[asterisk-users] Can not get my Eicon Diva running with Asterisk...

[Michelle Konzack]

Hi Guys,

asterisk drive me crazy!

Now I have tried to use FreePBX but it require MySQL  which  I  can  not
install du to a conflict with PostgreSQL.

Does someone know, how to configure FreePBX to use PostgreSQL?

Or does someone know another Asterisk Web-Frontend, without Database?

It is realy not funny, to force users to install this monster on an  ARM
Microcontroller.

I need only enterprise internal stuff to

1)  access my 4 Vodafone EasyBox 803A using
ISDN and the Eicon Diva 4port v2 Server Card
2)  access a 1port HFC Card to connect some ISDN Telephones
3)  access my account (20 Telephone numbers)
on 
4)  let me use my CISCO CP-3905 SIP phones on the LAN
5)  access the VoIP server of my ISP Alice/Hansenet

I have no customers which must accounted or such...

Thanks, Greetings and nice Day/Evening
Michelle Konzack

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Re: [asterisk-users] Remote party ID - sort of working...

[Richard Mudgett]

> > 
> > exten => 123,1,Verbose(1,Test)
> > exten => 123,n,Set(CONNECTEDLINE(number,i)="555-555-")
> > exten => 123,n,Set(rclidname="TestingB <123-444->")
> 
> This line is just setting an ordinary channel variable.
> What do you think is supposed to use this value?
> 
> > exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)
> > exten => 123,n,Set(CONNECTEDLINE(name,i)="Testing")
> 
> Please read about usage of the ,i in [1].  If anything you should
> have the ,i on *all* of the CONNECTEDLINE lines *except* the last
> one before a Dial.
> 
> > exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)
> > exten => 123,n,Dial(SIP/joesmithpolycomphone,20)
> 
> Since you are not using the 'I' option on Dial here, your preset
> CONNECTEDLINE information is being overwritten by what is sent by
> SIP/joesmithpolycomphone when it answers.
> 
> > exten => 123,n,Hangup()
> 
> [1]
> https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information

> Why would you NOT want the connectedline info sent immediately?

So you don't have the channel driver's protocol needlessly
generating update traffic on partial information.  Have the
protocol update the whole party ID information at once instead
of piecemeal.  Depending upon the channel protocol involved,
connected line updates look like call transfers to the peer.

Richard

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Re: [asterisk-users] asterisk 1.8 on Solaris/sparc

[Jeremy Kister]

On 7/18/2012 2:27 AM, Jeremy Kister wrote:

I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.


.. ok, if the system weren't Solaris - let's say it was Debian Linux, 
what would be on the list of things to check for ?


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Re: [asterisk-users] Remote party ID - sort of working...

[Eric Wieling]

Why would you NOT want the connectedline info sent immediately?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 18, 2012 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Remote party ID - sort of working...

> I’m trying to set my system to set a caller id using the diaplan when 
> calling an internal extension. In other words, when I dial Joe Smith’s 
> extension I want my own phone to show “Joe Smith 555”. I have sort of 
> managed that in the sense that my phone shows Joe Smith’s caller id 
> based on his sip.conf callerid. But I need this to be done 
> programmatically through the dial plan (Let’s say I want to show “Joe 
> Smith” or just “Joe” based on some condition)
> 
> 
> 
> I have this in the relevant dialplan snippet:
> 
> 
> 
> exten => 123,1,Verbose(1,Test)
> exten => 123,n,Set(CONNECTEDLINE(number,i)="555-555-")
> exten => 123,n,Set(rclidname="TestingB <123-444->")

This line is just setting an ordinary channel variable.
What do you think is supposed to use this value?

> exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)
> exten => 123,n,Set(CONNECTEDLINE(name,i)="Testing")

Please read about usage of the ,i in [1].  If anything you should have the ,i 
on *all* of the CONNECTEDLINE lines *except* the last one before a Dial.

> exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)
> exten => 123,n,Dial(SIP/joesmithpolycomphone,20)

Since you are not using the 'I' option on Dial here, your preset CONNECTEDLINE 
information is being overwritten by what is sent by SIP/joesmithpolycomphone 
when it answers.

> exten => 123,n,Hangup()
> 
> 
> 
> I am always seeing remotepolycomphone’s callerid number and name as 
> entered in sip.conf, not “Testing 555-555-”, neither am I seeing 
> “TestingB <123-444->”.
> 
> 
> 
> What am I missing for it for accept my dialplan remote-id name and 
> number?

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

Richard

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Re: [asterisk-users] Remote party ID - sort of working...

[Doug Lytle]

>> exten => 124,n,Set(CONNECTEDLINE(all,i)="Name <555-555->") instead of a 
>> separate name and number priority.

An example of my line is:

Set(CONNECTEDLINE(all)="${cid.name}" <${ARG1}>)

Doug

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Re: [asterisk-users] Remote party ID - sort of working...

[Mike]

> > I m trying to set my system to set a caller id using the diaplan when
> > calling an internal extension. In other words, when I dial Joe Smith s
> > extension I want my own phone to show  Joe Smith 555 . I have sort of
> > managed that in the sense that my phone shows Joe Smith s caller id
> > based on his sip.conf callerid. But I need this to be done
> > programmatically through the dial plan (Let s say I want to show  Joe
> > Smith  or just  Joe  based on some condition)
> >
> 
> [1]
> https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Informa
> tion

Richard,

For future ref: I used the straight forward dial-through example, pretty much 
as is on the link you included (thank you!), but although the number worked 
fine the name did not. I had to use 

exten => 124,n,Set(CONNECTEDLINE(all,i)="Name <555-555->") instead of a 
separate name and number priority.

...so the wiki might be wrong (or there might be a bug in my 1.8.x version). 
But it works now, thanks to you for pointing me in the right direction,

Mike




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Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium

[Kevin P. Fleming]

On 07/18/2012 10:51 AM, Eric Wieling wrote:

Thank you.  While you are at it, ask them to document where the audio / data from " 
fax set g711cap| t38cap on" is saved to. 8-)


That is documented in the CLI help for the commands themselves; the 
capture files are placed into subdirectories of the main Asterisk log 
directory.


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Re: [asterisk-users] Remote party ID - sort of working...

[Richard Mudgett]

> I’m trying to set my system to set a caller id using the diaplan when
> calling an internal extension. In other words, when I dial Joe
> Smith’s extension I want my own phone to show “Joe Smith 555”. I
> have sort of managed that in the sense that my phone shows Joe
> Smith’s caller id based on his sip.conf callerid. But I need this to
> be done programmatically through the dial plan (Let’s say I want to
> show “Joe Smith” or just “Joe” based on some condition)
> 
> 
> 
> I have this in the relevant dialplan snippet:
> 
> 
> 
> exten => 123,1,Verbose(1,Test)
> exten => 123,n,Set(CONNECTEDLINE(number,i)="555-555-")
> exten => 123,n,Set(rclidname="TestingB <123-444->")

This line is just setting an ordinary channel variable.
What do you think is supposed to use this value?

> exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)
> exten => 123,n,Set(CONNECTEDLINE(name,i)="Testing")

Please read about usage of the ,i in [1].  If anything you should have
the ,i on *all* of the CONNECTEDLINE lines *except* the last one before
a Dial.

> exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)
> exten => 123,n,Dial(SIP/joesmithpolycomphone,20)

Since you are not using the 'I' option on Dial here, your preset
CONNECTEDLINE information is being overwritten by what is sent by
SIP/joesmithpolycomphone when it answers.

> exten => 123,n,Hangup()
> 
> 
> 
> I am always seeing remotepolycomphone’s callerid number and name as
> entered in sip.conf, not “Testing 555-555-”, neither am I seeing
> “TestingB <123-444->”.
> 
> 
> 
> What am I missing for it for accept my dialplan remote-id name and
> number?

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

Richard

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Re: [asterisk-users] Remote party ID - sort of working...

[Eric Wieling]

Remove the ",i" to start with.  Do you have the various rpid related options in 
sip.conf set?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, July 18, 2012 12:08 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Remote party ID - sort of working...

Hi,

 

I'm trying to set my system to set a caller id using the diaplan when calling 
an internal extension. In other words, when I dial Joe Smith's extension I want 
my own phone to show "Joe Smith 555".  I have sort of managed that in the sense 
that my phone shows Joe Smith's caller id based on his sip.conf callerid. But I 
need this to be done programmatically through the dial plan (Let's say I want 
to show "Joe Smith" or just "Joe" based on some condition)

 

I have this in the relevant dialplan snippet:

 

exten => 123,1,Verbose(1,Test)

exten => 123,n,Set(CONNECTEDLINE(number,i)="555-555-")

exten => 123,n,Set(rclidname="TestingB <123-444->")

exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)

exten => 123,n,Set(CONNECTEDLINE(name,i)="Testing")

exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)

exten => 123,n,Dial(SIP/joesmithpolycomphone,20)

exten => 123,n,Hangup()

 

I am always seeing remotepolycomphone's callerid number and name as entered in 
sip.conf, not "Testing 555-555-", neither am I seeing "TestingB 
<123-444->".

 

What am I missing for it for accept my dialplan remote-id name and  number? 

 

Regards,

 

Mike

 

 

 

 


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[asterisk-users] Remote party ID - sort of working...

[Mike]

Hi,

 

I'm trying to set my system to set a caller id using the diaplan when
calling an internal extension. In other words, when I dial Joe Smith's
extension I want my own phone to show "Joe Smith 555".  I have sort of
managed that in the sense that my phone shows Joe Smith's caller id based on
his sip.conf callerid. But I need this to be done programmatically through
the dial plan (Let's say I want to show "Joe Smith" or just "Joe" based on
some condition)

 

I have this in the relevant dialplan snippet:

 

exten => 123,1,Verbose(1,Test)

exten => 123,n,Set(CONNECTEDLINE(number,i)="555-555-")

exten => 123,n,Set(rclidname="TestingB <123-444->")

exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)

exten => 123,n,Set(CONNECTEDLINE(name,i)="Testing")

exten => 123,n,Set(CONNECTEDLINE(pres)=allowed)

exten => 123,n,Dial(SIP/joesmithpolycomphone,20)

exten => 123,n,Hangup()

 

I am always seeing remotepolycomphone's callerid number and name as entered
in sip.conf, not "Testing 555-555-", neither am I seeing "TestingB
<123-444->".

 

What am I missing for it for accept my dialplan remote-id name and  number? 

 

Regards,

 

Mike

 

 

 

 

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Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium

[Eric Wieling]

Thank you.  While you are at it, ask them to document where the audio / data 
from " fax set g711cap| t38cap on" is saved to. 8-)

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Wednesday, July 18, 2012 11:28 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium

On 07/18/2012 10:06 AM, Eric Wieling wrote:
> We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
>
> The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf 
> indicate v34  is supported, but when I enable it I get the message 
> "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored."  Is 
> v34 only supported with SpanDSP?

Those docs are in error. V.34 is not supported. I'll notify our documentation 
people. Thanks for the report.

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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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] Asterisk 1.8.13 / res_fax / res_fax_digium

[Kevin P. Fleming]

On 07/18/2012 10:06 AM, Eric Wieling wrote:

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13

The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34  is 
supported, but when I enable it I get the message "res_fax_digium.c:1624 
dgm_fax_new: V.34 not supported, will be ignored."  Is v34 only supported with 
SpanDSP?


Those docs are in error. V.34 is not supported. I'll notify our 
documentation people. Thanks for the report.


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

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[asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium

[Eric Wieling]

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13

The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf 
indicate v34  is supported, but when I enable it I get the message 
"res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored."  Is 
v34 only supported with SpanDSP?

Also, the res_fax.conf.sample does not indicate v34 as a valid modem.

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Re: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway

[Kevin P. Fleming]

On 07/18/2012 06:30 AM, Alejandro Recarey wrote:

Hi all, and thanks for taking the time to read this.

I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:

[fax]
exten => _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten => _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)

I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
these things work. When we send a fax:


You say they don't work, but you don't provide any details (console 
output, log messages, etc.) The configuration you have provided above is 
*required* for T.38 support and T.38 gateway mode. If it's not working, 
we are going to need more details about what is actually happening (if 
anything is at all).



1. Asterisk does NOT send a REINVITE with the t38 offered. Reading the
documentation, it should detect the fax tone with the audiohook and
then send a REINVITE with t38 capability.


This is expected behavior. Proper implementations of T.38 require that 
the gateway in front of the *called* endpoint monitor for FAX tones and 
initiate the switch to T.38 mode. In your configuration, that would be 
your carrier's gateway, assuming it is terminating the call to a 
non-T.38 endpoint. If your carrier is handing off the call to another 
SIP provider, then the responsibility lies with them, and so on.


However, Asterisk's T.38 gateway functionality should still detect the 
V.21 preamble generated by the called FAX endpoint and initiate a switch 
to T.38, if the carrier does not do it first. If this is not happening, 
we'll need to see logs and console output to figure out why. What codec 
are you using for your SIP calls?



2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
is not a problem if it correctly detects and REINVITES for faxes, but
our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we
do not have that capability.


This is bizarre; there is no specification anywhere that would indicate 
that a carrier should do this, and there are plenty of documents 
describing how it is a *bad* idea to offer a second media stream for 
T.38 in the initial INVITE of a call. I would urge you to ask them to 
reconsider this behavior.



Obviously I would prefer to just detect the fax myself and have
asterisk do the REINVITE.


This is not as reliable as the far-end gateway doing it, especially if 
the codec in use for the VoIP leg(s) of the call distorts the V.21 
preamble in any significant way.



I have read all of the documentation on the asterisk wiki (which is
rather short) and anything else I could find online. Unfortunately
most of it is out of date and refers to asterisk versions 1.4 to 1.8,
which do not have T38 Gateway capability.


The documentation on the wiki is short, but it's complete. Enabling T.38 
gateway functionality in Asterisk 10 is in fact pretty simple :-) 
Problems arise, as they always do in T.38-land, because no two T.38 
implementations are the same, and the choices made by carriers, 
gateway/softswitch/SBC manufacturers, and others, result in 
interoperability problems.


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Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)

[Ishfaq Malik]

On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote:
> Hi
> 
> I'm having a problem with the entirety of a call being recorded in the
> following scenario
> I'm using asterisk 1.8.7.0
> 
> Person A (asterisk peer) calls Person B (not on asterisk, real world
> number via a SIP trunk)
> Mixmonitor is invoked by Person A in the outbound context and
> AUDIOHOOK_INHERIT(MixMonitor)=yes is also set
> Person a transfers Person B to Person C (another asterisk peer)
> Person A is no longer involved in the call and the call is bridged
> between Person B and Person C
> 
> The call recording stops as soon as Person A hangs up, even though
> AUDIOHOOK_INHERIT is set
> 
> Is there any way we can get the entire call recorded in one file?
> 
> Thanks in advance
> 
> Ish

Has anyone else encountered this as it's becoming a real problem. Does
anyone know a way of getting continuity of call recording in this
scenario?

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Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls

[Ishfaq Malik]

On Wed, 2012-07-18 at 09:16 -0500, Matthew Jordan wrote:
> 
> - Original Message -
> > From: "Ishfaq Malik" 
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> > 
> > Sent: Wednesday, July 18, 2012 3:13:13 AM
> > Subject: Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and 
> > BUSY calls
> > 
> > On Wed, 2012-07-11 at 15:08 +0100, Ishfaq Malik wrote:
> > > Hi
> > > 
> > > I'm using asterisk 1.8.7
> > > 
> > > My dialplan for an inbound number is along the lines of
> > > 
> > > [default]
> > > exten => idenfier,1,Goto(specific-context,s,1)
> > > 
> > > [specific-context]
> > > exten => s,1,NoOp()
> > > exten => s,2,Dial(SIP/some-extenion,20)
> 
> 
> 
> So, I mocked up what is essentially the same scenario using the following
> dialplan:
> 
> [default]
> exten => 100,1,NoOp()
> same => n,Goto(new_context,s,1)
> 
> [new_context]
> exten => s,1,NoOp()
> same => n,Dial(SIP/phone_b,10)
> 
> Assume that I have two SIP peers, phone_a and phone_b, where phone_a dials
> extension 100 in context default.
> 
> In my case, I had phone_a first call phone_b, and did not answer with phone_b.
> That resulted in a disposition of NOANSWER.  I then had phone_a call phone_b
> and had phone_b explicitly reject phone_a's call; that resulted in a 
> disposition
> of BUSY.
> 
> In that case, with the csv and csv-custom CDR backends, I get the following 
> CDR
> records:
> 
> [csv]
> 
> "","phone_a","100","default","""phone_a"" 
> ","SIP/phone_a-","SIP/phone_b-0001","Dial","SIP/phone_b,10","2012-07-18
>  13:52:50",,"2012-07-18 13:53:00",10,0,"NO 
> ANSWER","DOCUMENTATION","1342619570.0",""
> "","phone_a","100","default","""phone_a"" 
> ","SIP/phone_a-0002","SIP/phone_b-0003","Dial","SIP/phone_b,10","2012-07-18
>  13:53:10",,"2012-07-18 13:53:16",6,0,"BUSY","DOCUMENTATION","1342619590.2",""
> 
> [csv-custom]
> 
> """phone_a"" 
> ","phone_a","100","default","SIP/phone_a-","SIP/phone_b-0001","Dial","SIP/phone_b,10","2012-07-18
>  08:52:50","","2012-07-18 08:53:00","10","0","NO 
> ANSWER","DOCUMENTATION","","1342619570.0","",0
> """phone_a"" 
> ","phone_a","100","default","SIP/phone_a-0002","SIP/phone_b-0003","Dial","SIP/phone_b,10","2012-07-18
>  08:53:10","","2012-07-18 
> 08:53:16","6","0","BUSY","DOCUMENTATION","","1342619590.2","",2
> 
> You'll note that in both, the destination context is "default".  So at the 
> very
> least, the records are consistent.
> 
> I ran this test using the latest from 1.8 (1.8.15.0-rc1) - since the version
> you're using is older and 1.8 has had some bug fixes with respect to CDRs
> since then, that might explain the discrepancy.
> 
> Why is destination context "default" and not "new_context"?  The destination 
> context
> is initially set when the CDR for the call is initialized - in which case, 
> its initial
> value is "default", since that's where the call entered.  When a CDR is 
> updated, the
> destination context is also updated to the channel's current context (or 
> macrocontext).
> As it is, a CDR update is not the same thing as the CDR being ended, and does 
> not always
> occur before the CDR is ended.  In the case you've outlined, an explicit CDR 
> update
> does not occur before the CDR record is ended - which results in the original 
> context
> of default being recorded as opposed to the current context of the channel, 
> "new_context".

That makes sense to me

> 
> 
> 
> > > Has anyone else experienced this? Is it actually correct behaviour
> > > and
> > > if so, why? If it is a bug, has it already been raised?
> 
> A quick search through JIRA would answer this question.  I don't believe 
> anyone has
> filed a bug related to this issue.
> 
> As far as it being the correct behavior - the behavior of the destination 
> context
> feels more implied then well defined.  Currently, the destination context is 
> updated
> when the source and destination channels are bridged, as opposed to when the 
> Dial
> is executed to the destination channel.  If the bridge never occurs, then the
> destination context is the original context of the source channel, since 
> that's the
> most information that is known at the time of CDR creation.  While that 
> behavior
> may not be what you desired, it is at least the current implementation.

Fair enough, now that I know why it behaves as it does, I can go about
making it behave the way I want it to.

Your answers have been very helpful and I really appreciate it.

> 
> > Thanks
> > 
> > Ish
> 
> 
> --
> Matthew Jordan
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
> 
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> To UNSUBSC

Re: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway

[Steve Underwood]

On 07/18/2012 09:43 PM, Matthew Jordan wrote:


- Original Message -

From: "Alejandro Recarey" 
To: "Asterisk Users Mailing List" 
Sent: Wednesday, July 18, 2012 6:30:26 AM
Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway

Hi all, and thanks for taking the time to read this.

I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:

[fax]
exten => _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten => _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)

The correct setting is not FAXOPT(t38gateway) - that is not a valid parameter
to pass to the FAXOPT function.  As you mention below, the correct setting
is Set(FAXOPT(gateway)=yes).  The optional timeout is fine.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_FAXOPT


I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
these things work. When we send a fax:

1. Asterisk does NOT send a REINVITE with the t38 offered. Reading
the
documentation, it should detect the fax tone with the audiohook and
then send a REINVITE with t38 capability.

Have you confirmed that Asterisk does not send the re-INVITE using either
a packet sniffer or by monitoring the log with 'sip set debug on'?  Without
seeing the SIP message traffic and a DEBUG log, its hard to say what
might be the cause of your issues.

Typically, I would expect to see something like the following in a DEBUG log:

[Jul 18 08:29:18] DEBUG[20234] res_fax.c: detected v21 preamble from 
SIP/ast1-g711-0001
[Jul 18 08:29:18] DEBUG[20234] res_fax.c: requesting T.38 for gateway session 
for SIP/ast1-t38-
  
Note that this also answers your question in a subsequent e-mail: you

should be using res_fax, with either res_fax_spandsp or Fax for Asterisk.


2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
is not a problem if it correctly detects and REINVITES for faxes, but
our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we
do not have that capability.

Obviously I would prefer to just detect the fax myself and have
asterisk do the REINVITE.

I have read all of the documentation on the asterisk wiki (which is
rather short) and anything else I could find online. Unfortunately
most of it is out of date and refers to asterisk versions 1.4 to 1.8,
which do not have T38 Gateway capability.

There typically isn't a lot of configuration that is needed for T.38
gateway support.  The necessary dialplan configuration is documented
here:

https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway

One thing that page doesn't mention is only spandsp supports T.38 
gateway right now. The Digium FAX module does not.


Regards,
Steve


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Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls

[Matthew Jordan]

- Original Message -
> From: "Ishfaq Malik" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Wednesday, July 18, 2012 3:13:13 AM
> Subject: Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY 
> calls
> 
> On Wed, 2012-07-11 at 15:08 +0100, Ishfaq Malik wrote:
> > Hi
> > 
> > I'm using asterisk 1.8.7
> > 
> > My dialplan for an inbound number is along the lines of
> > 
> > [default]
> > exten => idenfier,1,Goto(specific-context,s,1)
> > 
> > [specific-context]
> > exten => s,1,NoOp()
> > exten => s,2,Dial(SIP/some-extenion,20)



So, I mocked up what is essentially the same scenario using the following
dialplan:

[default]
exten => 100,1,NoOp()
same => n,Goto(new_context,s,1)

[new_context]
exten => s,1,NoOp()
same => n,Dial(SIP/phone_b,10)

Assume that I have two SIP peers, phone_a and phone_b, where phone_a dials
extension 100 in context default.

In my case, I had phone_a first call phone_b, and did not answer with phone_b.
That resulted in a disposition of NOANSWER.  I then had phone_a call phone_b
and had phone_b explicitly reject phone_a's call; that resulted in a disposition
of BUSY.

In that case, with the csv and csv-custom CDR backends, I get the following CDR
records:

[csv]

"","phone_a","100","default","""phone_a"" 
","SIP/phone_a-","SIP/phone_b-0001","Dial","SIP/phone_b,10","2012-07-18
 13:52:50",,"2012-07-18 13:53:00",10,0,"NO 
ANSWER","DOCUMENTATION","1342619570.0",""
"","phone_a","100","default","""phone_a"" 
","SIP/phone_a-0002","SIP/phone_b-0003","Dial","SIP/phone_b,10","2012-07-18
 13:53:10",,"2012-07-18 13:53:16",6,0,"BUSY","DOCUMENTATION","1342619590.2",""

[csv-custom]

"""phone_a"" 
","phone_a","100","default","SIP/phone_a-","SIP/phone_b-0001","Dial","SIP/phone_b,10","2012-07-18
 08:52:50","","2012-07-18 08:53:00","10","0","NO 
ANSWER","DOCUMENTATION","","1342619570.0","",0
"""phone_a"" 
","phone_a","100","default","SIP/phone_a-0002","SIP/phone_b-0003","Dial","SIP/phone_b,10","2012-07-18
 08:53:10","","2012-07-18 
08:53:16","6","0","BUSY","DOCUMENTATION","","1342619590.2","",2

You'll note that in both, the destination context is "default".  So at the very
least, the records are consistent.

I ran this test using the latest from 1.8 (1.8.15.0-rc1) - since the version
you're using is older and 1.8 has had some bug fixes with respect to CDRs
since then, that might explain the discrepancy.

Why is destination context "default" and not "new_context"?  The destination 
context
is initially set when the CDR for the call is initialized - in which case, its 
initial
value is "default", since that's where the call entered.  When a CDR is 
updated, the
destination context is also updated to the channel's current context (or 
macrocontext).
As it is, a CDR update is not the same thing as the CDR being ended, and does 
not always
occur before the CDR is ended.  In the case you've outlined, an explicit CDR 
update
does not occur before the CDR record is ended - which results in the original 
context
of default being recorded as opposed to the current context of the channel, 
"new_context".



> > Has anyone else experienced this? Is it actually correct behaviour
> > and
> > if so, why? If it is a bug, has it already been raised?

A quick search through JIRA would answer this question.  I don't believe anyone 
has
filed a bug related to this issue.

As far as it being the correct behavior - the behavior of the destination 
context
feels more implied then well defined.  Currently, the destination context is 
updated
when the source and destination channels are bridged, as opposed to when the 
Dial
is executed to the destination channel.  If the bridge never occurs, then the
destination context is the original context of the source channel, since that's 
the
most information that is known at the time of CDR creation.  While that behavior
may not be what you desired, it is at least the current implementation.
 
> > Thanks in advance
> > 
> >  
> > 
> Would I be better off asking this question of the dev community?

Nope, as this isn't an Asterisk development question.

> Thanks
> 
> Ish


--
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Digium, Inc. | Software Developer
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] Using Asterisk 10.6 as a T38 Fax gateway

[Matthew Jordan]

- Original Message -
> From: "Alejandro Recarey" 
> To: "Asterisk Users Mailing List" 
> Sent: Wednesday, July 18, 2012 6:30:26 AM
> Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
> 
> Hi all, and thanks for taking the time to read this.
> 
> I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
> receiving calls through the PSTN and want to send them to my VoIP
> carriers as T38. This is my dialplan:
> 
> [fax]
> exten => _X.,1,Set(FAXOPT(t38gateway)=yes,20)
> exten => _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)

The correct setting is not FAXOPT(t38gateway) - that is not a valid parameter
to pass to the FAXOPT function.  As you mention below, the correct setting
is Set(FAXOPT(gateway)=yes).  The optional timeout is fine.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_FAXOPT

> I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
> also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
> these things work. When we send a fax:
> 
> 1. Asterisk does NOT send a REINVITE with the t38 offered. Reading
> the
> documentation, it should detect the fax tone with the audiohook and
> then send a REINVITE with t38 capability.

Have you confirmed that Asterisk does not send the re-INVITE using either
a packet sniffer or by monitoring the log with 'sip set debug on'?  Without
seeing the SIP message traffic and a DEBUG log, its hard to say what
might be the cause of your issues.  

Typically, I would expect to see something like the following in a DEBUG log:

[Jul 18 08:29:18] DEBUG[20234] res_fax.c: detected v21 preamble from 
SIP/ast1-g711-0001
[Jul 18 08:29:18] DEBUG[20234] res_fax.c: requesting T.38 for gateway session 
for SIP/ast1-t38-
 
Note that this also answers your question in a subsequent e-mail: you
should be using res_fax, with either res_fax_spandsp or Fax for Asterisk.

> 2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
> is not a problem if it correctly detects and REINVITES for faxes, but
> our destination carriers tell us that they cannot do the REINVITE
> themselves because we do not offer t38 in our SDP, so they believe we
> do not have that capability.
> 
> Obviously I would prefer to just detect the fax myself and have
> asterisk do the REINVITE.
> 
> I have read all of the documentation on the asterisk wiki (which is
> rather short) and anything else I could find online. Unfortunately
> most of it is out of date and refers to asterisk versions 1.4 to 1.8,
> which do not have T38 Gateway capability.

There typically isn't a lot of configuration that is needed for T.38
gateway support.  The necessary dialplan configuration is documented
here:

https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway

> Does anybody have any experience in making this work?
> 
> Thank you!
> 
> Alex
> 


--
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Digium, Inc. | Software Developer
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] How to work around asterisk ss7

[Bharat Lalcheta]

You can use asterisk 1.6+ and libss7 for this functionality. Any
Digium or Sangoma card working ok on this setup. Currently i am using
it on both of them.

On Wed, Jul 18, 2012 at 5:14 PM, Ashish Agarwal  wrote:
> Hello,
>
> Can someone give me an understanding about E1 with ISUP on CCS 7 signalling?
> Is it possible with asterisk + digium card and how
>
> Regards,
>
> Ashish
>
>
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Re: [asterisk-users] How to work around asterisk ss7

[Mitul Limbani]

you need to either use chan_ss7 or libss7.

Also look for mailing list archives of asterisk-ss7

Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-61447605
Cell: +91-9820332422




On Wed, Jul 18, 2012 at 5:14 PM, Ashish Agarwal wrote:

> Hello,
>
> Can someone give me an understanding about E1 with ISUP on CCS 7
> signalling? Is it possible with asterisk + digium card and how
>
> Regards,
>
> Ashish
>
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> _
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[asterisk-users] How to work around asterisk ss7

[Ashish Agarwal]

Hello,

Can someone give me an understanding about E1 with ISUP on CCS 7
signalling? Is it possible with asterisk + digium card and how

Regards,

Ashish
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Re: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway

[Alejandro Recarey]

I forgot to ask:

Do I have to load "res_fax" or "app_fax" to use the T38 gateway capability?

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Re: [asterisk-users] Telecom HU cannot callforward to external number

[Christian Gansberger]

Unfortunately not, I already tried different forms callerid(num). Always
the same error.

I came across this entry in asterisk changelogs - maybe an update of
asterisk will help.

* Asterisk 1.4.36-rc1 Released.

2010-08-20 16:46 + [r283048-283123]  Richard Mudgett 

* channels/chan_dahdi.c: Merged revision 278274 from
  https://origsvn.digium.com/svn/asterisk/trunk .. r278274
  | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
  line Reference correct struct member for unlikely event
  PRI_EVENT_CONFIG_ERR. ..

* channels/chan_dahdi.c: Q931 - Sending PROGRESS after sending
  ALERTING is a protocol error The PRI layer in chan_dadhi will
  check if a PROGRESS message has already been sent, and not allow
  sending another (although that is technically allowed by the Q931
  spec), however it does not protect against sending an ALERTING
  and then sending a PROGRESS message, which is a violation of the
  specification. Most switches don't seem to care too deeply about
  this, but some do, and will disconnect the call when receiving
  this invalid sequence. Protocol specification reference:
  T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
  protocol control (network side) point-point (sheet 3 of 8)"
  (closes issue #17874) Reported by: nic_bellamy Patches:
  asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
  nic bellamy (license 299)
  asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
  by nic bellamy (license 299)
  asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
  by nic bellamy (license 299)


thx
christian

On 18 July 2012 13:07, Mitul Limbani  wrote:

> Mebe your operator doesnt like the CallerID(num) set as NULL just remove
> the callerid(num) statement and let the standard callerId get set by
> network.
>
> Regards,
> Mitul Limbani,
> Chief Architech & Founder,
> Enterux Solutions Pvt. Ltd.
> 110 Reena Complex, Opp. Nathani Steel,
> Vidyavihar (W), Mumbai - 400 086. India
> http://www.enterux.com/
> http://www.entvoice.com/
> email: mi...@enterux.in
> DID: +91-22-61447605
> Cell: +91-9820332422
>
>
>
>
> On Wed, Jul 18, 2012 at 4:23 PM, Christian Gansberger <
> christian.gansber...@accm.at> wrote:
>
>> Hi List!
>>
>> I have a Problem with Telecom Hungary, if I set a callforwarding on the
>> Snom, to an external number (mobile).
>> Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0,
>> snom-7.7.30
>>
>> When I call the Snom (Extension 68), it responds with "302 moved
>> temporarily", and Asterisk try to dial out over the LOCAL channel using
>> DAHDI.
>> I get a Congestion back from Telecom. Channel 0/2, span 1 got hangup
>> request, cause 21
>>
>>
>> Here is cli output:
>>
>>   -- Accepting call from 'callerid' to '68' on channel 0/1, span 1
>>  -- Executing [s@macro-station-fallback-Q-VM:5] Dial("DAHDI/1-1",
>> "SIP/68|15|tTW") in new stack
>> -- Called 68
>>
>> -- Got SIP response 302 "Moved Temporarily" back from 10.70.x.xxx
>>
>> -- Now forwarding DAHDI/1-1 to 'Local/*1mobilenr@snom68' (thanks to
>> SIP/68-76b8)
>> -- Executing [*1mobilenr@snom68:1]
>> Macro("Local/*1mobilenr@snom68-2fe3,2",
>> "dialout-dahdi-test|mobilenr|g1|") in new stack
>> -- Executing [s@macro-dialout-dahdi-test:1]
>> Set("Local/*1mobilenr@snom68-2fe3,2", "CALLERID(number)=") in new stack
>> -- Executing [s@macro-dialout-dahdi-test:2]
>> Dial("Local/*1mobilenr@snom68-2fe3,2", "DAHDI/g1/mobilenr||") in new
>> stack
>> -- Requested transfer capability: 0x00 - SPEECH
>> -- Called g1/mobilnr
>> -- DAHDI/2-1 is proceeding passing it to Local/*1mobilenr@snom68-2fe3
>> ,2
>>
>> -- Local/*1mobilenr@snom68-2fe3,1 is proceeding passing it to
>> DAHDI/1-1
>>
>> -- Channel 0/2, span 1 got hangup request, cause 21
>>
>> -- DAHDI/2-1 is circuit-busy
>>
>> -- Hungup 'DAHDI/2-1'
>>
>>   == Everyone is busy/congested at this time (1:0/1/0)
>>
>>
>> I have also an output from "pri intense debug" - But I think the Telecom
>> is just not accepting the outgoing call.
>> What do you think?
>>
>>
>> thanks
>> yours
>> christian
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
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> 

[asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway

[Alejandro Recarey]

Hi all, and thanks for taking the time to read this.

I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:

[fax]
exten => _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten => _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)

I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
these things work. When we send a fax:

1. Asterisk does NOT send a REINVITE with the t38 offered. Reading the
documentation, it should detect the fax tone with the audiohook and
then send a REINVITE with t38 capability.

2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
is not a problem if it correctly detects and REINVITES for faxes, but
our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we
do not have that capability.

Obviously I would prefer to just detect the fax myself and have
asterisk do the REINVITE.

I have read all of the documentation on the asterisk wiki (which is
rather short) and anything else I could find online. Unfortunately
most of it is out of date and refers to asterisk versions 1.4 to 1.8,
which do not have T38 Gateway capability.

Does anybody have any experience in making this work?

Thank you!

Alex

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Re: [asterisk-users] Telecom HU cannot callforward to external number

[Mitul Limbani]

Mebe your operator doesnt like the CallerID(num) set as NULL just remove
the callerid(num) statement and let the standard callerId get set by
network.

Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-61447605
Cell: +91-9820332422




On Wed, Jul 18, 2012 at 4:23 PM, Christian Gansberger <
christian.gansber...@accm.at> wrote:

> Hi List!
>
> I have a Problem with Telecom Hungary, if I set a callforwarding on the
> Snom, to an external number (mobile).
> Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0,
> snom-7.7.30
>
> When I call the Snom (Extension 68), it responds with "302 moved
> temporarily", and Asterisk try to dial out over the LOCAL channel using
> DAHDI.
> I get a Congestion back from Telecom. Channel 0/2, span 1 got hangup
> request, cause 21
>
>
> Here is cli output:
>
>   -- Accepting call from 'callerid' to '68' on channel 0/1, span 1
>  -- Executing [s@macro-station-fallback-Q-VM:5] Dial("DAHDI/1-1",
> "SIP/68|15|tTW") in new stack
> -- Called 68
>
> -- Got SIP response 302 "Moved Temporarily" back from 10.70.x.xxx
>
> -- Now forwarding DAHDI/1-1 to 'Local/*1mobilenr@snom68' (thanks to
> SIP/68-76b8)
> -- Executing [*1mobilenr@snom68:1] Macro("Local/*1mobilenr@snom68-2fe3,2",
> "dialout-dahdi-test|mobilenr|g1|") in new stack
> -- Executing [s@macro-dialout-dahdi-test:1]
> Set("Local/*1mobilenr@snom68-2fe3,2", "CALLERID(number)=") in new stack
> -- Executing [s@macro-dialout-dahdi-test:2]
> Dial("Local/*1mobilenr@snom68-2fe3,2", "DAHDI/g1/mobilenr||") in new stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called g1/mobilnr
> -- DAHDI/2-1 is proceeding passing it to Local/*1mobilenr@snom68-2fe3
> ,2
>
> -- Local/*1mobilenr@snom68-2fe3,1 is proceeding passing it to
> DAHDI/1-1
>
> -- Channel 0/2, span 1 got hangup request, cause 21
>
> -- DAHDI/2-1 is circuit-busy
>
> -- Hungup 'DAHDI/2-1'
>
>   == Everyone is busy/congested at this time (1:0/1/0)
>
>
> I have also an output from "pri intense debug" - But I think the Telecom
> is just not accepting the outgoing call.
> What do you think?
>
>
> thanks
> yours
> christian
>
>
>
>
>
>
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Telecom HU cannot callforward to external number

[Christian Gansberger]

Hi List!

I have a Problem with Telecom Hungary, if I set a callforwarding on the
Snom, to an external number (mobile).
Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0, snom-7.7.30

When I call the Snom (Extension 68), it responds with "302 moved
temporarily", and Asterisk try to dial out over the LOCAL channel using
DAHDI.
I get a Congestion back from Telecom. Channel 0/2, span 1 got hangup
request, cause 21


Here is cli output:

  -- Accepting call from 'callerid' to '68' on channel 0/1, span 1
 -- Executing [s@macro-station-fallback-Q-VM:5] Dial("DAHDI/1-1",
"SIP/68|15|tTW") in new stack
-- Called 68

-- Got SIP response 302 "Moved Temporarily" back from 10.70.x.xxx

-- Now forwarding DAHDI/1-1 to 'Local/*1mobilenr@snom68' (thanks to
SIP/68-76b8)
-- Executing [*1mobilenr@snom68:1] Macro("Local/*1mobilenr@snom68-2fe3,2",
"dialout-dahdi-test|mobilenr|g1|") in new stack
-- Executing [s@macro-dialout-dahdi-test:1]
Set("Local/*1mobilenr@snom68-2fe3,2", "CALLERID(number)=") in new stack
-- Executing [s@macro-dialout-dahdi-test:2]
Dial("Local/*1mobilenr@snom68-2fe3,2", "DAHDI/g1/mobilenr||") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/mobilnr
-- DAHDI/2-1 is proceeding passing it to Local/*1mobilenr@snom68-2fe3,2

-- Local/*1mobilenr@snom68-2fe3,1 is proceeding passing it to DAHDI/1-1

-- Channel 0/2, span 1 got hangup request, cause 21

-- DAHDI/2-1 is circuit-busy

-- Hungup 'DAHDI/2-1'

  == Everyone is busy/congested at this time (1:0/1/0)


I have also an output from "pri intense debug" - But I think the Telecom is
just not accepting the outgoing call.
What do you think?


thanks
yours
christian
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Re: [asterisk-users] Asterisk with OpenBTS and mobile phone

[Ellen Apolinar]

Hey Ioan,

thanks for your answer.

It helped a little bit but I have no idea what exactly could work wrong.

My new situation:

*CLI> originate SIP/123456789101112 application MusicOnHold

>   == Using SIP RTP CoS mark 5
> -- Got SIP response 482 "Loop Detected" back from 192.168.0.102:5060
> [Jul 18 10:38:27] WARNING[4615]: chan_sip.c:3873 __sip_autodestruct:
> Autodestruct on dialog '
> 446588d34c8b0e2d1920fec416ef0b5d@192.168.0.102:5060' with owner in place
> (Method: INVITE)
>

*CLI> sip show peers

> Name/username  HostDyn
> Forcerport ACL Port Status
> 123456789101112/6202   192.168.0.102
> N 5060 OK (1 ms)
> 6000/6000  192.168.0.102D
> N 5061 Unmonitored
> 6001/6001  192.168.0.102D
> N 5061 Unmonitored
>

*CLI> sip show channels

> Peer User/ANR Call ID  Format
> Hold Last MessageExpiry Peer
> 192.168.0.102(None)   2dab9ef669bc9a4  0x0 (nothing)
> No   Rx: OPTIONS
> 1 active SIP dialog
>

I thought with 6201 I could build a connection to Asterisk. In the
extensions.conf and in the Asterisk-GUI the numbers from 6000 - 6300 (not
all, just a frew of them) are shown so I choosed one of them like I did
with the softphones.

asterisk -rx doesn't work.

What do you think is wrong with my extensions.conf?

Best regards.
Ellen


On Fri, Jul 13, 2012 at 4:06 PM, Ioan Indreias  wrote:

> On Thu, Jul 12, 2012 at 3:55 PM, Ellen Apolinar
>  wrote:
> > Hello mailinglist,
> >
> > I want to connect Asterisk with OpenBTS and make a call with a mobile
> phone.
> >
> > I use:
> > Ubuntu 11.10 + Kernel 3.0.22
> > GnuRadio 3.3.0
> > Asterisk 1.8.13
> > OpenBTS 2.8
> > Nokia Mobile Phone
> >
> > OpenBTS works and I can send sms from the OpenBTS server to the
> > mobile phone. What I also need is a call between Asterisk and OpenBTS.
> >
> > I have also two soft phones which works with Asterisk. And also OpenBSC
> > is working with Asterisk successfully (OpenBSC is another project).
> >
> > Perhaps you can help me because I think it is an issue with Asterisk.
> >
> >
> > sip.conf:
> >>
> >> ;SIP-Phones (Twinkle)
> >> [user1]
> >> callerid = 6000
> >> username = 6000
> >> secret = 6000
> >> canreinvite = no
> >> type = friend
> >> context = phones
> >> allow = all
> >> host = dynamic
> >> dtmfmode = info
> >>
> >> [user2]
> >> callerid = 6001
> >> username = 6001
> >> secret = 6001
> >> canreinvite = no
> >> type = friend
> >> context = phones
> >> allow = all
> >> host = dynamic
> >> dtmfmode = info
> >>
> >> ; Mobile phone
> >> [123456789101112]
> >> callerid = 6201
> >> username = 6201
> >> secret = 6201
> >> canreinvite = no
> >> type = friend
> >> context = sip_external
> >> ;context = open-bts
> >> disallow = all
> >> allow = gsm
> >> host = 192.168.0.102
> >> domain = 192.168.0.102
> >> dtmfmode = info
> >
> >
> > extensions.conf
> >>
> >> [internal]
> >> exten => s,1,Verbose(1|Echo test application)
> >> exten => s,n,Echo()
> >> exten => s,n,Hangup()
> >> exten => 6000,1,Verbose(1|Extension 6000)
> >> exten => 6000,n,Dial(SIP/user1,30)
> >> exten => 6000,n,Hangup()
> >> exten => 6001,1,Verbose(1|Extension 6001)
> >> exten => 6001,n,Dial(SIP/user2,30)
> >> exten => 6001,n,Hangup()
> >>
> >> [phones]
> >> include => internal
> >> include => default
> >>
> >> [open-bts]
> >> exten => 6002,1,Playback(demo-echotest)
> >> exten => 6002,n,Echo
> >> exten => 6002,n,Playback(demo-echodone)
> >> exten => 6002,n,HangUp
> >>
> >> [sip_external]
> >> exten => 6201,1,Macro(dialGSM,123456789101112)
> >>
> >> [macro-dialGSM]
> >> exten => s,1,Dial(SIP/${ARG1},20)
> >> exten => s,n,Goto(s-${DIALSTATUS},1)
> >> exten => s-CANCEL,1,Hangup
> >> exten => s-NOANSWER,1,Hangup
> >> exten => s-BUSY,1,Busy(30)
> >> exten => s-CONGESTION,1,Congestion (30)
> >> exten => s-CHANUNAVAIL,1,Read(extension_digits,pbx-invalid)
> >> exten => s-CHANUNAVAIL,n,GoTo(open-bts,${extension_digits},1)
> >
> > I have tried both contexts, [open-bts] and [sip_external] and both don't
> > work
> >
> >
> > If I want to call the mobile phone (6201) with a Twinkle soft phone
> (6000)
> > I get following message in the CLI-window from Asterisk:
> >>
> >>  == Using SIP RTP CoS mark 5
> >> -- Executing [6201@DLPN_DialPlan1:1] Macro("SIP/6000-0013",
> >> "stdexten,6201,SIP/6201") in new stack
> >> -- Executing [s@macro-stdexten:1] Set("SIP/6000-0013",
> >> "__DYNAMIC_FEATURES=") in new stack
> >> [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyerror:
> >> ast_yyerror():  syntax error: syntax error, unexpected '=', expecting
> $end;
> >> Input:
> >>  = 1
> >>  ^
> >> [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:472 ast_yyerror: If
> you
> >> have questions, please refer to
> >> https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
> >> -- Ex

Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls

[Ishfaq Malik]

On Wed, 2012-07-11 at 15:08 +0100, Ishfaq Malik wrote:
> Hi
> 
> I'm using asterisk 1.8.7
> 
> My dialplan for an inbound number is along the lines of
> 
> [default]
> exten => idenfier,1,Goto(specific-context,s,1)
> 
> [specific-context]
> exten => s,1,NoOp()
> exten => s,2,Dial(SIP/some-extenion,20)
> 
> I have been testing the following 2 scenarios:
> 1) Caller calls in to identifier, caller hangs up (NO ANSWER)
> 2) Caller calls in to identifier, callee rejects (BUSY)
> 
> In both scenarios the dialplan works properly and dials
> 'some-extension'.
> However, there is some divergence with what is entered into the CDR. In
> both scenarios the following are the same (as they should be)
> a) lastapp
> b) lastdata
> 
> But, in scenario 1 the dcontext is 'specific-context' (this is what I
> would expect) and in scenario 2 the dcontext remains 'default' even
> though the call moved to a different context.
> 
> This cannot possibly be intentional and it is causing problems with our
> set up.
> 
> Has anyone else experienced this? Is it actually correct behaviour and
> if so, why? If it is a bug, has it already been raised?
> 
> Thanks in advance
> 
>  
> 
Would I be better off asking this question of the dev community?

Thanks

Ish

-- 
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, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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