Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

[Matthew Jordan]

On Wed, Jul 20, 2016 at 12:14 PM, Yves biganiro  wrote:
> Asterisk 1.8.23.0-1_centos5.go
>
> DAHDI Version: 2.6.1 Echo Canceller: HWEC
>

I'm fairly sure that GOautodial is a packaged solution based on vicidial:

http://goautodial.org/projects/goautodialce/wiki/goautodial_getting_started_guide

As a result, you will almost certainly need to solicit help from the
GOautodial folks. Things that are packaged up in such a fashion
typically have a specialized configuration that is too specific for
the Asterisk project itself to support.

Matt

-- 
Matthew Jordan
Digium, Inc. | CTO
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] No Sangoma ISDN BRI cards detected by goautodial

[Yves biganiro]

Asterisk 1.8.23.0-1_centos5.go

DAHDI Version: 2.6.1 Echo Canceller: HWEC

On Wed, Jul 20, 2016 at 5:32 PM, A J Stiles 
wrote:

> On Wednesday 20 Jul 2016, Yves biganiro wrote:
> > Hi all
> >
> > Hi,I'm facing a strange  issue where by SANGOMA not detected  by
> goautodial
> > system ,
>
> Is this some kind of one-stop, pre-prepared distribution with Linux,
> Asterisk,
> DAHDI, a web server and some custom scripts, that all installs from one
> place?
>
> We really need to know your Asterisk and DAHDI versions.
>
> Type in a root terminal,
>
> # asterisk -V
>
> and note the version number displayed  (it will be on the first line).
> Then
> enter
>
> *CLI> dahdi show version
>
> and note the DAHDI version displayed.
>
> --
> AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
> --
> _
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>http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

Yves Biganiro

Senior IT Consultant - independent

Tel +250727612605


##A tech entrepreneur and web developer, Passionate about technology with
working experience in web development.##
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Re: [asterisk-users] PJSIP_DIAL_CONTACTS issue

[Faheem Muhammad]

Thanks Richord and Carlos.


On Wednesday, 20 July 2016, Carlos Chavez  wrote:

> On 7/20/16 9:58 AM, Faheem Muhammad wrote:
>
> Hi,
> I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.
>
> When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial
> command breaks and the call control go to hangup block instead of next
> priority. The error in CLI says "*Dial requires an argument
> (technology/resource)*".
> This error seems legit as there are no contacts for an offline endpoint.
> The dialplan should jump to the next priority.
>
> exten => 1001,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
> exten => 1001,2,,NoOP(${DIALSTATUS})
> exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)
>
> exten => h,1,NoOp()
> exten => h,n,NoOP(${DIALSTATUS})
>
> ---
> If i try to dial the same offline endpoint with the below code snippet, it
> jumps to next prirorty.
> exten => 1001,1,Dial(PJSIP/${EXTEN})
> exten => 1001,2,,NoOP(${DIALSTATUS})
> exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)
>
> exten => h,1,NoOp()
> exten => h,n,NoOP(${DIALSTATUS})
>
> The endpoint may register from multiple device, so I always have to dial
> it all contacts. Did anyone else face such problem?
>
> My solution to this problem was to use a gotoif and check if
> PJSIP_DIAL_CONTACTS has any contacts before trying to dial, if it does not
> then I skip the dial and goto the next step.  So:
>
> exten => 1001,1,GotoIf($["${PJSIP_DIAL_CONTACTS(${EXTEN})}" = ""]?nocon)
> exten => 1001,n,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
> exten => 1001,n(nocon),SomethingElse
>
> --
>
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> +52 (55)9116-91161
>
>

-- 
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Re: [asterisk-users] PJSIP_DIAL_CONTACTS issue

[Carlos Chavez]

On 7/20/16 9:58 AM, Faheem Muhammad wrote:


Hi,
I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.

When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the 
dial command breaks and the call control go to hangup block instead of 
next priority. The error in CLI says "*Dial requires an argument 
(technology/resource)*".
This error seems legit as there are no contacts for an offline 
endpoint. The dialplan should jump to the next priority.


exten => 1001,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
exten => 1001,2,,NoOP(${DIALSTATUS})
exten => 1001,3,|Dial(PJSIP/mytrunk/sip:${mob||}||@10.0.0.1 
||)|


exten => h,1,NoOp()
exten => h,n,NoOP(${DIALSTATUS})

---
If i try to dial the same offline endpoint with the below code 
snippet, it jumps to next prirorty.

exten => 1001,1,Dial(PJSIP/${EXTEN})
exten => 1001,2,,NoOP(${DIALSTATUS})
exten => 1001,3,|Dial(PJSIP/mytrunk/sip:${mob||}||@10.0.0.1 
||)|

|
|
exten => h,1,NoOp()
exten => h,n,NoOP(${DIALSTATUS})

The endpoint may register from multiple device, so I always have to 
dial it all contacts. Did anyone else face such problem?


My solution to this problem was to use a gotoif and check if 
PJSIP_DIAL_CONTACTS has any contacts before trying to dial, if it does 
not then I skip the dial and goto the next step.  So:


exten => 1001,1,GotoIf($["${PJSIP_DIAL_CONTACTS(${EXTEN})}" = ""]?nocon)
exten => 1001,n,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
exten => 1001,n(nocon),SomethingElse

--


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

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Re: [asterisk-users] PJSIP_DIAL_CONTACTS issue

[Richard Mudgett]

On Wed, Jul 20, 2016 at 9:58 AM, Faheem Muhammad 
wrote:

> Hi,
> I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.
>
> When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial
> command breaks and the call control go to hangup block instead of next
> priority. The error in CLI says "*Dial requires an argument
> (technology/resource)*".
> This error seems legit as there are no contacts for an offline endpoint.
> The dialplan should jump to the next priority.
>
> exten => 1001,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
> exten => 1001,2,,NoOP(${DIALSTATUS})
> exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)
>
> exten => h,1,NoOp()
> exten => h,n,NoOP(${DIALSTATUS})
>
> ---
> If i try to dial the same offline endpoint with the below code snippet, it
> jumps to next prirorty.
> exten => 1001,1,Dial(PJSIP/${EXTEN})
> exten => 1001,2,,NoOP(${DIALSTATUS})
> exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)
>
> exten => h,1,NoOp()
> exten => h,n,NoOP(${DIALSTATUS})
>
> The endpoint may register from multiple device, so I always have to dial
> it all contacts. Did anyone else face such problem?
>

You need to examine if the returned dial string is empty in your dialplan.
PJSIP_DIAL_CONTACTS returns
an '&' separated list of available contacts.  If there are no contacts the
list is empty.  Dial doesn't like an
empty list.

Richard
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Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

[A J Stiles]

On Wednesday 20 Jul 2016, Yves biganiro wrote:
> Hi all
> 
> Hi,I'm facing a strange  issue where by SANGOMA not detected  by goautodial
> system , 

Is this some kind of one-stop, pre-prepared distribution with Linux, Asterisk, 
DAHDI, a web server and some custom scripts, that all installs from one place?

We really need to know your Asterisk and DAHDI versions.

Type in a root terminal,

# asterisk -V

and note the version number displayed  (it will be on the first line).  Then 
enter

*CLI> dahdi show version

and note the DAHDI version displayed.

-- 
AJS

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

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Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

[Yves biganiro]

I have  forcefully installed everything but it says that the card is not
found.

On Wed, Jul 20, 2016 at 5:05 PM, Yves biganiro 
wrote:

> Hi all
>
> Hi,I'm facing a strange  issue where by SANGOMA not detected  by
> goautodial system ,  Thats the problem :
>  Configuring ISDN BRI cards [A500/B700]
> 
>
> No Sangoma ISDN BRI cards detected
>
> Press any key to continue:
> 
> Configuring GSM cards [W400]
> 
>
> No Sangoma GSM cards detected
>
>
>
> regards
>
>
>
>
>


-- 

Yves Biganiro

Senior IT Consultant - independent

Tel +250727612605


##A tech entrepreneur and web developer, Passionate about technology with
working experience in web development.##
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[asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

[Yves biganiro]

Hi all

Hi,I'm facing a strange  issue where by SANGOMA not detected  by goautodial
system ,  Thats the problem :
 Configuring ISDN BRI cards [A500/B700]


No Sangoma ISDN BRI cards detected

Press any key to continue:

Configuring GSM cards [W400]


No Sangoma GSM cards detected



regards
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[asterisk-users] PJSIP_DIAL_CONTACTS issue

[Faheem Muhammad]

Hi,
I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.

When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial
command breaks and the call control go to hangup block instead of next
priority. The error in CLI says "*Dial requires an argument
(technology/resource)*".
This error seems legit as there are no contacts for an offline endpoint.
The dialplan should jump to the next priority.

exten => 1001,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
exten => 1001,2,,NoOP(${DIALSTATUS})
exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)

exten => h,1,NoOp()
exten => h,n,NoOP(${DIALSTATUS})

---
If i try to dial the same offline endpoint with the below code snippet, it
jumps to next prirorty.
exten => 1001,1,Dial(PJSIP/${EXTEN})
exten => 1001,2,,NoOP(${DIALSTATUS})
exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)

exten => h,1,NoOp()
exten => h,n,NoOP(${DIALSTATUS})

The endpoint may register from multiple device, so I always have to dial it
all contacts. Did anyone else face such problem?

Thanks!
Faheem
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Re: [asterisk-users] PJSIP and the pound (#) as %23

[Matthew Jordan]

On Wed, Jul 20, 2016 at 6:47 AM, Saint Michael  wrote:
>
> Is there any way to make PJSIP send the "#" as "#" and not as %23 in the 
> INVITE?
> I cannot figure this out.
>

The '#' character is a delimiter in URIs, and must be escaped if not
being used as such. Quoting RFC 2396, 2.4.3 [1]:

> The angle-bracket "<" and ">" and double-quote (") characters are
> excluded because they are often used as the delimiters around URI in
> text documents and protocol fields.  The character "#" is excluded
> because it is used to delimit a URI from a fragment identifier in URI
> references (Section 4). The percent character "%" is excluded because
> it is used for the encoding of escaped characters.
>
> delims  = "<" | ">" | "#" | "%" | <">

PJSIP is doing the "right thing" by escape encoding the reserved character.

[1] https://www.ietf.org/rfc/rfc2396.txt

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

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[asterisk-users] PJSIP and the pound (#) as %23

[Saint Michael]

Is there any way to make PJSIP send the "#" as "#" and not as %23 in the
INVITE?
I cannot figure this out.
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[asterisk-users] 1.8.32.3 - billsec field does not increment after call answer - what triggers it?

[Stefan Viljoen]

Hi Guys

I've got a strange problem - on my asterisk instance, when a call starts to
ring, I do 

core show channel  

and I get the usual output with the duration and billsec fields included.

For most of my calls, things are normal, e. g. duration field starts
incrementing as the SIP phone rings, and the moment it is answered / the
call goes offhook the duration timer continues running, and the billsec
timer starts up. Disposition goes from NO ANSWER to ANSWERED the moment
billsec starts incrementing.

However, for certain calls from a certain SIP trunk provided by a local
trunk provider, this never happens.

E. g. the call comes in on this "problem trunk" and duration timer starts
running - RTP starts and the call is totally normal, both parties have
crystal clear bi-directional audio and the call records correctly - but the
billsec timer never starts incrementing and forever remains at 0.
Disposition forever remains at NO ANSWER -  even though the call is in
progress and has been answered, and is working perfectly.

Other calls from other trunks provided by the same provider on the same
logical and physical Asterisk instance work correctly - if the call is
answered, it becomes ANSWERED in "core show channel" display, and the
billsec timer starts incrementing.

Only this one trunk consistenly has this problem for all calls received over
it. The trunk provider is using sippy on their side.

What setting / config option for the particular SIP "problem trunk" have my
trunk provider changed on their side to stop Asterisk from recognising that
a call has been answered when it comes in over that trunk?

It appears some SIP traffic is not being sent by them (or not received by my
Asterisk) that indicates to it a call has been ANSWERED and that it must
start the billsec timer?

Thanks!

Stefan



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