on Asterisk A
to send calls to AsterisB is the following:

exten => s,n,Dial(SIP/AsteriskB/123,,f(""
<444555>)u(prohib_not_screened))

 

Again, so far so good. But, on Asterisk B in the appropriate context, on
extension 123, my first command is a Verbose to show Callerid(all) and the
received called id is shown as "Anonymous " with CALLERID
presence still "prohib_not_screened". I would like Asterisk B to receive
the actual callerid (" <444555>") along with the appropriate CallerID
presence value (which is correct already). 

 

Basically I want to "pass forward" both CALLERID and CALLERIDPRES exactly
as received on AteriskA to AsteriskB so that AsteriskB gets the exact same
info AsteriskA had in the first place.

 

How do I accomplish this?

 

 

Michael

 

 

 

 

 

 

 

 

 

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Re: [asterisk-users] CallerID fail with Voicetrading operator

[Administrator]

Hi Antony

Le 18/06/2020 à 20:19, Antony Stone a écrit :

On Thursday 18 June 2020 at 19:57:03, Administrator wrote:


does some people here use https://voicetrading.com which is a Dellmont
service from Netherlands. At the high begining they were also known as
Finarea (CH and DE mixed Co)
Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or
equal to CALLERID(num). We tried replacing + with 00, same problem.

There support said they don't receive any callerID and we know that it's
not correct as we don't have problem with other providers like
PeopleFone, Sipgate or French Operators.

Can you capture an INVITE packet from your system to theirs and show them the
headers?

Capture done and sended, thanks,

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[asterisk-users] CallerID fail with Voicetrading operator

[Administrator]

Hello,

does some people here use https://voicetrading.com which is a Dellmont 
service from Netherlands. At the high begining they were also known as 
Finarea (CH and DE mixed Co)


Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our 
callerID is no more seen by them. We use


Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or 
equal to CALLERID(num). We tried replacing + with 00, same problem.


There support said they don't receive any callerID and we know that it's 
not correct as we don't have problem with other providers like 
PeopleFone, Sipgate or French Operators.


If anyone had a clue, a ticket is open on their side but we don't fill 
that they want to react.


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Re: [asterisk-users] CallerID fail with Voicetrading operator

[Michel FACERIAS]

Hi.

You can continue tu use chan_sip on asterisk16. I use it on my own.

Have you got any capture of SIP dialog between operator sbc and your 
ipbx ?


++

Michel


Le 2020-06-18 19:57, Administrator a écrit :

Hello,

does some people here use https://voicetrading.com which is a Dellmont
service from Netherlands. At the high begining they were also known as
Finarea (CH and DE mixed Co)

Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our
callerID is no more seen by them. We use

Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or
equal to CALLERID(num). We tried replacing + with 00, same problem.

There support said they don't receive any callerID and we know that
it's not correct as we don't have problem with other providers like
PeopleFone, Sipgate or French Operators.

If anyone had a clue, a ticket is open on their side but we don't fill
that they want to react.

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Re: [asterisk-users] CallerID fail with Voicetrading operator

[Antony Stone]

On Thursday 18 June 2020 at 19:57:03, Administrator wrote:

> does some people here use https://voicetrading.com which is a Dellmont
> service from Netherlands. At the high begining they were also known as
> Finarea (CH and DE mixed Co)

> Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or
> equal to CALLERID(num). We tried replacing + with 00, same problem.
> 
> There support said they don't receive any callerID and we know that it's
> not correct as we don't have problem with other providers like
> PeopleFone, Sipgate or French Operators.

Can you capture an INVITE packet from your system to theirs and show them the 
headers?


Antony.

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[asterisk-users] CallerID(num-pres) not set during incomming call

[Benoit Panizzon]

Hello List

I can set CALLERID(num-pres)=prohib on a sip channel and asterisk is
setting the headers more or less correctly (PAI Header is missing
maching the call untrackable, which is a bit odd).

But when asterisk is handling an incomming call from:

From: Anonymous
;tag=3714732606-1699825061

And then I query CALLERID(num-pres) it contains: allowed_not_screened

Well the call has a prohibited CallerID. Why is the variable not being
set accordingly?

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Re: [asterisk-users] CallerID matching failure, possibly bug.

[Joshua Colp]

On Wed, Aug 2, 2017, at 08:37 AM, Blank Field wrote:
> Hello everyone.
> 
> Seems like i've managed to isolate a troubling behaviour on my asterisk.
> CallerID pattern matching does not work on the first try.
> 
> Technical info below:
> asterisk*CLI> core show version
> Asterisk 14.5.0 built by admin @ asterisk.domain on a x86_64 running
> Linux on 2017-06-13 14:26:54 UTC
> 
> I have an endpoint 616 with CALLERID(name) set to '616' and
> CALLERID(num) set to '616' at the user device. The endpoint is
> registered at asterisk as 616. Contact is 616@endpoint_ip.



This is likely a result of the PJSIP channel driver and not any core
things. It may not be applying the caller ID early enough. Since you
have a case that reproduces it go ahead and file an issue[1].

Cheers,

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

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[asterisk-users] CallerID matching failure, possibly bug.

[Blank Field]

Hello everyone.

Seems like i've managed to isolate a troubling behaviour on my asterisk.
CallerID pattern matching does not work on the first try.

Technical info below:
asterisk*CLI> core show version
Asterisk 14.5.0 built by admin @ asterisk.domain on a x86_64 running
Linux on 2017-06-13 14:26:54 UTC

I have an endpoint 616 with CALLERID(name) set to '616' and
CALLERID(num) set to '616' at the user device. The endpoint is
registered at asterisk as 616. Contact is 616@endpoint_ip.

[cidmatch]
exten => _.,1,NoOp()
exten => _./_6XX,1,SayDigits(1)
same => 2,SayDigits(2)
same => 3,SayDigits(3)
same => 4,SayDigits(4)

asterisk*CLI> dialplan show cidmatch
[ Context 'cidmatch' created by 'pbx_config' ]
  '_.' (CID match '_6XX') =>  1. SayDigits(1)
 [pbx_config]
2. SayDigits(2)
 [pbx_config]
3. SayDigits(3)
 [pbx_config]
4. SayDigits(4)
 [pbx_config]
  '_.' =>   1. NoOp()
 [pbx_config]

-= 2 extensions (5 priorities) in 1 context. =-

Please note two pattern matching attempts.

That way, SayDigits app works, and the digits are played.

If I comment out the first like, matching _., the following situation
happens:

[cidmatch]
;exten => _.,1,NoOp()
exten => _./_6XX,1,SayDigits(1)
same => 2,SayDigits(2)
same => 3,SayDigits(3)
same => 4,SayDigits(4)

asterisk*CLI> dialplan show cidmatch
[ Context 'cidmatch' created by 'pbx_config' ]
  '_.' (CID match '_6XX') =>  1. SayDigits(1)
 [pbx_config]
2. SayDigits(2)
 [pbx_config]
3. SayDigits(3)
 [pbx_config]
4. SayDigits(4)
 [pbx_config]

-= 1 extension (4 priorities) in 1 context. =-

[2017-07-25 15:03:32.037] NOTICE[13524]: res_pjsip_session.c:2141
new_invite: Call from '616' (UDP:IP:PORT) to extension '1' rejected
because extension not found in context 'cidmatch'.

To verify that CALLERID is correct:

[cidmatch]
exten => _.,1,Verbose(1,name: ${CALLERID(name)} num: ${CALLERID(num)})
;exten => _./_6XX,1,SayDigits(1)
same => 2,SayDigits(2)
same => 3,SayDigits(3)
same => 4,SayDigits(4)

asterisk*CLI> dialplan show cidmatch
[ Context 'cidmatch' created by 'pbx_config' ]
  '_.' =>   1. Verbose(1,name: ${CALLERID(name)} num:
${CALLERID(num)}) [pbx_config]
2. SayDigits(2)
 [pbx_config]
3. SayDigits(3)
 [pbx_config]
4. SayDigits(4)
 [pbx_config]

-= 1 extension (4 priorities) in 1 context. =-

 -- Executing [1@cidmatch:1] Verbose("PJSIP/616-0b2f", "1,name:
616 num: 616") in new stack
 name: 616 num: 616
-- Executing [1@cidmatch:2] SayDigits("PJSIP/616-0b2f", "2")
in new stack

That seriously took me some time to investigate.
https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching
The part on "Matching on Caller ID", aka "ex-girlfriend logic" seems
to be broken on my build.

Any advice appreciated, but it works for now.

Regards, Duelist.
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Re: [asterisk-users] CallerId presence issue

[Mike]

Thank you - At first glance it seems to have done the trick.

Mike


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: June 14, 2017 10:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerId presence issue

On Wed, Jun 14, 2017 at 10:18:19AM -0400, Mike wrote:
> I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
> PBX_A gets PRI calls on a 4 port Digium card, and each call naturally 
> has its own callerid values and presence.  I pass on those calls to 
> PBX_B via SI, and I'm trying to pass on this CALLERID info to PBX_B as
well.
> 
> My relevant dialplan snippet on PBX_A is:
> exten =>
> 1,1,Dial(SIP/pbx_b/55,,f(${CALLERID(all)})u(${CALLERID(pres)})
> ))
...
> I'm clearly missing something to pass on the callerid presence state 
> via the SIP link, but I can't figure out what.

Never heard of this method, are you sure this works for SIP, sound more
like for ISDN (look at packet captures).

But the/a standardized method is to use the P-Asserted-Identity and
Privacy headers (rfc3325). This should work if you set in the peer configs
in sip.conf on both sides:
sendrpid=pai
trustrpid=yes

Or you can do header manipulation/getting/setting manualy if desired.

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Re: [asterisk-users] CallerId presence issue

[Daniel Tryba]

On Wed, Jun 14, 2017 at 10:18:19AM -0400, Mike wrote:
> I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
> PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
> its own callerid values and presence.  I pass on those calls to PBX_B via
> SI, and I'm trying to pass on this CALLERID info to PBX_B as well. 
> 
> My relevant dialplan snippet on PBX_A is:
> exten =>
> 1,1,Dial(SIP/pbx_b/55,,f(${CALLERID(all)})u(${CALLERID(pres)})))
...
> I'm clearly missing something to pass on the callerid presence state via
> the SIP link, but I can't figure out what.

Never heard of this method, are you sure this works for SIP, sound more
like for ISDN (look at packet captures).

But the/a standardized method is to use the P-Asserted-Identity and
Privacy headers (rfc3325). This should work if you set in the peer configs 
in sip.conf on both sides:
sendrpid=pai
trustrpid=yes

Or you can do header manipulation/getting/setting manualy if desired.

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Re: [asterisk-users] CallerId presence issue

[Mike]

Actually, a correction: the callerid isn't passed on properly either: on
SIP_B I get "Anonymous " instead of "  <514-555-1234>"  that my
dial app is sending.

 

The exact dial command that is used, once variables are evaluated, is
this:

Dial(SIP/pbx3/555,,f("" <5145551234>)u(prohib_not_screened))"

 

While the log value found on the other end of the sip link are evaluated,
I get this:

Callerid name: Anonymous callerid number: number: anonymous  Presence
information : allowed_not_screened - allowed_not_screened -
allowed_not_screened

 

Somewhere in this Dial(SIP/) command callerid info is changed.  An
asterisk verbose check does not show me anything that would change
callerid info.

 

Mike

 

 

 

 

 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: June 14, 2017 10:18
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] CallerId presence issue

 

Hi,

 

I've run into a minor snag trying to pass on CALLERID presence from one
Asterisk to another via SIP (both running 13.16.0)

 

I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
its own callerid values and presence.  I pass on those calls to PBX_B via
SI, and I'm trying to pass on this CALLERID info to PBX_B as well. 

 

My relevant dialplan snippet on PBX_A is:

 

exten =>
1,1,Dial(SIP/pbx_b/55,,f(${CALLERID(all)})u(${CALLERID(pres)})))

 

*the u() value being dynamically taken from the channel itself.

 

On pbx_b, I  have a simply verbose line like this:

exten => 55,1,Verbose(1,Presence information :
${CALLERID(num-pres)} - ${CALLERID(name-pres)} - ${CALLERPRES()})

 

 

Here is my experience with this: whenever "prohib_not_screened" (tested
via a cell phone with hidden caller id info) is sent in the u() value of
the Dial application, pbx_b always gets "allowed_not_screened" as presence
state.Short version: the callerid presence seems lost on the SIP link.
The callerid info isn't, name and number are fine.

 

I'm clearly missing something to pass on the callerid presence state via
the SIP link, but I can't figure out what.

 

Any help or hint would be appreciated.

 

Michael

 

 

 

 

 

 

 

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[asterisk-users] CallerId presence issue

[Mike]

Hi,

 

I've run into a minor snag trying to pass on CALLERID presence from one
Asterisk to another via SIP (both running 13.16.0)

 

I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP.
PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has
its own callerid values and presence.  I pass on those calls to PBX_B via
SI, and I'm trying to pass on this CALLERID info to PBX_B as well. 

 

My relevant dialplan snippet on PBX_A is:

 

exten =>
1,1,Dial(SIP/pbx_b/55,,f(${CALLERID(all)})u(${CALLERID(pres)})))

 

*the u() value being dynamically taken from the channel itself.

 

On pbx_b, I  have a simply verbose line like this:

exten => 55,1,Verbose(1,Presence information :
${CALLERID(num-pres)} - ${CALLERID(name-pres)} - ${CALLERPRES()})

 

 

Here is my experience with this: whenever "prohib_not_screened" (tested
via a cell phone with hidden caller id info) is sent in the u() value of
the Dial application, pbx_b always gets "allowed_not_screened" as presence
state.Short version: the callerid presence seems lost on the SIP link.
The callerid info isn't, name and number are fine.

 

I'm clearly missing something to pass on the callerid presence state via
the SIP link, but I can't figure out what.

 

Any help or hint would be appreciated.

 

Michael

 

 

 

 

 

 

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Re: [asterisk-users] CALLERID on pjsip doesn't work?

[Andrew Ivins]

Thanks Joshua. That did the trick.

On 4 July 2016 at 19:18, Joshua Colp  wrote:

> Andrew Ivins wrote:
>
>> On 1 July 2016 at 17:41, Joshua Colp > > wrote:
>>
>>
>> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
>> same => n,Dial(PJSIP/phone123, 30)
>>
>>
>> Your exten line has no priority, is that how it is in your dialplan?
>>
>>
>> Actually no, I stole that line from an earlier email to this list. Mine
>> has a priority.
>>
>> If not you can isolate things a bit further by trying the following:
>>
>> Set(CALLERID(all)=Jon Doe <+123456789>)
>>
>> Or individually:
>>
>> Set(CALLERID(name)=Jon Doe)
>> Set(CALLERID(num)=+123456789)
>>
>>
>> Tried many permutations of this, and the only thing I can get to happen
>> is to make the call present as Anonymous by changing the
>> pres-name/pres-num setting.
>>
>> It's not a production system, dialplan is pretty simple:
>>
>> same => _X.1,Set(CALLERID(name-pres)=allowed)
>> same => n,Set(CALLERID(num-pres)=allowed)
>> same => n,Set(CALLERID(name)=Fred)
>> same => n,Set(CALLERID(num)=6123)
>> same => n,Dial(PJSIP/DEADDEADBEEF, 30)
>> same => n,Hangup()
>>
>> DEADDEADBEEF is the name of the endpoint and the endpoint works. I use
>> MAC addresses and plan to dynamically map extensions to them later on
>> (kind of like user mode in freepbx).
>>
>> In the console, if I log the value of CALLERID, it is what I expect to
>> it to be.
>>
>>
> 
>
> You have from_user set which will override the user in the From header
> which is where callerid would be. You also don't have send_rpid or send_pai
> turned on so there would be no alternate way to send it. Try setting
> send_rpid or send_pai to yes and trying again.
>
>
> --
> 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] CALLERID on pjsip doesn't work?

[Joshua Colp]

Andrew Ivins wrote:

On 1 July 2016 at 17:41, Joshua Colp > wrote:


exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
same => n,Dial(PJSIP/phone123, 30)


Your exten line has no priority, is that how it is in your dialplan?


Actually no, I stole that line from an earlier email to this list. Mine
has a priority.

If not you can isolate things a bit further by trying the following:

Set(CALLERID(all)=Jon Doe <+123456789>)

Or individually:

Set(CALLERID(name)=Jon Doe)
Set(CALLERID(num)=+123456789)


Tried many permutations of this, and the only thing I can get to happen
is to make the call present as Anonymous by changing the
pres-name/pres-num setting.

It's not a production system, dialplan is pretty simple:

same => _X.1,Set(CALLERID(name-pres)=allowed)
same => n,Set(CALLERID(num-pres)=allowed)
same => n,Set(CALLERID(name)=Fred)
same => n,Set(CALLERID(num)=6123)
same => n,Dial(PJSIP/DEADDEADBEEF, 30)
same => n,Hangup()

DEADDEADBEEF is the name of the endpoint and the endpoint works. I use
MAC addresses and plan to dynamically map extensions to them later on
(kind of like user mode in freepbx).

In the console, if I log the value of CALLERID, it is what I expect to
it to be.





You have from_user set which will override the user in the From header 
which is where callerid would be. You also don't have send_rpid or 
send_pai turned on so there would be no alternate way to send it. Try 
setting send_rpid or send_pai to yes and trying again.


--
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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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Re: [asterisk-users] CALLERID on pjsip doesn't work?

[Andrew Ivins]

On 1 July 2016 at 17:41, Joshua Colp  wrote:
>
>
>> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
>> same => n,Dial(PJSIP/phone123, 30)
>>
>
> Your exten line has no priority, is that how it is in your dialplan?
>

Actually no, I stole that line from an earlier email to this list. Mine has
a priority.


> If not you can isolate things a bit further by trying the following:
>
> Set(CALLERID(all)=Jon Doe <+123456789>)
>
> Or individually:
>
> Set(CALLERID(name)=Jon Doe)
> Set(CALLERID(num)=+123456789)
>

Tried many permutations of this, and the only thing I can get to happen is
to make the call present as Anonymous by changing the pres-name/pres-num
setting.

It's not a production system, dialplan is pretty simple:

same => _X.1,Set(CALLERID(name-pres)=allowed)
same => n,Set(CALLERID(num-pres)=allowed)
same => n,Set(CALLERID(name)=Fred)
same => n,Set(CALLERID(num)=6123)
same => n,Dial(PJSIP/DEADDEADBEEF, 30)
same => n,Hangup()

DEADDEADBEEF is the name of the endpoint and the endpoint works. I use MAC
addresses and plan to dynamically map extensions to them later on (kind of
like user mode in freepbx).

In the console, if I log the value of CALLERID, it is what I expect to it
to be.

In the pjsip debug, the callerid I am trying to set doesn't appear anywhere.

I'm using your Sorcery stuff backing into astb for pjsip, but I've done a
little script to dump it back into text so I can override it in the config
file. Therefore it's a bit verbose. Thanks for looking.

[DEADDEADBEEF]
type=aor
support_path=true
default_expiration=3600
qualify_timeout=3.00
mailboxes=
minimum_expiration=60
outbound_proxy=
voicemail_extension=
maximum_expiration=7200
qualify_frequency=0
authenticate_qualify=false
contact=
max_contacts=1
remove_existing=true

[DEADDEADBEEF]
type=auth
md5_cred=
realm=
auth_type=userpass
password=4D7D9A7F1822
nonce_lifetime=32
username=507B495E565B

[DEADDEADBEEF]
type=endpoint
timers_sess_expires=1800
device_state_busy_at=0
dtls_cipher=
from_domain=
dtls_rekey=0
dtls_fingerprint=SHA-256
direct_media_method=invite
send_rpid=false
pickup_group=
sdp_session=Asterisk
dtls_verify=No
message_context=
mailboxes=
named_pickup_group=
record_on_feature=automixmon
dtls_private_key=
named_call_group=
t38_udptl_maxdatagram=0
media_encryption_optimistic=false
aors=DEADDEADBEEF
rpid_immediate=false
outbound_proxy=
identify_by=username
inband_progress=false
rtp_symmetric=false
transport=transport-udp
rtp_keepalive=0
t38_udptl_ec=none
fax_detect=false
t38_udptl_nat=false
allow_transfer=true
tos_video=0
srtp_tag_32=false
timers_min_se=90
call_group=
sub_min_expiry=0
100rel=yes
direct_media=true
rtp_timeout_hold=0
g726_non_standard=false
dtmf_mode=rfc4733
voicemail_extension=
rtp_timeout=0
dtls_cert_file=
media_encryption=no
media_use_received_transport=false
direct_media_glare_mitigation=none
trust_id_inbound=false
force_avp=false
record_off_feature=automixmon
send_diversion=true
language=
mwi_from_user=
rtp_ipv6=false
ice_support=false
callerid=unknown
aggregate_mwi=true
one_touch_recording=false
cos_video=0
accountcode=
allow=(g722|ulaw|alaw)
rewrite_contact=false
t38_udptl_ipv6=false
tone_zone=
user_eq_phone=false
allow_subscribe=true
rtp_engine=asterisk
auth=DEADDEADBEEF
from_user=DEADDEADBEEF
bind_rtp_to_media_address=false
disable_direct_media_on_nat=false
set_var=
use_ptime=false
outbound_auth=
media_address=
tos_audio=0
dtls_ca_path=
dtls_setup=active
force_rport=false
connected_line_method=invite
callerid_tag=
timers=yes
sdp_owner=-
trust_id_outbound=false
use_avpf=false
context=default
moh_suggest=default
send_pai=false
t38_udptl=false
dtls_ca_file=
callerid_privacy=allowed_not_screened
mwi_subscribe_replaces_unsolicited=false
cos_audio=0
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Re: [asterisk-users] CALLERID on pjsip doesn't work?

[Joshua Colp]

Andrew Ivins wrote:

Asterisk 13.8

Is CALLERID(all) supposed to wok for pjsip? When I do this:

exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
same => n,Dial(PJSIP/phone123, 30)


Your exten line has no priority, is that how it is in your dialplan?

If not you can isolate things a bit further by trying the following:

Set(CALLERID(all)=Jon Doe <+123456789>)

Or individually:

Set(CALLERID(name)=Jon Doe)
Set(CALLERID(num)=+123456789)

If those don't work I'd suggest showing the console and your 
configuration for the endpoint as it's something there instead.


Cheers,

--
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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] CALLERID on pjsip doesn't work?

[Andrew Ivins]

Asterisk 13.8

Is CALLERID(all) supposed to wok for pjsip? When I do this:

exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
same => n,Dial(PJSIP/phone123, 30)

I expect the callerid to be as set, but is always seems to be "phone123",
the name of the endpoint.

Andrew
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[asterisk-users] CALLERID(ani2) inserting

[CDR]

I checked
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information

But I cannot find a way to insert CALLERID(ani2), which I can read, but
when I try to set it for a new call, I get a runtime error.
This information, known as isup-oli comes embedded in the From header,like
this
sip:9087311878@64.45.157.104:5060;isup-oli=62;tag=sansay1724414rdb124
and it can be read by using
Set(var=${CALLERID(ani2)}
But how do we add that information to the outbound INVITE?  This is
critical in the toll-free industry and call-from-jail industries.
Thanks for your help.
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Re: [asterisk-users] CALLERID(ani2) inserting

[Ishfaq Malik]

Hi

According to this:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables

It is read only.

On 22 January 2015 at 16:22, CDR vene...@gmail.com wrote:

 I checked

 https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information

 But I cannot find a way to insert CALLERID(ani2), which I can read, but
 when I try to set it for a new call, I get a runtime error.
 This information, known as isup-oli comes embedded in the From header,like
 this
 sip:9087311878@64.45.157.104:5060;isup-oli=62;tag=sansay1724414rdb124
 and it can be read by using
 Set(var=${CALLERID(ani2)}
 But how do we add that information to the outbound INVITE?  This is
 critical in the toll-free industry and call-from-jail industries.
 Thanks for your help.


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-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] CALLERID(num) and CDR(clid) - originate

[Gabriel Ortiz Lour]

All right Matt,
thanks

2014-10-03 5:37 GMT-03:00 Matthew Jordan mjor...@digium.com:

 On Wed, Oct 1, 2014 at 8:00 AM, Gabriel Ortiz Lour
 ortiz.ad...@gmail.com wrote:
  Hello,
 
A question on channel originating (call files and AMI Originate):
 
How can I change the CALLERID(num) var (because of the E1 provider
 needs),
  but having another númber (the original one) stored on the clid CDR
 field
  on the database?

 You can't. The clid CDR field cannot be modified from the dialplan,
 and is always set to the caller ID of the channel. If you change the
 caller ID on the channel, you can expect the CDR clid field to reflect
 that.

 That being said, if you are using a flexible backend (such as
 cdr_custom or cdr_adaptive_odbc), you can add a custom column to your
 CDR records - such as 'clid_original' - and use the CDR function to
 set that value prior to changing the caller ID:

 exten = Set(CDR(clid_original)=${CALLERID(num)})
 exten = Set(CALLERID(num)=6575309)

 Matt

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

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Re: [asterisk-users] CALLERID(num) and CDR(clid) - originate

[Matthew Jordan]

On Wed, Oct 1, 2014 at 8:00 AM, Gabriel Ortiz Lour
ortiz.ad...@gmail.com wrote:
 Hello,

   A question on channel originating (call files and AMI Originate):

   How can I change the CALLERID(num) var (because of the E1 provider needs),
 but having another númber (the original one) stored on the clid CDR field
 on the database?

You can't. The clid CDR field cannot be modified from the dialplan,
and is always set to the caller ID of the channel. If you change the
caller ID on the channel, you can expect the CDR clid field to reflect
that.

That being said, if you are using a flexible backend (such as
cdr_custom or cdr_adaptive_odbc), you can add a custom column to your
CDR records - such as 'clid_original' - and use the CDR function to
set that value prior to changing the caller ID:

exten = Set(CDR(clid_original)=${CALLERID(num)})
exten = Set(CALLERID(num)=6575309)

Matt

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[asterisk-users] CALLERID(num) and CDR(clid) - originate

[Gabriel Ortiz Lour]

Hello,

  A question on channel originating (call files and AMI Originate):

  How can I change the CALLERID(num) var (because of the E1 provider
needs), but having another númber (the original one) stored on the clid
CDR field on the database?

  A channel agnostic solution would be the best one, without having to deal
with the problem based on what type of Tech used for the outgoing call.

Thanks already,
Gabriel
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Re: [asterisk-users] callerid overwrite

[motty cruz]

look like the issue continues, I am unable to overwrite callerid from
sip.conf in extensions.conf,

In sip.conf under
[general]
trustrpid = no  should i change it to yes?

Thanks




On Tue, Jan 28, 2014 at 1:06 PM, motty cruz motty.c...@gmail.com wrote:

 Thank you for your reply, I updated extensions.conf file to reflect your
 suggestion, I will monitor Asterisk for any more issues,

 Thanks,



 On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:

  On 1/28/14, 1:55 PM, motty cruz wrote:

  Hi all,
 I'm having issues with overwrite caller id, when I call someone my caller
 id should be mycompanyinc but instead my id shows up as my extension
 number 101.

  this is what i have in sip.conf
  [101]
 type=friend
 context=sipphones
 call-limit=99
 callerid=iuser 101
 disallow=all
 allow=ulaw
 allow=alaw
 username=101
 secret=Passwd
 dtmfmode=rfc2833
 host=dynamic
 mailbox=101@default
 nat=yes
 canreinvite=no


  this is what i have in extensions.conf
 [outbound]
 exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)

 This is how we have it and it works fine on Asterisk 1.8:
 Set(CALLERID(number)=insert your number here)

  exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
 exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
 exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)

  any ideas? as this happens random,





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Re: [asterisk-users] callerid overwrite

[Justin Hester]

Howdy,

Before changing any configuration I would highly recommend reading through
the entry in the sample file. Trust remote party ID may be set to 'no' for
a very good reason on your PBX, please take care to understand why it
should be changed before doing so.

Before digging into that though, what does the CLI tell you if you do a
NoOp() after having Set() the Caller ID function [1]?

[1]  Something like;

exten = _9NXX,1,Set(CALLERID(name)=mycompanyinc)
 same = n,NoOp(The caller ID has been set to ${CALLERID(name)})
 same = n,Dial(SIP/att/${EXTEN:1},80)

Hope this helps.

Justin Hester
Digium, Inc. · Technical Trainer
445 Jan Davis Drive NW · Huntsville, AL 35806 · USA
ph: +1 256 428 6238
Check us out at: http://digium.com · http://asterisk.org


On Thu, Jan 30, 2014 at 5:29 PM, motty cruz motty.c...@gmail.com wrote:

 look like the issue continues, I am unable to overwrite callerid from
 sip.conf in extensions.conf,

 In sip.conf under
 [general]
 trustrpid = no  should i change it to yes?

 Thanks




 On Tue, Jan 28, 2014 at 1:06 PM, motty cruz motty.c...@gmail.com wrote:

 Thank you for your reply, I updated extensions.conf file to reflect your
 suggestion, I will monitor Asterisk for any more issues,

 Thanks,



 On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:

  On 1/28/14, 1:55 PM, motty cruz wrote:

  Hi all,
 I'm having issues with overwrite caller id, when I call someone my
 caller id should be mycompanyinc but instead my id shows up as my
 extension number 101.

  this is what i have in sip.conf
  [101]
 type=friend
 context=sipphones
 call-limit=99
 callerid=iuser 101
 disallow=all
 allow=ulaw
 allow=alaw
 username=101
 secret=Passwd
 dtmfmode=rfc2833
 host=dynamic
 mailbox=101@default
 nat=yes
 canreinvite=no


  this is what i have in extensions.conf
 [outbound]
 exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)

 This is how we have it and it works fine on Asterisk 1.8:
 Set(CALLERID(number)=insert your number here)

  exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
 exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
 exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)

  any ideas? as this happens random,





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[asterisk-users] callerid overwrite

[motty cruz]

Hi all,
I'm having issues with overwrite caller id, when I call someone my caller
id should be mycompanyinc but instead my id shows up as my extension
number 101.

this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
allow=alaw
username=101
secret=Passwd
dtmfmode=rfc2833
host=dynamic
mailbox=101@default
nat=yes
canreinvite=no


this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)

any ideas? as this happens random,
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Re: [asterisk-users] callerid overwrite

[Andres]

On 1/28/14, 1:55 PM, motty cruz wrote:

Hi all,
I'm having issues with overwrite caller id, when I call someone my 
caller id should be mycompanyinc but instead my id shows up as my 
extension number 101.


this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
callerid=iuser 101
disallow=all
allow=ulaw
allow=alaw
username=101
secret=Passwd
dtmfmode=rfc2833
host=dynamic
mailbox=101@default
nat=yes
canreinvite=no


this is what i have in extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)

This is how we have it and it works fine on Asterisk 1.8:
Set(CALLERID(number)=insert your number here)

exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)

any ideas? as this happens random,






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Re: [asterisk-users] callerid overwrite

[Chad Wallace]

On Tue, 28 Jan 2014 10:55:58 -0800
motty cruz motty.c...@gmail.com wrote:

 this is what i have in extensions.conf
 [outbound]
 exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)
 exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
 exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
 exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)
 
 any ideas? as this happens random,

You're setting CALLERID(num) to a name.  Use CALLERID(name) instead.
Additionally, you might want to set CALLERID(num) to your DID.

You can do both name and number at the same time by using
CALLERID(all), something like this:

exten = _91NXXNXX,1,Set(CALLERID(all)=mycompanyinc123-456-7890)


-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0


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Re: [asterisk-users] callerid overwrite

[motty cruz]

Thank you for your reply, I updated extensions.conf file to reflect your
suggestion, I will monitor Asterisk for any more issues,

Thanks,



On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote:

  On 1/28/14, 1:55 PM, motty cruz wrote:

  Hi all,
 I'm having issues with overwrite caller id, when I call someone my caller
 id should be mycompanyinc but instead my id shows up as my extension
 number 101.

  this is what i have in sip.conf
  [101]
 type=friend
 context=sipphones
 call-limit=99
 callerid=iuser 101
 disallow=all
 allow=ulaw
 allow=alaw
 username=101
 secret=Passwd
 dtmfmode=rfc2833
 host=dynamic
 mailbox=101@default
 nat=yes
 canreinvite=no


  this is what i have in extensions.conf
 [outbound]
 exten = _91NXXNXX,1,Set(CALLERID(num)=mycompanyinc)

 This is how we have it and it works fine on Asterisk 1.8:
 Set(CALLERID(number)=insert your number here)

  exten = _91NXXNXX,2,Dial(SIP/att/${EXTEN:1},80)
 exten = _9NXX,1,Set(CALLERID(num)=mycompanyinc)
 exten = _9NXX,2,Dial(SIP/att/${EXTEN:1},80)

  any ideas? as this happens random,





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[asterisk-users] CallerID settings

[Gabriel Ortiz Lour]

Hi all,

  What should I do when my E1/SIP provider need a specific callerid(num)
setting for my outgoing calls?
  My problem is that if I use Set(CALLERID(num)=XXYY) then I got XXYY on
the cdr src field, losing info of the real src of the call.
  What is the best way out? (Preferably tech independent)

Thanks,
Gabriel
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Re: [asterisk-users] CallerID settings

[jg]

For outgoing calls you can write additional information into the userfield,  or you can define 
your own additional fields using an adaptive-odbc setup. For ISDN and POTS channels you can 
typically set the callerid (just the number) for outgoing calls only to those numbers given to 
you by your telco (or they pick a default number). There are exceptions, but not for mere mortals.


jg

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Re: [asterisk-users] CallerID external call after Attended Transfer

[Karsten Wemheuer]

Hi,

Am Montag, den 04.02.2013, 14:45 +0100 schrieb Jonas Kellens:
 Hello,
 thanks you for your answer.
 The IP-phones in this case are Yealink T32G.
 What setting is needed in this IP-phone ?

as Kevin already written, set this in asterisk:
sendrpid=pai 
trustrpid=yes 

I don't know the T32G, but in T2x series there is a setting under
Accounts-Advanced called Caller ID Header. Select PAI or PAI
+FROM. The default is FROM which won't work.

HTH,

Karsten



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[asterisk-users] CallerID external call after Attended Transfer

[Jonas Kellens]

Hello,

using Asterisk 1.8.12.2

case :

I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended transfer 
to my colleague ( A )
Colleague answers and the receptionist tells him that I am on the other 
side.

Receptionist transfers the call and I am connected to my colleague ( B )


My question is about the CallerID that the colleague sees on his IP-phone.

In step A the colleague sees the CallerID of the receptionist, which I 
normal.
In step B, after I am connected to my colleague, the colleague still 
sees the CallerID of the receptionist (and not my cellphone number).


How come my colleague does not see my cellphone number ? What is the 
correct setting ( IP-phone ? Asterisk ? ) to obtain this functionality.



Thanks.
Jonas.
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Re: [asterisk-users] CallerID external call after Attended Transfer

[Steven Howes]

On 4 Feb 2013, at 12:53, Jonas Kellens wrote:
 I call with my cellphone to our public telephone number
 Our receptionist answers the incoming call and does an attended transfer to 
 my colleague ( A )
 Colleague answers and the receptionist tells him that I am on the other side.
 Receptionist transfers the call and I am connected to my colleague ( B )
 
 
 My question is about the CallerID that the colleague sees on his IP-phone.
 
 In step A the colleague sees the CallerID of the receptionist, which I normal.
 In step B, after I am connected to my colleague, the colleague still sees the 
 CallerID of the receptionist (and not my cellphone number).
 
 How come my colleague does not see my cellphone number ? What is the correct 
 setting ( IP-phone ? Asterisk ? ) to obtain this functionality.

It's called connected line ID (it sends clid updates when things change). 
Asterisk supports it in recent versions (i believe 1.8 is sufficient) - your 
handsets may or may not (their method of transfer, and their ability to process 
the updates can affect it's workability). Given you've not mentioned your 
handsets, we cant make that judgement for you.

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Re: [asterisk-users] CallerID external call after Attended Transfer

[Jonas Kellens]

Hello,

thanks you for your answer.

The IP-phones in this case are Yealink T32G.

What setting is needed in this IP-phone ?


Jonas.

On 02/04/2013 02:29 PM, Steven Howes wrote:

On 4 Feb 2013, at 12:53, Jonas Kellens wrote:

I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended 
transfer to my colleague ( A )
Colleague answers and the receptionist tells him that I am on the 
other side.

Receptionist transfers the call and I am connected to my colleague ( B )


My question is about the CallerID that the colleague sees on his 
IP-phone.


In step A the colleague sees the CallerID of the receptionist, which 
I normal.
In step B, after I am connected to my colleague, the colleague still 
sees the CallerID of the receptionist (and not my cellphone number).


How come my colleague does not see my cellphone number ? What is the 
correct setting ( IP-phone ? Asterisk ? ) to obtain this functionality.


It's called connected line ID (it sends clid updates when things 
change). Asterisk supports it in recent versions (i believe 1.8 is 
sufficient) - your handsets may or may not (their method of transfer, 
and their ability to process the updates can affect it's workability). 
Given you've not mentioned your handsets, we cant make that judgement 
for you.


Steve


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Re: [asterisk-users] CallerID external call after Attended Transfer

[Steven Howes]

On 4 Feb 2013, at 13:45, Jonas Kellens wrote:
 The IP-phones in this case are Yealink T32G.
 
 What setting is needed in this IP-phone ?

Quick google doesn't turn up any results. Handsets probably dont support it.

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Re: [asterisk-users] CallerID external call after Attended Transfer

[Jonas Kellens]

Hello,

and is there any setting in Asterisk to turn this functionality on/off ? 
Maybe mine is not enabled.



Jonas


On 02/04/2013 03:30 PM, Steven Howes wrote:

On 4 Feb 2013, at 13:45, Jonas Kellens wrote:

The IP-phones in this case are Yealink T32G.

What setting is needed in this IP-phone ?


Quick google doesn't turn up any results. Handsets probably dont 
support it.


Steve


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Re: [asterisk-users] CallerID external call after Attended Transfer

[Kevin Larsen]

One thing you can try is to set the following in your sip.conf.

sendrpid=pai
trustrpid=yes

You can put that on individual phone configurations in sip.conf or, as I 
do, in a template that is applied to a set of phones.

I believe that was what I had to set so that the remote caller ID would 
show up properly on my Polycom phones. I made no changes to the Polycom 
configuration to make it work. It might work with the Yealink T32G phones 
as well. 

In the case originally presented, I get the following:

Call comes into Operator showing cell phone caller id. Operator performs 
an attended transfer. I get the Operator caller ID. Upon completion of the 
transfer, I get the cell phone caller ID. If a blind transfer is 
performed, I get the cell phone caller ID (there might be a flash of the 
operators caller ID for just the split second it takes her to hit the 
transfer button a second time to turn it from attended to blind transfer 
on my phones). 

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



From:   Steven Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com, 
Date:   02/04/2013 08:31 AM
Subject:Re: [asterisk-users] CallerID external call after Attended 
Transfer
Sent by:asterisk-users-boun...@lists.digium.com



On 4 Feb 2013, at 13:45, Jonas Kellens wrote:
The IP-phones in this case are Yealink T32G.

What setting is needed in this IP-phone ?

Quick google doesn't turn up any results. Handsets probably dont support 
it.

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Re: [asterisk-users] CallerID external call after Attended Transfer

[Frank]

What is the PAI option below that you are talking about, for sendrpid ?
The manual only says that yes or no can be used..


On 2/4/13 9:39 AM, Kevin Larsen wrote:

One thing you can try is to set the following in your sip.conf.

sendrpid=pai
trustrpid=yes

You can put that on individual phone configurations in sip.conf or, as I
do, in a template that is applied to a set of phones.

I believe that was what I had to set so that the remote caller ID would
show up properly on my Polycom phones. I made no changes to the Polycom
configuration to make it work. It might work with the Yealink T32G
phones as well.

In the case originally presented, I get the following:

Call comes into Operator showing cell phone caller id. Operator performs
an attended transfer. I get the Operator caller ID. Upon completion of
the transfer, I get the cell phone caller ID. If a blind transfer is
performed, I get the cell phone caller ID (there might be a flash of the
operators caller ID for just the split second it takes her to hit the
transfer button a second time to turn it from attended to blind transfer
on my phones).

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



From: Steven Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 02/04/2013 08:31 AM
Subject: Re: [asterisk-users] CallerID external call after Attended
Transfer
Sent by: asterisk-users-boun...@lists.digium.com




On 4 Feb 2013, at 13:45, Jonas Kellens wrote:
The IP-phones in this case are Yealink T32G.

What setting is needed in this IP-phone ?

Quick google doesn't turn up any results. Handsets probably dont support
it.

Steve--
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Re: [asterisk-users] CallerID external call after Attended Transfer

[Kevin Larsen]

According to the default sip.conf file:

sendrpid=yes ; If Remote-Party-ID should be sent (defaults to no)

sendrpid=rpid ; Use the Remote-Party-ID header to send the identity of 
the remote party. This is identical to sendrpid=yes

sendrpid=pai ; Use the P-Asserted-Identity header to send the identity 
of the remote party.

In my case, pai works. I could also see yes or the equivalent rpid also 
working depending on what the phone expects. I have to think that the 
reason the options are there is because different endpoints behave 
differently.

I believe the pai option was added in 1.8.

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



From:   Frank fr...@efirehouse.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com, 
Date:   02/04/2013 09:47 AM
Subject:Re: [asterisk-users] CallerID external call after Attended 
Transfer
Sent by:asterisk-users-boun...@lists.digium.com



What is the PAI option below that you are talking about, for sendrpid ?
The manual only says that yes or no can be used..


On 2/4/13 9:39 AM, Kevin Larsen wrote:
 One thing you can try is to set the following in your sip.conf.

 sendrpid=pai
 trustrpid=yes

 You can put that on individual phone configurations in sip.conf or, as I
 do, in a template that is applied to a set of phones.

 I believe that was what I had to set so that the remote caller ID would
 show up properly on my Polycom phones. I made no changes to the Polycom
 configuration to make it work. It might work with the Yealink T32G
 phones as well.

 In the case originally presented, I get the following:

 Call comes into Operator showing cell phone caller id. Operator performs
 an attended transfer. I get the Operator caller ID. Upon completion of
 the transfer, I get the cell phone caller ID. If a blind transfer is
 performed, I get the cell phone caller ID (there might be a flash of the
 operators caller ID for just the split second it takes her to hit the
 transfer button a second time to turn it from attended to blind transfer
 on my phones).

 Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



 From: Steven Howes steve-li...@geekinter.net
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com,
 Date: 02/04/2013 08:31 AM
 Subject: Re: [asterisk-users] CallerID external call after Attended
 Transfer
 Sent by: asterisk-users-boun...@lists.digium.com
 



 On 4 Feb 2013, at 13:45, Jonas Kellens wrote:
 The IP-phones in this case are Yealink T32G.

 What setting is needed in this IP-phone ?

 Quick google doesn't turn up any results. Handsets probably dont support
 it.

 Steve--
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 asterisk-users mailing list
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Re: [asterisk-users] callerid not received from dahdi

[Shaun Ruffell]

On Tue, Dec 11, 2012 at 10:07:42AM +0530, Harish Mandowara wrote:
 Hi,
 
 Thank you for your reply.
 77 ext. number is connected with my asterisk. so any one want to
 talk with jitsi(pc), they have to dial 77 then 2000#(jitsi sip
 user number).
 
  my pbx is sending callerid. i can see on other analog phone display.
 
 Yes pbx is sending callerid. When i dial any ext. number from
 jitsi. On the recipient phone display shows 77 ext number.
 
 i tried all combination from
 https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India
 
 but it does not work.

Does the manual for your PBX say anything about how it's
sending the CallerID information?

Knowing specifically how the CallerID is supposed to be sent can
allow us to more quickly narrow in on why you may not be detecting
it.

Also, in the configs you originally posted you had #include
dahdi-channels.conf. Would it be possible to send that configuration
options you have attached in that file as well?

Cheers,
Shaun

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] callerid not received from dahdi

[Christopher Harrington]

From the last time you sent this to the list, here's the response from Richard
Mudgett rmudg...@digium.com...

 my scenario is below

 analog phone (10 to 99)-- pbx--(77)asterisk
 jitsi(2000)

 i have analog telephone interface numbered 77 attached with asterisk
 and
 other sip user is 2000 on jitsi.

 I can call from any number from 10 to 99(in intercom) on 77 and ivr
 response will come then i can typed 2000# and call go to 2000 named
 user
 in asterisk.

 Now my problem is when i am calling from 10 to 99 (any number) this
 number
 should display to sip 2000's user. But its not showing to user. Its
 shows
 asterisk@my_asterisk_server_ip.

 my config. as follow

 extension.conf

 exten = s,1,Goto(phrase-menu,s,1)

 [phrase-menu]

 exten = s,1,Answer()
 exten = s,2,Wait(1)
 exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
 exten = s,4,Wait(2)
 exten = s,5,Set(CALLERID(num,CID)=${CALLERID})

Remove the CID option.  It does nothing in this case because
it does not apply.  The CID option here only applies to reading
not writing.  Please re-read the documentation for CALLERID().

 exten = s,6,Dial(SIP/${PHRASEID},40,tT)
 exten = h,1,Hangup()


 and in chan_dahdi.conf

 ; General options
 [channels]
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 threewaycalling=yes
 transfer=yes
 echocancel=yes
 echocancelwhenbridged=yes

 cidsignalling=dtmf
 cidstart=polarity
 callerid=asreceived

 rxgain=0.0
 txgain=0.0
 ;FXO Modules
 group=1
 echocancel=yes
 signalling=fxs_ks
 context=default
 channel=1-20

 #include dahdi-channels.conf

From your description, the link between the pbx and (77)asterisk
is analog.  Analog can only pass caller id information in one
direction.  It looks like you have it setup to pass caller id
from the pbx to (77)asterisk.  Is the pbx even sending caller id?
Is it sending it in the form you have configured in Asterisk?
(dtmf, polarity start, dtmfcidlevel=???)


On Sun, Dec 9, 2012 at 11:42 PM, Harish Mandowara 
asteriskhelp2...@gmail.com wrote:

 my scenario is below

 analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)

 i have analog telephone interface numbered 77 attached with asterisk and
 other sip user is 2000 on jitsi.

 I can call from any number from 10 to 99(in intercom) on 77 and ivr
 response will come then i can typed 2000# and call go to 2000 named user
 in asterisk.

 Now my problem is when i am calling from 10 to 99 (any number) this number
 should display to sip 2000's user. But its not showing to user. Its 
 showsasterisk@my_asterisk_server_ip 
 https://webmail.cdac.in/twig/index.php?s[mailbox]=mail%2Fsent-mails[mailGroup]=%2As[mail_startmsg]=1s[sortby]=dates[sortbyway]=1s[delete-return]=msgviews[mailtree]=0%7Cc[f]=mailc[a]=composeform[to]=asterisk@my_asterisk_server_ip.

 my config. as follow

 extension.conf

 exten = s,1,Goto(phrase-menu,s,1)

 [phrase-menu]

 exten = s,1,Answer()
 exten = s,2,Wait(1)
 exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
 exten = s,4,Wait(2)
 exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
 exten = s,6,Dial(SIP/${PHRASEID},40,tT)
 exten = h,1,Hangup()


 and in chan_dahdi.conf

 ; General options
 [channels]
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 threewaycalling=yes
 transfer=yes
 echocancel=yes
 echocancelwhenbridged=yes
 cidsignalling=dtmf
 cidstart=polarity
 callerid=asreceived
 rxgain=0.0
 txgain=0.0
 ;FXO Modules
 group=1
 echocancel=yes
 signalling=fxs_ks
 context=default
 channel=1-20

 #include dahdi-channels.conf


 any help

 thanks..


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-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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Re: [asterisk-users] callerid not received from dahdi

[Harish Mandowara]

Hi,

Thank you for your reply.
77 ext. number is connected with my asterisk. so any one want to talk with
jitsi(pc), they have to dial 77 then 2000#(jitsi sip user number).

 my pbx is sending callerid. i can see on other analog phone display.

Yes pbx is sending callerid. When i dial any ext. number from jitsi. On the
recipient phone display shows 77 ext number.

i tried all combination from
https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India

but it does not work.


any help



On Mon, Dec 10, 2012 at 9:39 PM, Christopher Harrington ch...@acsdi.comwrote:

 From the last time you sent this to the list, here's the response from Richard
 Mudgett rmudg...@digium.com...

  my scenario is below
 
  analog phone (10 to 99)-- pbx--(77)asterisk
  jitsi(2000)
 
  i have analog telephone interface numbered 77 attached with asterisk
  and
  other sip user is 2000 on jitsi.
 
  I can call from any number from 10 to 99(in intercom) on 77 and ivr
  response will come then i can typed 2000# and call go to 2000 named
  user
  in asterisk.
 
  Now my problem is when i am calling from 10 to 99 (any number) this
  number
  should display to sip 2000's user. But its not showing to user. Its
  shows
  asterisk@my_asterisk_server_ip.
 
  my config. as follow
 
  extension.conf
 
  exten = s,1,Goto(phrase-menu,s,1)
 
  [phrase-menu]
 
  exten = s,1,Answer()
  exten = s,2,Wait(1)
  exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
  exten = s,4,Wait(2)
  exten = s,5,Set(CALLERID(num,CID)=${CALLERID})

 Remove the CID option.  It does nothing in this case because
 it does not apply.  The CID option here only applies to reading
 not writing.  Please re-read the documentation for CALLERID().


  exten = s,6,Dial(SIP/${PHRASEID},40,tT)
  exten = h,1,Hangup()
 
 
  and in chan_dahdi.conf
 
  ; General options
  [channels]
  usecallerid=yes
  hidecallerid=no
  callwaiting=yes
  threewaycalling=yes
  transfer=yes
  echocancel=yes
  echocancelwhenbridged=yes

  cidsignalling=dtmf
  cidstart=polarity
  callerid=asreceived

  rxgain=0.0
  txgain=0.0
  ;FXO Modules
  group=1
  echocancel=yes
  signalling=fxs_ks
  context=default
  channel=1-20
 
  #include dahdi-channels.conf

 From your description, the link between the pbx and (77)asterisk
 is analog.  Analog can only pass caller id information in one
 direction.  It looks like you have it setup to pass caller id
 from the pbx to (77)asterisk.  Is the pbx even sending caller id?
 Is it sending it in the form you have configured in Asterisk?
 (dtmf, polarity start, dtmfcidlevel=???)


 On Sun, Dec 9, 2012 at 11:42 PM, Harish Mandowara 
 asteriskhelp2...@gmail.com wrote:

 my scenario is below

 analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)

 i have analog telephone interface numbered 77 attached with asterisk and
 other sip user is 2000 on jitsi.

 I can call from any number from 10 to 99(in intercom) on 77 and ivr
 response will come then i can typed 2000# and call go to 2000 named user
 in asterisk.

 Now my problem is when i am calling from 10 to 99 (any number) this number
 should display to sip 2000's user. But its not showing to user. Its 
 showsasterisk@my_asterisk_server_ip 
 https://webmail.cdac.in/twig/index.php?s[mailbox]=mail%2Fsent-mails[mailGroup]=%2As[mail_startmsg]=1s[sortby]=dates[sortbyway]=1s[delete-return]=msgviews[mailtree]=0%7Cc[f]=mailc[a]=composeform[to]=asterisk@my_asterisk_server_ip.

 my config. as follow

 extension.conf

 exten = s,1,Goto(phrase-menu,s,1)

 [phrase-menu]

 exten = s,1,Answer()
 exten = s,2,Wait(1)
 exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
 exten = s,4,Wait(2)
 exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
 exten = s,6,Dial(SIP/${PHRASEID},40,tT)
 exten = h,1,Hangup()


 and in chan_dahdi.conf

 ; General options
 [channels]
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 threewaycalling=yes
 transfer=yes
 echocancel=yes
 echocancelwhenbridged=yes
 cidsignalling=dtmf
 cidstart=polarity
 callerid=asreceived
 rxgain=0.0
 txgain=0.0
 ;FXO Modules
 group=1
 echocancel=yes
 signalling=fxs_ks
 context=default
 channel=1-20

 #include dahdi-channels.conf


 any help

 thanks..


 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
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 --
 -Chris Harrington
 ACSDi Office: 763.559.5800
 Mobile Phone: 612.326.4248



 --
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 To UNSUBSCRIBE or 

[asterisk-users] callerid not received from dahdi

[Harish Mandowara]

my scenario is below

analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)

i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.

I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named user
in asterisk.

Now my problem is when i am calling from 10 to 99 (any number) this number
should display to sip 2000's user. But its not showing to user. Its
showsasterisk@my_asterisk_server_ip
https://webmail.cdac.in/twig/index.php?s[mailbox]=mail%2Fsent-mails[mailGroup]=%2As[mail_startmsg]=1s[sortby]=dates[sortbyway]=1s[delete-return]=msgviews[mailtree]=0%7Cc[f]=mailc[a]=composeform[to]=asterisk@my_asterisk_server_ip.

my config. as follow

extension.conf

exten = s,1,Goto(phrase-menu,s,1)

[phrase-menu]

exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()


and in chan_dahdi.conf

; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20

#include dahdi-channels.conf


any help

thanks..
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[asterisk-users] callerid not received from dahdi

[Harish Mandowara]

Hi,

my scenario is below

analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)

i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.

I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named user
in asterisk.

Now my problem is when i am calling from 10 to 99 (any number) this number
should display to sip 2000's user. But its not showing to user. Its shows
asterisk@my_asterisk_server_ip.

my config. as follow

extension.conf

exten = s,1,Goto(phrase-menu,s,1)

[phrase-menu]

exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)  
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()


and in chan_dahdi.conf

; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20

#include dahdi-channels.conf


any help

thanks..

Do not bother about below message. That is auto-generated by my mail
server.

-- 
With Warm Regards

Harish Mandowara




---

This e-mail is for the sole use of the intended recipient(s) and may
contain confidential and privileged information. If you are not the
intended recipient, please contact the sender by reply e-mail and destroy
all copies and the original message. Any unauthorized review, use,
disclosure, dissemination, forwarding, printing or copying of this email
is strictly prohibited and appropriate legal action will be taken.
---


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Re: [asterisk-users] callerid not received from dahdi

[Shaun Ruffell]

On Fri, Nov 30, 2012 at 04:54:28PM +0530, Harish Mandowara wrote:
 
 Do not bother about below message. That is auto-generated by my mail
 server.

[snip]

 ---
 
 This e-mail is for the sole use of the intended recipient(s) and may
 contain confidential and privileged information. If you are not the
 intended recipient, please contact the sender by reply e-mail and destroy
 all copies and the original message. Any unauthorized review, use,
 disclosure, dissemination, forwarding, printing or copying of this email
 is strictly prohibited and appropriate legal action will be taken.
 ---

I realize this probably seems silly, but I do not think it's
in my best interest to ignore threats of appropriate legal action
for forwarding this email to someone who might be able to help or
archiving on a message board, etc..

Do you think you could talk to the people who manage your mail
server and have the disclaimer removed?  They may be interested in:

http://www.goldmark.org/jeff/stupid-disclaimers

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] callerid not received from dahdi

[Richard Mudgett]

 my scenario is below
 
 analog phone (10 to 99)-- pbx--(77)asterisk
 jitsi(2000)
 
 i have analog telephone interface numbered 77 attached with asterisk
 and
 other sip user is 2000 on jitsi.
 
 I can call from any number from 10 to 99(in intercom) on 77 and ivr
 response will come then i can typed 2000# and call go to 2000 named
 user
 in asterisk.
 
 Now my problem is when i am calling from 10 to 99 (any number) this
 number
 should display to sip 2000's user. But its not showing to user. Its
 shows
 asterisk@my_asterisk_server_ip.
 
 my config. as follow
 
 extension.conf
 
 exten = s,1,Goto(phrase-menu,s,1)
 
 [phrase-menu]
 
 exten = s,1,Answer()
 exten = s,2,Wait(1)
 exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
 exten = s,4,Wait(2)
 exten = s,5,Set(CALLERID(num,CID)=${CALLERID})

Remove the CID option.  It does nothing in this case because
it does not apply.  The CID option here only applies to reading
not writing.  Please re-read the documentation for CALLERID().

 exten = s,6,Dial(SIP/${PHRASEID},40,tT)
 exten = h,1,Hangup()
 
 
 and in chan_dahdi.conf
 
 ; General options
 [channels]
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 threewaycalling=yes
 transfer=yes
 echocancel=yes
 echocancelwhenbridged=yes

 cidsignalling=dtmf
 cidstart=polarity
 callerid=asreceived

 rxgain=0.0
 txgain=0.0
 ;FXO Modules
 group=1
 echocancel=yes
 signalling=fxs_ks
 context=default
 channel=1-20
 
 #include dahdi-channels.conf

From your description, the link between the pbx and (77)asterisk
is analog.  Analog can only pass caller id information in one
direction.  It looks like you have it setup to pass caller id
from the pbx to (77)asterisk.  Is the pbx even sending caller id?
Is it sending it in the form you have configured in Asterisk?
(dtmf, polarity start, dtmfcidlevel=???)

Richard

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[asterisk-users] callerid channel variable vs. CDR(src)?

[Stefan at WPF]

Actually, is there a difference between the callerid channel variable and
CDR(src)? Is CDR(src) actually set to the callerid channel variable? Thanks
:-)
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[asterisk-users] CallerID

[Jerry Geis]

When I use a call file to start a call I set the
CallerID: field and the polycom phone shows the correct information.

When I use a call file to start a conf call I set the
CallerID: field and my polycom phones show asterisk not the callerID I 
have set.


Is there something additional needed to set the CallerID in this case?
I am also using the Local channel does that make a difference?

Jerry

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[asterisk-users] CallerId back to incoming

[Stephen Collier]

I'm currently doing some testing with Asterisk ( 1.8.11.0) on RHEL6
using realtime for sippeers, sipusers and musiconhold

I have  Avaya definity - PRI E1 - Asterisk 1 - IAX2  - Asterisk
2 

I have peers (sip) snom 821s on both Asterisk 1 and 2 all calls working
between all systems.

CallerID from Asterisk to Avaya is working correctly.

The problem is a caller from Avaya to Asterisk displays correctly the
CID of the Asterisk Extension to the calling party on the Avaya but only
if the peer is on Asterisk 1. If the peer is on Asterisk 2 only the CID
of the PRI on the avaya side is displayed. I hope this makes sense.

I'm not sure where to start looking or whether its even possible.

I can of course supply any of the configs that may help.

Cheers

Stephen Collier

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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

Hello,

Revisiting this old thread, following Richard's suggestion, I modified
Asterisk config so that it would set RDNIS for every forwarded call.

I kept at hand, the results gathered in another test session :
the output of a successful call (with appropriate CallerID) and the
output of an unsuccessful one.

2011/11/8 Olivier oza_4...@yahoo.fr

 Hi,

 As promised, here is a follow up on my quest to get CallerID correctly
 presented when forwarding calls to cellphones.

 Here is a reminder of the issue at hand:

 Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension) which
 forwards to Cory (GSM handset)
 What I would like to get is to see Alice's number (not Bob's number)
 presented to Cory.
 Sometimes, I get Alice's number, sometimes, I get Bob's number (new
 findings from last sunday trials).
 And of course, if Daniel or Eric would call Bob, the CallerID number
 presented to Cory would either be Daniel's number, Eric's number or Bob's
 number depending on a root cause I'm looking after for several days now.



 To check if CallerID is filtered or controlled by Telco, I originated
 calls from Asterisk using hand crafted caller ids: any CallerID was
 correctly presented.
 So I originally thought the root cause I'm after is a telco equipment
 switching ANI and CID.
 But a close look at some last trials output makes me asking for opinions
 from this list readers.

 Here follows, the anonymized (and hand indented) output of command PRI
 debug command.
 I focused on the end of call setup dialog.

 For the successfully presented call, the output is:
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [6c 0b 21 83 37 38 36 XX
 XX XX XX XX XX]
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Calling Number (len=13) [
 Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan
 (E.164/E.163) (1)
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c: 
 Presentation: Presentation allowed of network provided number (3)
 '78649' ]
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [70 0b 80 30 36 37 31 XX
 XX XX XX XX XX]
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Called Number (len=13) [
 Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
 '067100' ]
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [74 0e 21 01 8f 33 33 33
 34 34 XX XX XX XX XX XX]
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
 (len=16) [ Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering
 Plan (E.164/E.163) (1)
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:
Ext: 0  Presentation: Presentation
 permitted, user number passed network screening (1)
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:
Ext: 1  Reason: Forwarded unconditionally
 (15)
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:   '3334436' ]
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [a1]
 [Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Sending Complete (len= 1)


 For the unsuccessfully presented call, the output is:
 [Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [6c 0b 21 83 36 37 38 XX
 XX XX XX XX XX]
 [Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Calling Number (len=13) [
 Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan
 (E.164/E.163) (1)
 [Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c: 
 Presentation: Presentation allowed of network provided number (3)
 '67854' ]
 [Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [70 0b 80 30 36 37 31 XX
 XX XX XX XX XX]
 [Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Called Number (len=13) [
 Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
 '067100' ]
 [Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [a1]
 [Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Sending Complete (len= 1)


 Am I correctly interpreting when saying that in the successful call,
 Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX XX]
 message which is not otherwise sent ?
 What can explains this difference ?
 Is this something I can (should) control ?

 For reference:
 dahdi show version
 DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC
 pri show version
 libpri version: 1.4.10.2



 Regards



From another unsuccessful try, I got the following (anonymized) output:
[Dec  3 09:21:32] VERBOSE[6201] chan_dahdi.c:  Calling Number (len=13) [
Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1)
[Dec  3 09:21:32] VERBOSE[6201] chan_dahdi.c: 
Presentation: Presentation allowed of network provided number (3)
'95135' ]
[Dec  3 09:21:32] VERBOSE[6201] chan_dahdi.c:  [70 0b 80 30 36 37 31 30 XX
XX XX XX XX]
[Dec  3 09:21:32] VERBOSE[6201] chan_dahdi.c:  Called Number (len=13) [
Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
'06710X' ]
[Dec  3 09:21:32] VERBOSE[6201] chan_dahdi.c:  [74 0e 21 01 8f 33 33 33 34
34 33 XX XX XX XX XX]
[Dec  3 09:21:32] VERBOSE[6201] chan_dahdi.c:  Redirecting Number (len=16)
[ Ext: 0  

Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[giovanni.v]

Il 09/11/2011 17.37, Richard Mudgett ha scritto:

You would then use
the DAHDISendCallreroutingFacility application*before*  you answer
the call to forward/deflect the incoming call back to the network.


I think Answer makes no sense at all because the network will redirect 
then continue to cancel the preceding call setup.


ETSI Call Deflection/Rerouting message flow:

  Deflection
CallerCarrier   PBX   destination
  |---SETUP-|  '  '
 |SETUP|  '
 |---CALL PROCEEDING---|  '
 |---FACILITY+ |  '
 |DISCONNECT---|  '
 |---RELEASE---|  '
 |RELEASE COMPLETE-|  '
 |SETUP---|


This will also require this settings on the dahdi channel?
facilityenable=yes
transfer=yes
??

Also, according to this document http://www.asterisk.org/node/48611 
seems that DAHDISendCallreroutingFacility() doesn't return anything so 
how we can test if the command was successful or refused?


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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Richard Mudgett]

   As promised, here is a follow up on my quest to get CallerID
   correctly
   presented when forwarding calls to cellphones.
  
   Here is a reminder of the issue at hand:
  
   Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
   which forwards to Cory (GSM handset)
   What I would like to get is to see Alice's number (not Bob's
   number)
   presented to Cory.
   Sometimes, I get Alice's number, sometimes, I get Bob's number
   (new
   findings from last sunday trials).
   And of course, if Daniel or Eric would call Bob, the CallerID
   number
   presented to Cory would either be Daniel's number, Eric's number
   or
   Bob's number depending on a root cause I'm looking after for
   several
   days now.
  
  
  
   To check if CallerID is filtered or controlled by Telco, I
   originated
   calls from Asterisk using hand crafted caller ids: any CallerID
   was
   correctly presented.
   So I originally thought the root cause I'm after is a telco
   equipment
   switching ANI and CID.
   But a close look at some last trials output makes me asking for
   opinions from this list readers.
  
   Here follows, the anonymized (and hand indented) output of command
   PRI
   debug command.
   I focused on the end of call setup dialog.
  
   For the successfully presented call, the output is:
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [6c 0b 21 83 37 38
   36
   XX XX XX XX XX XX]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Calling Number
   (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
   Numbering Plan (E.164/E.163) (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Presentation:
   Presentation allowed of network provided number (3) '78649' ]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [70 0b 80 30 36 37
   31
   XX XX XX XX XX XX]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Called Number
   (len=13)
   [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
   '067100' ]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [74 0e 21 01 8f 33
   33
   33 34 34 XX XX XX XX XX XX]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
   (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
   Numbering Plan (E.164/E.163) (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0
   Presentation:
   Presentation permitted, user number passed network screening (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
   Forwarded unconditionally (15)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [a1]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Sending Complete
   (len=
   1)
  
  
   For the unsuccessfully presented call, the output is:
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [6c 0b 21 83 36 37
   38
   XX XX XX XX XX XX]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Calling Number
   (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
   Numbering Plan (E.164/E.163) (1)
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Presentation:
   Presentation allowed of network provided number (3) '67854' ]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [70 0b 80 30 36 37
   31
   XX XX XX XX XX XX]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Called Number
   (len=13)
   [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
   '067100' ]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [a1]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Sending Complete
   (len=
   1)
  
  
   Am I correctly interpreting when saying that in the successful
   call,
   Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX
   XX
   XX] message which is not otherwise sent ?
   What can explains this difference ?
   Is this something I can (should) control ?
  
   For reference:
   dahdi show version
   DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC
   pri show version
   libpri version: 1.4.10.2
 
  Improved support for manipulation of redirecting number is available
  with the REDIRECTING dialplan function in Asterisk v1.8.x and
  libpri v1.4.12. Prior to Asterisk v1.8.x you only have
  CALLERID(RDNIS).
 
  https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
 
 
  Richard
 
 
  Hi Richard,
 
  1. Could you elaborate a bit ?
  Do you imply that the lines bellow were present (or missing) because
  I
  did somewhere set CALLERID(RDNIS) and that I should use them ?
 
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
   (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
   Numbering Plan (E.164/E.163) (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0
   Presentation:
   Presentation permitted, user number passed network screening (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
   Forwarded unconditionally (15)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]
 
 No. I was trying to say 

Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

2011/11/9 Richard Mudgett rmudg...@digium.com

As promised, here is a follow up on my quest to get CallerID
correctly
presented when forwarding calls to cellphones.
   
Here is a reminder of the issue at hand:
   
Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
which forwards to Cory (GSM handset)
What I would like to get is to see Alice's number (not Bob's
number)
presented to Cory.
Sometimes, I get Alice's number, sometimes, I get Bob's number
(new
findings from last sunday trials).
And of course, if Daniel or Eric would call Bob, the CallerID
number
presented to Cory would either be Daniel's number, Eric's number
or
Bob's number depending on a root cause I'm looking after for
several
days now.
   
   
   
To check if CallerID is filtered or controlled by Telco, I
originated
calls from Asterisk using hand crafted caller ids: any CallerID
was
correctly presented.
So I originally thought the root cause I'm after is a telco
equipment
switching ANI and CID.
But a close look at some last trials output makes me asking for
opinions from this list readers.
   
Here follows, the anonymized (and hand indented) output of command
PRI
debug command.
I focused on the end of call setup dialog.
   
For the successfully presented call, the output is:
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [6c 0b 21 83 37 38
36
XX XX XX XX XX XX]
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Calling Number
(len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Presentation:
Presentation allowed of network provided number (3) '78649' ]
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [70 0b 80 30 36 37
31
XX XX XX XX XX XX]
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Called Number
(len=13)
[ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
'067100' ]
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [74 0e 21 01 8f 33
33
33 34 34 XX XX XX XX XX XX]
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
(len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0
Presentation:
Presentation permitted, user number passed network screening (1)
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
Forwarded unconditionally (15)
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [a1]
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Sending Complete
(len=
1)
   
   
For the unsuccessfully presented call, the output is:
[Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [6c 0b 21 83 36 37
38
XX XX XX XX XX XX]
[Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Calling Number
(len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)
[Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Presentation:
Presentation allowed of network provided number (3) '67854' ]
[Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [70 0b 80 30 36 37
31
XX XX XX XX XX XX]
[Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Called Number
(len=13)
[ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
'067100' ]
[Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [a1]
[Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Sending Complete
(len=
1)
   
   
Am I correctly interpreting when saying that in the successful
call,
Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX
XX
XX] message which is not otherwise sent ?
What can explains this difference ?
Is this something I can (should) control ?
   
For reference:
dahdi show version
DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC
pri show version
libpri version: 1.4.10.2
  
   Improved support for manipulation of redirecting number is available
   with the REDIRECTING dialplan function in Asterisk v1.8.x and
   libpri v1.4.12. Prior to Asterisk v1.8.x you only have
   CALLERID(RDNIS).
  
  
 https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
  
  
   Richard
  
  
   Hi Richard,
  
   1. Could you elaborate a bit ?
   Do you imply that the lines bellow were present (or missing) because
   I
   did somewhere set CALLERID(RDNIS) and that I should use them ?
  
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
(len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)
[Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0
Presentation:
Presentation permitted, user number passed network screening (1)
[Nov 

Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Richard Mudgett]

  2. As I feel specically new to this RDNIS concept, how should I set
  CALLERID(RDNIS), before or after Answer() statement ?
 
 It does not matter in this case. Asterisk v1.6.1 will keep both legs
 of the call anyway.
 
 If you ultimately want to get the call entirely off of your Asterisk
 server, you will need Asterisk v1.6.2 or later. You would also need
 libpri 1.4.12 to do this with ETSI(EuroISDN). You would then use
 the DAHDISendCallreroutingFacility application *before* you answer
 the call to forward/deflect the incoming call back to the network.
 
 
 
 
 Richard
 
 
 That's definitively worth to try.
 I can't think of any use case but does this
 DAHDISendCallreroutingFacility generates AMI events, for curiosity's
 sake ?

No.  The application just asks libpri to send a FACILITY message to the
network.  Other AMI events are generated as a result of the redirected
call clearing.

Richard

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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

Hi,

As promised, here is a follow up on my quest to get CallerID correctly
presented when forwarding calls to cellphones.

Here is a reminder of the issue at hand:

Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension) which
forwards to Cory (GSM handset)
What I would like to get is to see Alice's number (not Bob's number)
presented to Cory.
Sometimes, I get Alice's number, sometimes, I get Bob's number (new
findings from last sunday trials).
And of course, if Daniel or Eric would call Bob, the CallerID number
presented to Cory would either be Daniel's number, Eric's number or Bob's
number depending on a root cause I'm looking after for several days now.



To check if CallerID is filtered or controlled by Telco, I originated calls
from Asterisk using hand crafted caller ids: any CallerID was correctly
presented.
So I originally thought the root cause I'm after is a telco equipment
switching ANI and CID.
But a close look at some last trials output makes me asking for opinions
from this list readers.

Here follows, the anonymized (and hand indented) output of command PRI
debug command.
I focused on the end of call setup dialog.

For the successfully presented call, the output is:
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [6c 0b 21 83 37 38 36 XX
XX XX XX XX XX]
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Calling Number (len=13) [
Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1)
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c: 
Presentation: Presentation allowed of network provided number (3)
'78649' ]
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [70 0b 80 30 36 37 31 XX
XX XX XX XX XX]
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Called Number (len=13) [
Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
'067100' ]
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [74 0e 21 01 8f 33 33 33
34 34 XX XX XX XX XX XX]
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
(len=16) [ Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering
Plan (E.164/E.163) (1)
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:
   Ext: 0  Presentation: Presentation
permitted, user number passed network screening (1)
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:
   Ext: 1  Reason: Forwarded unconditionally
(15)
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:   '3334436' ]
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [a1]
[Nov  6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Sending Complete (len= 1)


For the unsuccessfully presented call, the output is:
[Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [6c 0b 21 83 36 37 38 XX
XX XX XX XX XX]
[Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Calling Number (len=13) [
Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan
(E.164/E.163) (1)
[Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c: 
Presentation: Presentation allowed of network provided number (3)
'67854' ]
[Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [70 0b 80 30 36 37 31 XX
XX XX XX XX XX]
[Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Called Number (len=13) [
Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
'067100' ]
[Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [a1]
[Nov  6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Sending Complete (len= 1)


Am I correctly interpreting when saying that in the successful call,
Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX XX]
message which is not otherwise sent ?
What can explains this difference ?
Is this something I can (should) control ?

For reference:
dahdi show version
DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC
pri show version
libpri version: 1.4.10.2



Regards
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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Richard Mudgett]

 As promised, here is a follow up on my quest to get CallerID correctly
 presented when forwarding calls to cellphones.
 
 Here is a reminder of the issue at hand:
 
 Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
 which forwards to Cory (GSM handset)
 What I would like to get is to see Alice's number (not Bob's number)
 presented to Cory.
 Sometimes, I get Alice's number, sometimes, I get Bob's number (new
 findings from last sunday trials).
 And of course, if Daniel or Eric would call Bob, the CallerID number
 presented to Cory would either be Daniel's number, Eric's number or
 Bob's number depending on a root cause I'm looking after for several
 days now.
 
 
 
 To check if CallerID is filtered or controlled by Telco, I originated
 calls from Asterisk using hand crafted caller ids: any CallerID was
 correctly presented.
 So I originally thought the root cause I'm after is a telco equipment
 switching ANI and CID.
 But a close look at some last trials output makes me asking for
 opinions from this list readers.
 
 Here follows, the anonymized (and hand indented) output of command PRI
 debug command.
 I focused on the end of call setup dialog.
 
 For the successfully presented call, the output is:
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [6c 0b 21 83 37 38 36
 XX XX XX XX XX XX]
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Calling Number
 (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
 Numbering Plan (E.164/E.163) (1)
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Presentation:
 Presentation allowed of network provided number (3) '78649' ]
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [70 0b 80 30 36 37 31
 XX XX XX XX XX XX]
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Called Number (len=13)
 [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
 '067100' ]
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [74 0e 21 01 8f 33 33
 33 34 34 XX XX XX XX XX XX]
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
 (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
 Numbering Plan (E.164/E.163) (1)
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0 Presentation:
 Presentation permitted, user number passed network screening (1)
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
 Forwarded unconditionally (15)
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [a1]
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Sending Complete (len=
 1)
 
 
 For the unsuccessfully presented call, the output is:
 [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [6c 0b 21 83 36 37 38
 XX XX XX XX XX XX]
 [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Calling Number
 (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
 Numbering Plan (E.164/E.163) (1)
 [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Presentation:
 Presentation allowed of network provided number (3) '67854' ]
 [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [70 0b 80 30 36 37 31
 XX XX XX XX XX XX]
 [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Called Number (len=13)
 [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
 '067100' ]
 [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [a1]
 [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Sending Complete (len=
 1)
 
 
 Am I correctly interpreting when saying that in the successful call,
 Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX
 XX] message which is not otherwise sent ?
 What can explains this difference ?
 Is this something I can (should) control ?
 
 For reference:
 dahdi show version
 DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC
 pri show version
 libpri version: 1.4.10.2

Improved support for manipulation of redirecting number is available
with the REDIRECTING dialplan function in Asterisk v1.8.x and
libpri v1.4.12.  Prior to Asterisk v1.8.x you only have CALLERID(RDNIS).

https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information


Richard

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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

2011/11/8 Richard Mudgett rmudg...@digium.com

  As promised, here is a follow up on my quest to get CallerID correctly
  presented when forwarding calls to cellphones.
 
  Here is a reminder of the issue at hand:
 
  Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
  which forwards to Cory (GSM handset)
  What I would like to get is to see Alice's number (not Bob's number)
  presented to Cory.
  Sometimes, I get Alice's number, sometimes, I get Bob's number (new
  findings from last sunday trials).
  And of course, if Daniel or Eric would call Bob, the CallerID number
  presented to Cory would either be Daniel's number, Eric's number or
  Bob's number depending on a root cause I'm looking after for several
  days now.
 
 
 
  To check if CallerID is filtered or controlled by Telco, I originated
  calls from Asterisk using hand crafted caller ids: any CallerID was
  correctly presented.
  So I originally thought the root cause I'm after is a telco equipment
  switching ANI and CID.
  But a close look at some last trials output makes me asking for
  opinions from this list readers.
 
  Here follows, the anonymized (and hand indented) output of command PRI
  debug command.
  I focused on the end of call setup dialog.
 
  For the successfully presented call, the output is:
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [6c 0b 21 83 37 38 36
  XX XX XX XX XX XX]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Calling Number
  (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
  Numbering Plan (E.164/E.163) (1)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Presentation:
  Presentation allowed of network provided number (3) '78649' ]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [70 0b 80 30 36 37 31
  XX XX XX XX XX XX]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Called Number (len=13)
  [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
  '067100' ]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [74 0e 21 01 8f 33 33
  33 34 34 XX XX XX XX XX XX]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
  (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
  Numbering Plan (E.164/E.163) (1)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0 Presentation:
  Presentation permitted, user number passed network screening (1)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
  Forwarded unconditionally (15)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [a1]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Sending Complete (len=
  1)
 
 
  For the unsuccessfully presented call, the output is:
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [6c 0b 21 83 36 37 38
  XX XX XX XX XX XX]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Calling Number
  (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
  Numbering Plan (E.164/E.163) (1)
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Presentation:
  Presentation allowed of network provided number (3) '67854' ]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [70 0b 80 30 36 37 31
  XX XX XX XX XX XX]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Called Number (len=13)
  [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
  '067100' ]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [a1]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Sending Complete (len=
  1)
 
 
  Am I correctly interpreting when saying that in the successful call,
  Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX
  XX] message which is not otherwise sent ?
  What can explains this difference ?
  Is this something I can (should) control ?
 
  For reference:
  dahdi show version
  DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC
  pri show version
  libpri version: 1.4.10.2

 Improved support for manipulation of redirecting number is available
 with the REDIRECTING dialplan function in Asterisk v1.8.x and
 libpri v1.4.12.  Prior to Asterisk v1.8.x you only have CALLERID(RDNIS).


 https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information


 Richard


Hi Richard,

1. Could you elaborate a bit ?
Do you imply that the lines bellow were present (or missing) because I did
somewhere set CALLERID(RDNIS) and that I should use them ?

 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
 (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
 Numbering Plan (E.164/E.163) (1)
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0 Presentation:
 Presentation permitted, user number passed network screening (1)
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
 Forwarded unconditionally (15)
 [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]

2. More generally, if I may ask, how do you understand both outputs (from
my previous post) ?

Regards


 --
 

Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Richard Mudgett]

  As promised, here is a follow up on my quest to get CallerID
  correctly
  presented when forwarding calls to cellphones.
 
  Here is a reminder of the issue at hand:
 
  Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
  which forwards to Cory (GSM handset)
  What I would like to get is to see Alice's number (not Bob's number)
  presented to Cory.
  Sometimes, I get Alice's number, sometimes, I get Bob's number (new
  findings from last sunday trials).
  And of course, if Daniel or Eric would call Bob, the CallerID number
  presented to Cory would either be Daniel's number, Eric's number or
  Bob's number depending on a root cause I'm looking after for several
  days now.
 
 
 
  To check if CallerID is filtered or controlled by Telco, I
  originated
  calls from Asterisk using hand crafted caller ids: any CallerID was
  correctly presented.
  So I originally thought the root cause I'm after is a telco
  equipment
  switching ANI and CID.
  But a close look at some last trials output makes me asking for
  opinions from this list readers.
 
  Here follows, the anonymized (and hand indented) output of command
  PRI
  debug command.
  I focused on the end of call setup dialog.
 
  For the successfully presented call, the output is:
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [6c 0b 21 83 37 38
  36
  XX XX XX XX XX XX]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Calling Number
  (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
  Numbering Plan (E.164/E.163) (1)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Presentation:
  Presentation allowed of network provided number (3) '78649' ]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [70 0b 80 30 36 37
  31
  XX XX XX XX XX XX]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Called Number
  (len=13)
  [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
  '067100' ]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [74 0e 21 01 8f 33
  33
  33 34 34 XX XX XX XX XX XX]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
  (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
  Numbering Plan (E.164/E.163) (1)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0 Presentation:
  Presentation permitted, user number passed network screening (1)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
  Forwarded unconditionally (15)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [a1]
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Sending Complete
  (len=
  1)
 
 
  For the unsuccessfully presented call, the output is:
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [6c 0b 21 83 36 37
  38
  XX XX XX XX XX XX]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Calling Number
  (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
  Numbering Plan (E.164/E.163) (1)
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Presentation:
  Presentation allowed of network provided number (3) '67854' ]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [70 0b 80 30 36 37
  31
  XX XX XX XX XX XX]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Called Number
  (len=13)
  [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
  '067100' ]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [a1]
  [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Sending Complete
  (len=
  1)
 
 
  Am I correctly interpreting when saying that in the successful call,
  Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX
  XX] message which is not otherwise sent ?
  What can explains this difference ?
  Is this something I can (should) control ?
 
  For reference:
  dahdi show version
  DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC
  pri show version
  libpri version: 1.4.10.2
 
 Improved support for manipulation of redirecting number is available
 with the REDIRECTING dialplan function in Asterisk v1.8.x and
 libpri v1.4.12. Prior to Asterisk v1.8.x you only have
 CALLERID(RDNIS).
 
 https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
 
 
 Richard
 
 
 Hi Richard,
 
 1. Could you elaborate a bit ?
 Do you imply that the lines bellow were present (or missing) because I
 did somewhere set CALLERID(RDNIS) and that I should use them ?
 
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
  (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
  Numbering Plan (E.164/E.163) (1)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0 Presentation:
  Presentation permitted, user number passed network screening (1)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
  Forwarded unconditionally (15)
  [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]

No.  I was trying to say that the value in the redirecting ie is
controllable by setting/clearing the CALLERID(RDNIS) value.

 2. More generally, if I may ask, how do you 

Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

2011/11/8 Richard Mudgett rmudg...@digium.com

   As promised, here is a follow up on my quest to get CallerID
   correctly
   presented when forwarding calls to cellphones.
  
   Here is a reminder of the issue at hand:
  
   Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
   which forwards to Cory (GSM handset)
   What I would like to get is to see Alice's number (not Bob's number)
   presented to Cory.
   Sometimes, I get Alice's number, sometimes, I get Bob's number (new
   findings from last sunday trials).
   And of course, if Daniel or Eric would call Bob, the CallerID number
   presented to Cory would either be Daniel's number, Eric's number or
   Bob's number depending on a root cause I'm looking after for several
   days now.
  
  
  
   To check if CallerID is filtered or controlled by Telco, I
   originated
   calls from Asterisk using hand crafted caller ids: any CallerID was
   correctly presented.
   So I originally thought the root cause I'm after is a telco
   equipment
   switching ANI and CID.
   But a close look at some last trials output makes me asking for
   opinions from this list readers.
  
   Here follows, the anonymized (and hand indented) output of command
   PRI
   debug command.
   I focused on the end of call setup dialog.
  
   For the successfully presented call, the output is:
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [6c 0b 21 83 37 38
   36
   XX XX XX XX XX XX]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Calling Number
   (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
   Numbering Plan (E.164/E.163) (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Presentation:
   Presentation allowed of network provided number (3) '78649' ]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [70 0b 80 30 36 37
   31
   XX XX XX XX XX XX]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Called Number
   (len=13)
   [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
   '067100' ]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [74 0e 21 01 8f 33
   33
   33 34 34 XX XX XX XX XX XX]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
   (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
   Numbering Plan (E.164/E.163) (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0 Presentation:
   Presentation permitted, user number passed network screening (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
   Forwarded unconditionally (15)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  [a1]
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Sending Complete
   (len=
   1)
  
  
   For the unsuccessfully presented call, the output is:
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [6c 0b 21 83 36 37
   38
   XX XX XX XX XX XX]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Calling Number
   (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
   Numbering Plan (E.164/E.163) (1)
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Presentation:
   Presentation allowed of network provided number (3) '67854' ]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [70 0b 80 30 36 37
   31
   XX XX XX XX XX XX]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Called Number
   (len=13)
   [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
   '067100' ]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  [a1]
   [Nov 6 09:25:29] VERBOSE[27927] chan_dahdi.c:  Sending Complete
   (len=
   1)
  
  
   Am I correctly interpreting when saying that in the successful call,
   Asterisk is sending a [74 0e 21 01 8f 33 33 33 34 34 XX XX XX XX XX
   XX] message which is not otherwise sent ?
   What can explains this difference ?
   Is this something I can (should) control ?
  
   For reference:
   dahdi show version
   DAHDI Version: SVN-trunk-r8853M Echo Canceller: OSLEC
   pri show version
   libpri version: 1.4.10.2
 
  Improved support for manipulation of redirecting number is available
  with the REDIRECTING dialplan function in Asterisk v1.8.x and
  libpri v1.4.12. Prior to Asterisk v1.8.x you only have
  CALLERID(RDNIS).
 
 
 https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
 
 
  Richard
 
 
  Hi Richard,
 
  1. Could you elaborate a bit ?
  Do you imply that the lines bellow were present (or missing) because I
  did somewhere set CALLERID(RDNIS) and that I should use them ?
 
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Redirecting Number
   (len=16) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
   Numbering Plan (E.164/E.163) (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 0 Presentation:
   Presentation permitted, user number passed network screening (1)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c:  Ext: 1 Reason:
   Forwarded unconditionally (15)
   [Nov 6 09:32:07] VERBOSE[27954] chan_dahdi.c: '3334436' ]

 No.  I 

Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[giovanni.v]

Il 04/11/2011 9.32, Olivier ha scritto:

I don't know how Telcos manage their networks.
I would have naturally thought that going from one point to another in
this network would always pass through the same set of equipements.


It was ... about 30 (or more) years ago, today most moved from switched 
network to packet network.



Maybe it's not the case and specifically when the end destination is a
cellphone and there is a large gateway between the landline and the
mobile networks.


I think most is carried on ss7(*) over ip, passing through a wide range 
of equipments from many manufacturers. Each of these could have it's own 
bugs or inaccurate configuration.



Is this reasonning correct ?


I think it is, you are facing some internetwork problem out of your control.

Anyway you're lucky, changing the caller presentation number to anything 
that not belongs to the subscriber circuit would be simply impossible 
here due to regulations.


(*) - http://www.ss7-training.net/index.html

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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

2011/11/3 Danny Nicholas da...@debsinc.com

 snip

 [callbob]


 Exten = _XX.,1,answer

 Exten = _XX.,n,Set(CALLERID(num)=${EXTEN})

 Exten = _XX.,n,Dial(DAHDI/1/5551212,30)


From memory, I did it this way :
Exten = _XX.,1,Set(CALLERID(num)=whatever)
Exten = _XX.,n,answer
Exten = _XX.,n,Dial(DAHDI/1/5551212,30)

This seems to work for roughly 2/3 of calls.
I'm trying to get as close to 100% as possible.

I can't think of any configuration issue or bug in Asterisk that would
cause 1/3 of calls to be incorrectly presented.
I rather suspect that different calls MAY use different routes and a
misconfiguration in one equipement in one of those would cause the issue.

I don't know how Telcos manage their networks.
I would have naturally thought that going from one point to another in this
network would always pass through the same set of equipements.
Maybe it's not the case and specifically when the end destination is a
cellphone and there is a large gateway between the landline and the mobile
networks.

Is this reasonning correct ?
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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Danny Nicholas]

When you say 2/3 of calls, is there an inconsistency to the same recipient
or could it be a carrier issue (Verizon only, T-Mobile only, etc.)?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, November 04, 2011 3:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks

 

 

2011/11/3 Danny Nicholas da...@debsinc.com

snip


[callbob]


Exten = _XX.,1,answer

Exten = _XX.,n,Set(CALLERID(num)=${EXTEN})

Exten = _XX.,n,Dial(DAHDI/1/5551212,30)


From memory, I did it this way :
Exten = _XX.,1,Set(CALLERID(num)=whatever)
Exten = _XX.,n,answer
Exten = _XX.,n,Dial(DAHDI/1/5551212,30)

This seems to work for roughly 2/3 of calls.
I'm trying to get as close to 100% as possible.

I can't think of any configuration issue or bug in Asterisk that would cause
1/3 of calls to be incorrectly presented.
I rather suspect that different calls MAY use different routes and a
misconfiguration in one equipement in one of those would cause the issue.

I don't know how Telcos manage their networks.
I would have naturally thought that going from one point to another in this
network would always pass through the same set of equipements.
Maybe it's not the case and specifically when the end destination is a
cellphone and there is a large gateway between the landline and the mobile
networks.

Is this reasonning correct ?

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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

2011/11/4 Danny Nicholas da...@debsinc.com

 When you say 2/3 of calls, is there an inconsistency to the same recipient
 or could it be a carrier issue (Verizon only, T-Mobile only, etc.)?



When we tested, calls were originated by 32 different cellphones to a
unique ISDN-connected Asterisk box which then forwarded this call to a
unique destination cellphone.
Naming things this way A calls B which forwards to C, we used 32
different cellphones as A, and only one asterisk and one cellphone as
respectively B and C.

A doubt I still get at the moment, is I can't swear B always received a
correct CallerID from A (maybe, it received 32 correct Caller IDs, maybe
not).
That's why I'm gonna try again this weekend to record full logs of calls
leaving Asterisk to the destination cellphone.

Asterisk and cellphones are connected to each other with France Telecom
ISDN and GSM networks.
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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

Hi,

I'm still strugling with my CallerID presentation problem.
Let me remind it :

My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone

Ive configured Asterisk so that whenever Bob forwards its incoming call to
its cellphone, the later phone should present Alice's number.

I was originally told that sometimes Bob would be presented Alice's number,
sometimes the dialed number (which in this case, also match the ANI or the
receptionnist ID).
Now, I can't certify this ever happened : maybe, it did happen, maybe not.

What I can certify is this:
1. out of 32 different callers, 20 callers are presented with the correct
number and 12 with the dialed number,
2. all those 32 callers are cellphones operated by the same (incumbent)
telco which also operates ISDN,
3. Bob's cellphone is also operated by the same (incumbent) telco,
4. all this tries were done the same day, one after the other.

The best explanation I can think of is this:
Depending on the route used, the ANI is used instead of the presented
caller ID.

To prove that, I'll try to record 2 calls for the same caller and toward
the same destination: one with the awaited presentation, one with a wrong
one.
(Sending this to the telco and have them change anything is an other story).


Comments and suggestions are welcome.

Regards
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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Danny Nicholas]

What version of Asterisk?  Is the forwarding done using Followme, attended
transfer or blind transfer?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks

 

Hi,

I'm still strugling with my CallerID presentation problem.
Let me remind it :

My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone

Ive configured Asterisk so that whenever Bob forwards its incoming call to
its cellphone, the later phone should present Alice's number.

I was originally told that sometimes Bob would be presented Alice's number,
sometimes the dialed number (which in this case, also match the ANI or the
receptionnist ID).
Now, I can't certify this ever happened : maybe, it did happen, maybe not.

What I can certify is this:
1. out of 32 different callers, 20 callers are presented with the correct
number and 12 with the dialed number,
2. all those 32 callers are cellphones operated by the same (incumbent)
telco which also operates ISDN,
3. Bob's cellphone is also operated by the same (incumbent) telco,
4. all this tries were done the same day, one after the other.

The best explanation I can think of is this:
Depending on the route used, the ANI is used instead of the presented
caller ID.

To prove that, I'll try to record 2 calls for the same caller and toward
the same destination: one with the awaited presentation, one with a wrong
one.
(Sending this to the telco and have them change anything is an other story).


Comments and suggestions are welcome.

Regards

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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

2011/11/3 Danny Nicholas da...@debsinc.com

 What version of Asterisk?

1.6.1.18

   Is the forwarding done using Followme, attended transfer or blind
 transfer?

a plain Answer plus Dial


 

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
 *Sent:* Thursday, November 03, 2011 8:14 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] CallerID inconsistently presented through
 ISDN/cellular networks

 ** **

 Hi,

 I'm still strugling with my CallerID presentation problem.
 Let me remind it :

 My setup is:
 Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
 cellphone

 Ive configured Asterisk so that whenever Bob forwards its incoming call to
 its cellphone, the later phone should present Alice's number.

 I was originally told that sometimes Bob would be presented Alice's
 number, sometimes the dialed number (which in this case, also match the ANI
 or the receptionnist ID).
 Now, I can't certify this ever happened : maybe, it did happen, maybe not.

 What I can certify is this:
 1. out of 32 different callers, 20 callers are presented with the correct
 number and 12 with the dialed number,
 2. all those 32 callers are cellphones operated by the same (incumbent)
 telco which also operates ISDN,
 3. Bob's cellphone is also operated by the same (incumbent) telco,
 4. all this tries were done the same day, one after the other.

 The best explanation I can think of is this:
 Depending on the route used, the ANI is used instead of the presented
 caller ID.

 To prove that, I'll try to record 2 calls for the same caller and toward
 the same destination: one with the awaited presentation, one with a wrong
 one.
 (Sending this to the telco and have them change anything is an other
 story).


 Comments and suggestions are welcome.

 Regards

 --
 _
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 asterisk-users mailing list
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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Danny Nicholas]

Something like this?
[callbob]

Exten = start,1,answer

Exten = start,n,Dial(DAHDI/1/5551212,30)

If that is the case, Bob should always get the Caller ID of your asterisk
installation - I would suggest this instead

[callbob]

Exten = start,1,answer

Exten = start,n,Set(CALLERID(num)=${EXTEN})

Exten = start,n,Dial(DAHDI/1/5551212,30)

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks

 

 

2011/11/3 Danny Nicholas da...@debsinc.com

What version of Asterisk?

1.6.1.18 

  Is the forwarding done using Followme, attended transfer or blind
transfer?

a plain Answer plus Dial
 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks

 

Hi,

I'm still strugling with my CallerID presentation problem.
Let me remind it :

My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone

Ive configured Asterisk so that whenever Bob forwards its incoming call to
its cellphone, the later phone should present Alice's number.

I was originally told that sometimes Bob would be presented Alice's number,
sometimes the dialed number (which in this case, also match the ANI or the
receptionnist ID).
Now, I can't certify this ever happened : maybe, it did happen, maybe not.

What I can certify is this:
1. out of 32 different callers, 20 callers are presented with the correct
number and 12 with the dialed number,
2. all those 32 callers are cellphones operated by the same (incumbent)
telco which also operates ISDN,
3. Bob's cellphone is also operated by the same (incumbent) telco,
4. all this tries were done the same day, one after the other.

The best explanation I can think of is this:
Depending on the route used, the ANI is used instead of the presented
caller ID.

To prove that, I'll try to record 2 calls for the same caller and toward
the same destination: one with the awaited presentation, one with a wrong
one.
(Sending this to the telco and have them change anything is an other story).


Comments and suggestions are welcome.

Regards


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  http://lists.digium.com/mailman/listinfo/asterisk-users

 

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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Eric Wieling]

In your example the CallerID number will always be start.   Not what he is 
looking for.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, November 03, 2011 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CallerID inconsistently presented through 
ISDN/cellular networks

Something like this?
[callbob]

Exten = start,1,answer

Exten = start,n,Dial(DAHDI/1/5551212,30)

If that is the case, Bob should always get the Caller ID of your asterisk 
installation - I would suggest this instead

[callbob]

Exten = start,1,answer

Exten = start,n,Set(CALLERID(num)=${EXTEN})

Exten = start,n,Dial(DAHDI/1/5551212,30)

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through 
ISDN/cellular networks

 

 

2011/11/3 Danny Nicholas da...@debsinc.com

What version of Asterisk?

1.6.1.18 

  Is the forwarding done using Followme, attended transfer or blind 
transfer?

a plain Answer plus Dial
 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through 
ISDN/cellular networks

 

Hi,

I'm still strugling with my CallerID presentation problem.
Let me remind it :

My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob 
cellphone

Ive configured Asterisk so that whenever Bob forwards its incoming call 
to its cellphone, the later phone should present Alice's number.

I was originally told that sometimes Bob would be presented Alice's 
number, sometimes the dialed number (which in this case, also match the ANI or 
the receptionnist ID).
Now, I can't certify this ever happened : maybe, it did happen, maybe 
not.

What I can certify is this:
1. out of 32 different callers, 20 callers are presented with the 
correct number and 12 with the dialed number,
2. all those 32 callers are cellphones operated by the same (incumbent) 
telco which also operates ISDN,
3. Bob's cellphone is also operated by the same (incumbent) telco,
4. all this tries were done the same day, one after the other.

The best explanation I can think of is this:
Depending on the route used, the ANI is used instead of the presented 
caller ID.

To prove that, I'll try to record 2 calls for the same caller and 
toward the same destination: one with the awaited presentation, one with a 
wrong one.
(Sending this to the telco and have them change anything is an other 
story).


Comments and suggestions are welcome.

Regards


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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Danny Nicholas]

Trying to save a few keystrokes - better example
[callbob]

Exten = _XX.,1,answer

Exten = _XX.,n,Dial(DAHDI/1/5551212,30)

If that is the case, Bob should always get the Caller ID of your asterisk
installation - I would suggest this instead

[callbob]

Exten = _XX.,1,answer

Exten = _XX.,n,Set(CALLERID(num)=${EXTEN})

Exten = _XX.,n,Dial(DAHDI/1/5551212,30)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Thursday, November 03, 2011 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks

In your example the CallerID number will always be start.   Not what he is
looking for.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, November 03, 2011 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks

Something like this?
[callbob]

Exten = start,1,answer

Exten = start,n,Dial(DAHDI/1/5551212,30)

If that is the case, Bob should always get the Caller ID of your asterisk
installation - I would suggest this instead

[callbob]

Exten = start,1,answer

Exten = start,n,Set(CALLERID(num)=${EXTEN})

Exten = start,n,Dial(DAHDI/1/5551212,30)

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks

 

 

2011/11/3 Danny Nicholas da...@debsinc.com

What version of Asterisk?

1.6.1.18 

  Is the forwarding done using Followme, attended transfer or blind
transfer?

a plain Answer plus Dial
 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented
through ISDN/cellular networks

 

Hi,

I'm still strugling with my CallerID presentation problem.
Let me remind it :

My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM--
Bob cellphone

Ive configured Asterisk so that whenever Bob forwards its incoming
call to its cellphone, the later phone should present Alice's number.

I was originally told that sometimes Bob would be presented Alice's
number, sometimes the dialed number (which in this case, also match the ANI
or the receptionnist ID).
Now, I can't certify this ever happened : maybe, it did happen,
maybe not.

What I can certify is this:
1. out of 32 different callers, 20 callers are presented with the
correct number and 12 with the dialed number,
2. all those 32 callers are cellphones operated by the same
(incumbent) telco which also operates ISDN,
3. Bob's cellphone is also operated by the same (incumbent) telco,
4. all this tries were done the same day, one after the other.

The best explanation I can think of is this:
Depending on the route used, the ANI is used instead of the
presented caller ID.

To prove that, I'll try to record 2 calls for the same caller and
toward the same destination: one with the awaited presentation, one with a
wrong one.
(Sending this to the telco and have them change anything is an other
story).


Comments and suggestions are welcome.

Regards


--

_
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--
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Thurs:
  http://www.asterisk.org/hello

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  http://lists.digium.com/mailman/listinfo/asterisk-users

 


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[asterisk-users] ${CALLERID(num)} after doing transfer from extension to extension

[bilal ghayyad]

Hi All;

As I am using the ${CALLERID(num)} to be part of the filename that I am 
recording it, I am facing the following problem:

If the incoming call (via PSTN) reached for an extension (which is the 
reception), and then the extension transferred the call to the proper person, 
and we need to do recording for the call at this proper person, the problem 
that at this point the ${CALLERID(num)} will represnt the reception guy 
extension and not the original caller id of the caller who called from outside 
via the PSTN. How can I get this original caller id?

Regards
Bilal

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Re: [asterisk-users] ${CALLERID(num)} after doing transfer from extension to extension

[Sammy Govind]

Set CDR(destination) or whichever field you need to get recorded in CDRs to
get your desired stats.

On Mon, Oct 24, 2011 at 2:19 PM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hi All;

 As I am using the ${CALLERID(num)} to be part of the filename that I am
 recording it, I am facing the following problem:

 If the incoming call (via PSTN) reached for an extension (which is the
 reception), and then the extension transferred the call to the proper
 person, and we need to do recording for the call at this proper person, the
 problem that at this point the ${CALLERID(num)} will represnt the reception
 guy extension and not the original caller id of the caller who called from
 outside via the PSTN. How can I get this original caller id?

 Regards
 Bilal

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Re: [asterisk-users] ${CALLERID(num)} after doing transfer from extension to extension

[A J Stiles]

On Monday 24 October 2011, bilal ghayyad wrote:
 If the incoming call (via PSTN) reached for an extension (which is the
 reception), and then the extension transferred the call to the proper
 person, and we need to do recording for the call at this proper person,
 the problem that at this point the ${CALLERID(num)} will represnt the
 reception guy extension and not the original caller id of the caller who
 called from outside via the PSTN. How can I get this original caller id?

As soon as the incoming call lands in a context, store the caller's number in 
a variable; for instance,
Set(ORIG_NUM=${CALLERID(num)})
and then when building up the call filename, just use ${ORIG_NUM} instead of 
=${CALLERID(num)}


-- 
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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Daniel Tryba]

On Thu, Oct 13, 2011 at 12:37:28AM +0200, Andreas Sikkema wrote:
 So normally calls to these destinations have nice caller id as if A was
 calling C (at least that's what C sees in their display) but every now
 and then I flow over to the alternative route and the information is
 lost, C doesn't see A, but B.
 
 Nothing I can do about it, been fighting over it for ages but I just
 doesn't seem to be able to make it work.

Suddenly I feel very lucky. It only took me a couple of weeks of sending
10 test calls per day of resulting callerids mishaps with Verizon to get
them to finaly trace the problem and correct a misconfigured switch. It
also helps to be able to route all mobile traffic through an other
provider, if they start to lose lots of minutes providers will act.

-- 

   Daniel Tryba

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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Andreas Sikkema]

On 10/11/11 8:10 PM, Olivier wrote:

 I'll start a test session in a couple of minutes and report here.
 
 The strangest things is this inconsistency: I can imagine million of
 reasons why a number is not presented but I can't think of any
 explaining why it would change in a couple of hours.

Inconsistent configuration over multiple routes probably. I know I have
one route (the default actually) to a number of destinations where I am
100% percent able to send redirected number information, but another
route just will not pass it on to the destination.

So normally calls to these destinations have nice caller id as if A was
calling C (at least that's what C sees in their display) but every now
and then I flow over to the alternative route and the information is
lost, C doesn't see A, but B.

Nothing I can do about it, been fighting over it for ages but I just
doesn't seem to be able to make it work.

-- 
Andreas Sikkema

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[asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

Hi,

I'm facing a strange problem.

My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone

When Alice calls Asterisk which forwards the incoming call to Bob, sometimes
Bob sees Alice's number, sometimes he sees a default CallerID (which happens
to match the dialed number and the ANI).
For various reasons, Bob really needs to see Alice's number when Alice is
calling.

When I compare one successful (ie presented with Alice ID) calls with one
unsuccessful (with debug and verbose levels respectively set to 0 and 3),
I can't see any difference between both calls within Asterisk logs :
every time Asterisk, receives Alice CallerID and set outgoing channel
CallerID with the same value.
(The only I could find, at the moment, to distinguish a successful call is
to call Bob and ask him to tell what happened).

If that matters, let me add this:
- each incoming call is forwarded with a simple Answer(), Dial() sequence,
- when I'm presenting an outgoing with too many digits, the call is
presented with a default CallerID.

My understanding is as each network used is purely digital, you can't loose
CallerID.
Is this roughly correct ?
In which direction, shall I dig ?


Regards
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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Warren Selby]

On Tue, Oct 11, 2011 at 11:06 AM, Olivier oza_4...@yahoo.fr wrote:

 Hi,

 I'm facing a strange problem.

 My setup is:
 Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
 cellphone

 When Alice calls Asterisk which forwards the incoming call to Bob,
 sometimes Bob sees Alice's number, sometimes he sees a default CallerID
 (which happens to match the dialed number and the ANI).
 For various reasons, Bob really needs to see Alice's number when Alice is
 calling.


Is the outbound leg of the call always going out over the same outbound
provider?

Step up the verbose level to 10 and then make a few test calls until you've
got a successful callerID and an unsuccessful callerID, then paste an
example of each call (the complete call, from the initial inbound to the
hangup on the other end) in another email and we can review them for any
discrepancies.

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[A J Stiles]

On Tuesday 11 October 2011, Olivier wrote:
 Hi,
 
 I'm facing a strange problem.
 
 My setup is:
 Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
 cellphone
 
 When Alice calls Asterisk which forwards the incoming call to Bob,
 sometimes Bob sees Alice's number, sometimes he sees a default CallerID
 (which happens to match the dialed number and the ANI).
 For various reasons, Bob really needs to see Alice's number when Alice is
 calling.
 
 When I compare one successful (ie presented with Alice ID) calls with one
 unsuccessful (with debug and verbose levels respectively set to 0 and 3),
 I can't see any difference between both calls within Asterisk logs :
 every time Asterisk, receives Alice CallerID and set outgoing channel
 CallerID with the same value.
 (The only I could find, at the moment, to distinguish a successful call is
 to call Bob and ask him to tell what happened).
 
 If that matters, let me add this:
 - each incoming call is forwarded with a simple Answer(), Dial() sequence,
 - when I'm presenting an outgoing with too many digits, the call is
 presented with a default CallerID.
 
 My understanding is as each network used is purely digital, you can't
 loose CallerID.
 Is this roughly correct ?
 In which direction, shall I dig ?

Most telcos won't let you present a caller ID number that doesn't belong to 
you; so it's possible that the number you are presenting to Bob is being 
munged on the way to his mobile.  Otherwise, anybody with the right equipment 
would be able to pretend to be anybody else, and caller ID would be all but 
useless.

Anyway, what you really need to do is separate the two legs of the call, to 
see whether the number is changing between Alice and your Asterisk or between 
your Asterisk and Bob.  So put a Verbose() or NoOp() in your dialplan to see 
what caller ID Alice is sending, and get her to call you several times.  Then 
create a context to call Bob  (presenting Alice's number)  from a SIP phone, 
and call him several times.



-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

2011/10/11 Warren Selby wcse...@selbytech.com

 On Tue, Oct 11, 2011 at 11:06 AM, Olivier oza_4...@yahoo.fr wrote:

 Hi,

 I'm facing a strange problem.

 My setup is:
 Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
 cellphone

 When Alice calls Asterisk which forwards the incoming call to Bob,
 sometimes Bob sees Alice's number, sometimes he sees a default CallerID
 (which happens to match the dialed number and the ANI).
 For various reasons, Bob really needs to see Alice's number when Alice is
 calling.


 Is the outbound leg of the call always going out over the same outbound
 provider?


Yes: in this case, ISDN and GSM networks are both operated by incumbent
telco.



 Step up the verbose level to 10 and then make a few test calls until you've
 got a successful callerID and an unsuccessful callerID, then paste an
 example of each call (the complete call, from the initial inbound to the
 hangup on the other end) in another email and we can review them for any
 discrepancies.


I'll start a test session in a couple of minutes and report here.

The strangest things is this inconsistency: I can imagine million of reasons
why a number is not presented but I can't think of any explaining why it
would change in a couple of hours.


 --
 Thanks,
 --Warren Selby, dCAP
 http://www.SelbyTech.com http://www.selbytech.com


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Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks

[Olivier]

2011/10/11 A J Stiles asterisk_l...@earthshod.co.uk

 On Tuesday 11 October 2011, Olivier wrote:
  Hi,
 
  I'm facing a strange problem.
 
  My setup is:
  Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
  cellphone
 
  When Alice calls Asterisk which forwards the incoming call to Bob,
  sometimes Bob sees Alice's number, sometimes he sees a default CallerID
  (which happens to match the dialed number and the ANI).
  For various reasons, Bob really needs to see Alice's number when Alice is
  calling.
 
  When I compare one successful (ie presented with Alice ID) calls with
 one
  unsuccessful (with debug and verbose levels respectively set to 0 and
 3),
  I can't see any difference between both calls within Asterisk logs :
  every time Asterisk, receives Alice CallerID and set outgoing channel
  CallerID with the same value.
  (The only I could find, at the moment, to distinguish a successful call
 is
  to call Bob and ask him to tell what happened).
 
  If that matters, let me add this:
  - each incoming call is forwarded with a simple Answer(), Dial()
 sequence,
  - when I'm presenting an outgoing with too many digits, the call is
  presented with a default CallerID.
 
  My understanding is as each network used is purely digital, you can't
  loose CallerID.
  Is this roughly correct ?
  In which direction, shall I dig ?

 Most telcos won't let you present a caller ID number that doesn't belong to
 you; so it's possible that the number you are presenting to Bob is being
 munged on the way to his mobile.  Otherwise, anybody with the right
 equipment
 would be able to pretend to be anybody else, and caller ID would be all but
 useless.

 Anyway, what you really need to do is separate the two legs of the call, to
 see whether the number is changing between Alice and your Asterisk or
 between
 your Asterisk and Bob.  So put a Verbose() or NoOp() in your dialplan to
 see
 what caller ID Alice is sending, and get her to call you several times.
  Then
 create a context to call Bob  (presenting Alice's number)  from a SIP
 phone,
 and call him several times.


It's on my ToDo list.
I'll report here.




 --
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 Answers come *after* questions.

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[asterisk-users] Callerid issue

[mahesh katta]

Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid   Extensions
44578900  100
44578901  101
44578902  102
44578902  103
44578903  104
44578905  200
44578906  275
44578907  277
44578908  354

I need to setup the callerid with this extensions . for example whenever I
am dial from 354 extension callerID will show 44578908.
fro this scenarion I need logical dialplan because I have 100 extenstions ,
so 100 extentions should be have different extensions.


Below dialplan is for sequence callerid and extesions. like 101 to 199
should callerid is going 44578900 to 44578999 .

exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log)
exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,9,Hangup
-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] Callerid issue

[A J Stiles]

On Friday 10 Jun 2011, mahesh katta wrote:
 Hi,
 I have 44578900 to 44578999 DID's. and I have extensions(100) for this
 DID's. but problem is
 callerid   Extensions
 44578900  100
 44578901  101
 44578902  102
 44578902  103
 44578903  104
 44578905  200
 44578906  275
 44578907  277
 44578908  354

 I need to setup the callerid with this extensions . for example whenever I
 am dial from 354 extension callerID will show 44578908.
 fro this scenarion I need logical dialplan because I have 100 extenstions ,
 so 100 extentions should be have different extensions.

Sounds like a case for either  (1)  a different context per originating 
extension  (or maybe, per group of originating extensions which all happen to 
obey the same mathematical formula for determining outside callerID from 
inside extension number);  (2)  an AGI script, accessing a database which 
links internal extensions to external numbers; or  (3)  rethinking your 
internal extension numbering scheme so there is a consistent mapping from 
internal to external numbers, thus allowing you to do it all mathematically.

-- 
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Answers come *after* questions.

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Re: [asterisk-users] Callerid issue

[Steve Totaro]

On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta maheshka...@flexydial.com wrote:
 Hi,
 I have 44578900 to 44578999 DID's. and I have extensions(100) for this
 DID's. but problem is
 callerid   Extensions
 44578900  100
 44578901  101
 44578902  102
 44578902  103
 44578903  104
 44578905  200
 44578906  275
 44578907  277
 44578908  354

 I need to setup the callerid with this extensions . for example whenever I
 am dial from 354 extension callerID will show 44578908.
 fro this scenarion I need logical dialplan because I have 100 extenstions ,
 so 100 extentions should be have different extensions.


 Below dialplan is for sequence callerid and extesions. like 101 to 199
 should callerid is going 44578900 to 44578999 .

 exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
 exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
 exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
 exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
 exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log)
 exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
 exten =
 _0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
 exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
 exten = _0X,9,Hangup
 --
 Best Regards,

 Mahesh Katta
 BUZZWORKS Business Services Private Limited
 BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
 Mumbai 400069
 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
 Web http://www.buzzworks.com


Are you using SIP phones?  I assume you are but just checking.

Look into sip.conf

For each phone, add callerid=Joe Smith 1551212  no quotes in sip.conf

Be sure the telco allows it, some times they will only allow the BTN.
I have run into troubles with toll free callerid.  Everything made
sense because the problem was calling other toll free numbers and who
pays.  Good point.  Especially because ANI can be manipulated in
Asterisk as well.

I never understood hy people who have block of DIDs in a row choose to
make life difficult by not incrementing extensions by one, send caller
ID by prepending the common numbers and only sending four digits.

I guess if you get huge there could be duplicates of the last four.

Just set it in sip.conf

Thanks,
Steve T
Thanks
Steve Totaro

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Re: [asterisk-users] Callerid issue

[Steve Totaro]

On Fri, Jun 10, 2011 at 6:27 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
 On Friday 10 Jun 2011, mahesh katta wrote:
 Hi,
 I have 44578900 to 44578999 DID's. and I have extensions(100) for this
 DID's. but problem is
 callerid           Extensions
 44578900      100
 44578901      101
 44578902      102
 44578902      103
 44578903      104
 44578905      200
 44578906      275
 44578907      277
 44578908      354

 I need to setup the callerid with this extensions . for example whenever I
 am dial from 354 extension callerID will show 44578908.
 fro this scenarion I need logical dialplan because I have 100 extenstions ,
 so 100 extentions should be have different extensions.

 Sounds like a case for either  (1)  a different context per originating
 extension  (or maybe, per group of originating extensions which all happen to
 obey the same mathematical formula for determining outside callerID from
 inside extension number);  (2)  an AGI script, accessing a database which
 links internal extensions to external numbers; or  (3)  rethinking your
 internal extension numbering scheme so there is a consistent mapping from
 internal to external numbers, thus allowing you to do it all mathematically.

 --
 AJS

 Answers come *after* questions.

Why do programmers try to make solution so elegant when an entries for
each phone in sip.conf is all that is needed.

No need for mathematical formulas, AGIs, and databases.  You just took
over engineering to a new level.

Thanks,
Steve T

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Re: [asterisk-users] Callerid issue

[Doug Lytle]

Steve Totaro wrote:

For each phone, add callerid=Joe Smith1551212   no quotes in sip.conf


The problem with that solution is that station to station calls will 
show the same CID and not the extension.


I'd vote for the database approach.

Doug


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deserve neither Liberty nor Safety.


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Re: [asterisk-users] Callerid issue

[A J Stiles]

On Friday 10 Jun 2011, Steve Totaro wrote:
 Why do programmers try to make solution so elegant when an entries for
 each phone in sip.conf is all that is needed.

 No need for mathematical formulas, AGIs, and databases.  You just took
 over engineering to a new level.

Because doing it your way would cause the external caller ID number always 
to show up on the phone you dialled, even when making an internal call.

This probably is not what you want.

-- 
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Answers come *after* questions.

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Re: [asterisk-users] Callerid issue

[isrlgb]

-Original Message-
From: Steve Totaro stot...@asteriskhelpdesk.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 10 Jun 2011 06:30:53 
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Callerid issue

On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta maheshka...@flexydial.com wrote:
 Hi,
 I have 44578900 to 44578999 DID's. and I have extensions(100) for this
 DID's. but problem is
 callerid   Extensions
 44578900  100
 44578901  101
 44578902  102
 44578902  103
 44578903  104
 44578905  200
 44578906  275
 44578907  277
 44578908  354

 I need to setup the callerid with this extensions . for example whenever I
 am dial from 354 extension callerID will show 44578908.
 fro this scenarion I need logical dialplan because I have 100 extenstions ,
 so 100 extentions should be have different extensions.


 Below dialplan is for sequence callerid and extesions. like 101 to 199
 should callerid is going 44578900 to 44578999 .

 exten =_0X,1,NoOp(Int exten:${CALLERID(num)})
 exten =_0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
 exten =_0X,3,NoOp(Ext ident:${outgoing_ident})
 exten =_0X,4,Set(CALLERID(name)=${outgoing_ident})
 exten =_0X,5,AGI(agi://127.0.0.1:4577/call_log)
 exten =_0X,6,Set(CALLERID(num)=${outgoing_ident})
 exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
 exten =_0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
 exten =_0X,9,Hangup
 --
 Best Regards,

 Mahesh Katta
 BUZZWORKS Business Services Private Limited
 BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
 Mumbai 400069
 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
 Web http://www.buzzworks.com


Are you using SIP phones?  I assume you are but just checking.

Look into sip.conf

For each phone, add callerid=Joe Smith 1551212  no quotes in sip.conf

Be sure the telco allows it, some times they will only allow the BTN.
I have run into troubles with toll free callerid.  Everything made
sense because the problem was calling other toll free numbers and who
pays.  Good point.  Especially because ANI can be manipulated in
Asterisk as well.

I never understood hy people who have block of DIDs in a row choose to
make life difficult by not incrementing extensions by one, send caller
ID by prepending the common numbers and only sending four digits.

I guess if you get huge there could be duplicates of the last four.

Just set it in sip.conf

Thanks,
Steve T
Thanks
Steve Totaro

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Re: [asterisk-users] Callerid issue

[mahesh katta]

On Fri, Jun 10, 2011 at 4:00 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:

 On Fri, Jun 10, 2011 at 5:35 AM, mahesh katta maheshka...@flexydial.com
 wrote:
  Hi,
  I have 44578900 to 44578999 DID's. and I have extensions(100) for this
  DID's. but problem is
  callerid   Extensions
  44578900  100
  44578901  101
  44578902  102
  44578902  103
  44578903  104
  44578905  200
  44578906  275
  44578907  277
  44578908  354
 
  I need to setup the callerid with this extensions . for example whenever
 I
  am dial from 354 extension callerID will show 44578908.
  fro this scenarion I need logical dialplan because I have 100 extenstions
 ,
  so 100 extentions should be have different extensions.
 
 
  Below dialplan is for sequence callerid and extesions. like 101 to 199
  should callerid is going 44578900 to 44578999 .
 
  exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
  exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
  exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
  exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
  exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log)
  exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
  exten =
 
 _0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
  exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
  exten = _0X,9,Hangup
  --
  Best Regards,
 
  Mahesh Katta
  BUZZWORKS Business Services Private Limited
  BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
  201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
 (E)
  Mumbai 400069
  GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
  Web http://www.buzzworks.com
 

 Are you using SIP phones?  I assume you are but just checking.

 sir  I am using IP phones means sip id's only
[100]
username=100
secret=123
mailbox=100
type=friend
host=dynamic
canreinvite=no
context=default
qualify=yes



 Thanks,
 Steve T
 Thanks
 Steve Totaro

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-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] Callerid issue

[A J Stiles]

On Friday 10 Jun 2011, Steve Totaro wrote:
 I never understood hy people who have block of DIDs in a row choose to
 make life difficult by not incrementing extensions by one, send caller
 ID by prepending the common numbers and only sending four digits.

Well, to be fair, that's what most people usually start out trying to do -- 
make it all line up neatly, with each department having numbers in a certain 
range  (1xx for management, 2xx for purchasing, 3xx for sales, 4xx for IT, 
5xx for shop floor, 8xx as short codes for direct access to selected external 
numbers from phones that shouldn't normally have access to outside lines but 
still need to call certain numbers occasionally)  and so forth.

But then, once you have invested considerable time and effort devising a plan 
for allocating numbers, somebody On High inevitably makes a decision that 
ruins the whole thing.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Callerid issue

[mahesh katta]

On Fri, Jun 10, 2011 at 5:23 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:

 On Friday 10 Jun 2011, Steve Totaro wrote:
  I never understood hy people who have block of DIDs in a row choose to
  make life difficult by not incrementing extensions by one, send caller
  ID by prepending the common numbers and only sending four digits.

 Well, to be fair, that's what most people usually start out trying to do --
 make it all line up neatly, with each department having numbers in a
 certain
 range  (1xx for management, 2xx for purchasing, 3xx for sales, 4xx for IT,
 5xx for shop floor, 8xx as short codes for direct access to selected
 external
 numbers from phones that shouldn't normally have access to outside lines
 but
 still need to call certain numbers occasionally)  and so forth.

  yes sir there is some depart ments.

 But then, once you have invested considerable time and effort devising a
 plan
 for allocating numbers, somebody On High inevitably makes a decision that
 ruins the whole thing.

 --
 AJS

 Answers come *after* questions.

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-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] Callerid issue

[mahesh katta]

On Fri, Jun 10, 2011 at 5:23 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:

 On Friday 10 Jun 2011, Steve Totaro wrote:
  I never understood hy people who have block of DIDs in a row choose to
  make life difficult by not incrementing extensions by one, send caller
  ID by prepending the common numbers and only sending four digits.

 Well, to be fair, that's what most people usually start out trying to do --
 make it all line up neatly, with each department having numbers in a
 certain
 range  (1xx for management, 2xx for purchasing, 3xx for sales, 4xx for IT,
 5xx for shop floor, 8xx as short codes for direct access to selected
 external
 numbers from phones that shouldn't normally have access to outside lines
 but
 still need to call certain numbers occasionally)  and so forth.

  Yes sir, I have different departments and, different extensions.
and I am using sipphones only and have PRI line with 100 DID's which i
mention above.


But then, once you have invested considerable time and effort devising a
 plan
 for allocating numbers, somebody On High inevitably makes a decision that
 ruins the whole thing.

 --
 AJS

 Answers come *after* questions.

 --
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-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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[asterisk-users] CallerID issue

[virendra bhati]

Hi List,

I am making outgoing call from asterisk to GSM network with the help of VoIP
trunk(SIP trunk) then I am not geting any caller ID at destination end. Is
this the asterisk issue or VoIP trunk issue?
Is this is due to asterisk then how we solve it? I already user
Set(CALLERID(num)=XXX) in dialplan.




-
Thanks and regards

 Virendra Bhati
+91-9172341457
Asterisk Engineer
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Re: [asterisk-users] CallerID and URL pop up for windows...

[Gilles]

On Thu, 13 Jan 2011 17:59:10 -0500, Bruce B bruceb...@gmail.com
wrote:
http://bestof.nerdvittles.com/applications/screenpop/But better thing
would be to a have TAPI for outlook to query Outlook contact as well because
it allows for making notes on the contact. I am willing to pay for that if
it is added to URANG II

Has someone tried IdentaPop?

www.identafone.com/cidpop.html


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Re: [asterisk-users] CallerID and URL pop up for windows...

[Bruce B]

I tried a lot of these softwares in the past few days and lots of them are
just a pile of .. lots of compatibility issues with various versions of
Outlook and Windows or simply don't do either of inbound or outbound.
However, I have been testing Ingeniussoftware and their product so far works
with Inbound and pulls up Outlook contact. Haven't tried outbound.



On Fri, Jan 14, 2011 at 9:19 AM, Gilles codecompl...@free.fr wrote:

 On Thu, 13 Jan 2011 17:59:10 -0500, Bruce B bruceb...@gmail.com
 wrote:
 http://bestof.nerdvittles.com/applications/screenpop/But better thing
 would be to a have TAPI for outlook to query Outlook contact as well
 because
 it allows for making notes on the contact. I am willing to pay for that if
 it is added to URANG II

 Has someone tried IdentaPop?

 www.identafone.com/cidpop.html


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[asterisk-users] CallerID and URL pop up for windows...

[Carlos Chavez]

Anyone has a good recommendation for a Windows program that will open a
browser URL when your phone receives a call?  We had been using Yaacid
but since it is no longer being developed we need to look for an
alternative.  It should be light weight and work on all versions of
Windows.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] CallerID and URL pop up for windows...

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Thursday, January 13, 2011 11:37 AM
To: Asterisk
Subject: [asterisk-users] CallerID and URL pop up for windows...

Anyone has a good recommendation for a Windows program that will
open a
browser URL when your phone receives a call?  We had been using Yaacid
but since it is no longer being developed we need to look for an
alternative.  It should be light weight and work on all versions of
Windows.

This might be useful (or not)
http://articleresource.org/internet-and-businesses-online/web-hosting/use-de
sktop-pop-up-application-with-asterisk-pbx-111454

Unless you need a canned app, this would be an easy program to develop on
your own.  The easiest way (IMO) to do this would be to put a small
instance of Apache on your Asterisk server and run a CGI program that
interfaces to the local instance of Asterisk and pops a new window when a
call comes in.  This would have the added benefit of being self-contained on
the Asterisk machine (no programs to install or ports to open).


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