[asterisk-users] SIP realm

[jonas kellens]

Can I define the realm on a per peer basis ??
Can I define a realm to be used for one peer and another realm for
another peer in sip.conf ??

I have an ITSP that I need to authenticate with a realm that they set.
But this realm is not valuable for other peers.

Jonas.
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Re: [asterisk-users] Showing "name of extension" when calling

[Tilghman Lesher]

On Wednesday 23 December 2009 01:31:04 Zhang Shukun wrote:
> 2009/12/23 Tilghman Lesher :
> > On Tuesday 22 December 2009 23:08:36 Rob Hillis wrote:
> >> On 12/23/09 12:23, Russell Bryant wrote:
> >> >>> Wasn't this scheduled for 1.6.2?
> >> >>
> >> >> I don't believe so, but I could be mistaken. It's certainly not in
> >> >> 1.6.2 as best I can tell from looking over the source code :-)
> >> >
> >> > Nope, it's not in 1.6.2.  It went into trunk after the 1.6.2 feature
> >> > freeze.
> >>
> >> Wouldn't that imply that it will be in 1.6.3?
> >
> > What was once going to be 1.6.3 is now going to be 1.8.  There is no
> > 1.6.3 in the planning stages at this time.
>
> why the edition jump from 1.6.2 to 1.8 , what's the reason? the number
> of the edition always
>
> confuse me.

http://blogs.asterisk.org/2009/11/10/asterisk-project-update-astricon-2009/

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Showing "name of extension" when calling

[Zhang Shukun]

why the edition jump from 1.6.2 to 1.8 , what's the reason? the number
of the edition always

confuse me.

2009/12/23 Tilghman Lesher :
> On Tuesday 22 December 2009 23:08:36 Rob Hillis wrote:
>> On 12/23/09 12:23, Russell Bryant wrote:
>> >>> Wasn't this scheduled for 1.6.2?
>> >>
>> >> I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
>> >> as best I can tell from looking over the source code :-)
>> >
>> > Nope, it's not in 1.6.2.  It went into trunk after the 1.6.2 feature
>> > freeze.
>>
>> Wouldn't that imply that it will be in 1.6.3?
>
> What was once going to be 1.6.3 is now going to be 1.8.  There is no 1.6.3
> in the planning stages at this time.
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
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-- 
Regards,
Sucan

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Re: [asterisk-users] Showing "name of extension" when calling

[Tilghman Lesher]

On Tuesday 22 December 2009 23:08:36 Rob Hillis wrote:
> On 12/23/09 12:23, Russell Bryant wrote:
> >>> Wasn't this scheduled for 1.6.2?
> >>
> >> I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
> >> as best I can tell from looking over the source code :-)
> >
> > Nope, it's not in 1.6.2.  It went into trunk after the 1.6.2 feature
> > freeze.
>
> Wouldn't that imply that it will be in 1.6.3?

What was once going to be 1.6.3 is now going to be 1.8.  There is no 1.6.3
in the planning stages at this time.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] E1 R2 Congestion Status

[Bruce Nik]

Hello,

You are dialing 00223344 with what you show: "DAHDI/g1/00223344"

That is not a real PSTN number in any country as for as I know. Do you have
the proper outbound route setup? Is your outbound route stripping digits?!

-Bruce



On Tue, Dec 22, 2009 at 9:08 AM, Khaled W Chehab wrote:

>   I have a 'CONGESTION' Status with R2 protocol.
>
> While testing this scenario sip GW--àAsterisk –Digium E1 R2 ProtocolàCisco
> E1 R2 protocolàsip Gw
>
> Find below my error and configuration ,where are the errors in my
> configuration ?
>
>
>
> =
>
> Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on
> rev-212-98-156-56 (pid = 3614)
>
> Verbosity is at least 3
>
>   == Using SIP RTP CoS mark 5
>
> -- Executing [00223...@default:1] Dial("SIP/98.34.56.216-000e",
> "DAHDI/g1/00223344") in new stack
>
> [Dec 22 06:02:49] WARNING[4756]: app_dial.c:1745 dial_exec_full: Unable to
> create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
>
>   == Everyone is busy/congested at this time (1:0/1/0)
>
> -- Auto fallthrough, channel 'SIP/98.34.56.216-000e' status is
> 'CONGESTION'
>
>
>
> Cisco Gateway
>
> show controller
>
> Controller E1 slot(0)/port(0)
>
>E1 Link is UP
>
>   No Alarm detected.
>
>   Applique type is Channelized E1.
>
>   Framing is CRC4, Line Code is HDB3.
>
>   Signalling type is R2-MFC.
>
>   0 Line Code Violations, 0 Framing Bit Errors
>
>   0 Far End Block Errors, 0 CRC Errors
>
>signalling type = r2
>
>clock source = slave
>
>channel group 0 = 1-31
>
>   1 2 3
>
>allocated timeslots = YYYNYYY
>
>outgoing barred channel group =
>
>channel order = ascending
>
>b-channel negotiation = exclusive
>
>overlap receiving by forced = disabled
>
>overlap sending by forced = disabled
>
>protocol side = network
>
>R2 get calling number = none
>
>ISDN virtual connect = disabled
>
>ISDN Layer 2 is DOWN
>
>ISDN Values
>
>   ISDN Layer 2 values
>
>  k= 7
>
>  N200 = 3
>
>  N201 = 260
>
>  T200 = 1 seconds
>
>  T203 = 10 seconds
>
>   ISDN Layer 3 values
>
>  T301 = 180 seconds
>
>  T303 = 4 seconds
>
>  T304 = 20 seconds
>
>  T305 = 30 seconds
>
>  T306 = 30 seconds
>
>  T308 = 4 seconds
>
>  T310 = 10 seconds
>
>  T313 = 10 seconds
>
>  T316 = 120 seconds
>
>  T322 = 4 seconds
>
>  T309 = 90 seconds
>
>  N303 = 1
>
>
>
> ---
>
> /etc/asterisk/chan_dahdi.conf
>
> [trunkgroups]
>
> signalling=mfcr2
>
> mfcr2_variant=mx
>
> trunkgroup => 1,16
>
> spanmap => 1,1,1
>
>
>
> [channels]
>
> signalling=mfcr2
>
> mfcr2_variant=mx
>
> context=default
>
> signalling=mfcr2
>
> mfcr2_variant=mx
>
> signalling=mfcr2
>
> mfcr2_variant=mx
>
> usecallerid=yes
>
> callwaiting=yes
>
> usecallingpres=yes
>
> callwaitingcallerid=yes
>
> threewaycalling=yes
>
> transfer=yes
>
> canpark=yes
>
> cancallforward=yes
>
> callreturn=yes
>
> echocancel=yes
>
> echocancelwhenbridged=yes
>
>
>
> group=1
>
> callgroup=1
>
> pickupgroup=1
>
> callerid = asreceived
>
> useincomingcalleridondahditransfer = yes
>
> tonezone = 0 ; 0 is US
>
> channel => 1-15,17-31
>
> signalling=mfcr2
>
> mfcr2_variant=itu
>
> mfcr2_max_ani=7
>
> mfcr2_max_dnis=8
>
> mfcr2_get_ani_first=no
>
> mfcr2_category=national_subscriber
>
> mfcr2_logdir=span1
>
> mfcr2_logging=all
>
>
>
> ;EOF
>
> cat /etc/dahdi/system.conf
>
> # Autogenerated by /usr/sbin/dahdi_genconf on Tue Dec 22 01:59:02 2009
>
> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
>
> # your manual changes will be LOST.
>
> # Dahdi Configuration File
>
> #
>
> # This file is parsed by the Dahdi Configurator, dahdi_cfg
>
> #
>
> # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 RED
>
> span=1,1,0,cas,hdb3,crc4
>
> # termtype: te
>
> #bchan=1-15,17-31
>
> #dchan=16
>
> cas=1-15:1101
>
> dchan=16
>
> cas=17-31:1101
>
> echocanceller=mg2,1-15,17-31
>
>
>
> # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS/CRC4 RED
>
> span=2,2,0,ccs,hdb3,crc4
>
> # termtype: te
>
> bchan=32-46,48-62
>
> dchan=47
>
> echocanceller=mg2,32-46,48-62
>
>
>
> # Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3" HDB3/CCS/CRC4 RED
>
> span=3,3,0,ccs,hdb3,crc4
>
> # termtype: te
>
> bchan=63-77,79-93
>
> dchan=78
>
> echocanceller=mg2,63-77,79-93
>
>
>
> # Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" HDB3/CCS/CRC4 RED
>
> span=4,4,0,ccs,hdb3,crc4
>
> # termtype: te
>
> bchan=94-108,110-124
>
> dchan=109
>
> echocanceller=mg2,94-108,110-124
>
>
>
> # Global data
>
>
>
> loadzone= us
>
> defaultzone = us
>
> [default]
>
> exten => _X.,1,Dial(DAHDI/g1/${EXTEN})
>
>
>
>
>
>
> --
> *
>
>

[asterisk-users] Session Refresh or Codec change

[Prashantm]

Hi,

How asterisk distinguish whether the re-invite is for codec change or for 
a session refresh? I know that it checks the session version and decides 
the same. But even if session version is different from the initial invite 
and but it has the same codec, asterisk identifies that it is a session 
refresh and does not pass this INVITE to the other end. 

Whether asterisk compares complete SDP? Is it not a burden on the system? 

Regards,
Prashant MurthyDisclaimer : This message is proprietary to Smartlink Network 
Systems Limited and is intended solely for the use of the individual to whom it 
is addressed. It may contain privileged or confidential information and should 
not be circulated or used for any purpose other than for what it is intended. 
If you have received this message in error, please notify the originator 
immediately. If you are not the intended recipient, you are notified that you 
are strictly prohibited from using, copying, altering, or disclosing the 
contents of this message. Smartlink Network Systems Ltd. accepts no 
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Re: [asterisk-users] Showing "name of extension" when calling

[Rob Hillis]

On 12/23/09 12:23, Russell Bryant wrote:
>>> Wasn't this scheduled for 1.6.2?
>> I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
>> as best I can tell from looking over the source code :-)
>> 
> Nope, it's not in 1.6.2.  It went into trunk after the 1.6.2 feature freeze.
>   

Wouldn't that imply that it will be in 1.6.3?

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Re: [asterisk-users] Showing "name of extension" when calling

[Russell Bryant]

On 12/22/09 6:00 PM, Kevin P. Fleming wrote:
> Doug Lytle wrote:
>> Kevin P. Fleming wrote:
>>> This is called Connected Party information display, and it will be in
>>> Asterisk 1.8.
>>>
>>>
>> Wasn't this scheduled for 1.6.2?
>
> I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
> as best I can tell from looking over the source code :-)
>

Nope, it's not in 1.6.2.  It went into trunk after the 1.6.2 feature freeze.

-- 
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
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] Showing "name of extension" when calling

[Kevin P. Fleming]

Doug Lytle wrote:
> Kevin P. Fleming wrote:
>> This is called Connected Party information display, and it will be in
>> Asterisk 1.8.
>>
>>
> Wasn't this scheduled for 1.6.2?

I don't believe so, but I could be mistaken. It's certainly not in 1.6.2
as best I can tell from looking over the source code :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] call queue with external numbers??

[C. Chad Wallace]

At 5:01 PM on 22 Dec 2009, Oguzhan Kayhan wrote:

> Hello,
> Our asterisk is connected to an ericsson pbx by PRI.
> What i want is the asterisk clients should call operator numbers by
> dialing 0
> 
> But, when a call is made to ericsson via number 0, it assumes that the
> call is made from outside, so it doesnt allow to be dialed.
> There are 3 real operator extensions which is grouped by ericsson for
> operators. Lets assume  1112 1113.
> 
> What i want to know is, is there a way for me to create such group in
> asterisk and add that external extension numbers which should be
> dialed by order, or by 3 rings at a time etcso that i can create
> that operator group on asterisk side also.
> 
> PS: I can call real extensions on ericsson without a problem.

How about this:

exten => 0,1,Dial(DAHDI/G1/,18)
exten => 0,n,Dial(DAHDI/G1/1112,18)
exten => 0,n,Dial(DAHDI/G1/1113,18)

...where DAHDI/G1 is the PRI connected to the ericsson (group=1 in
chan_dahdi.conf), and 18 seconds is 3 rings.

You might be able to use Queue(), but I'm not sure if you can add a
hunt group and external number as a queue member--you might have to use
the Local channel for that.


-- 

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



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Re: [asterisk-users] asterisk & x-lite

[Roman Pahuacho Bonilla]

serach the option en sip.conf:

externip = you public ip
localnet=tus direcciones locales (address local)

saludos

Roman

On Tue, Dec 22, 2009 at 4:26 AM, zehra yildiz  wrote:

> Hello All,
>
> I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
> softphone can call the other one but I can' t hear any voice. My
> configuration files are below:
>
> [r...@localhost asterisk]# cat sip.conf
> [general]
> canreinvite=yes
>
> [1001]
> username=1001
> password=1001
> type=friend
> context=phones
> host=dynamic
>
> [1002]
> callerid=1002
> username=1002
> password=1002
> type=friend
> context=phones
> host=dynamic
>
> [r...@localhost asterisk]# cat extensions.conf
> [globals]
>
> [general]
> autofallthrough=yes
>
> [default]
>
> [incoming_calls]
>
> [phones]
> exten => _1XXX,1,NoOp()
> exten => _1XXX,n,Dial(SIP/${EXTEN},30)
> exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
> exten => _1XXX,n,Hangup()
>
>
> PS: My sip server and softphones are in the same network subnet. There are
> not any firewall or iptables rules. I tried the "nat=yes" parameter but no
> changes.
>
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Re: [asterisk-users] anonymous calls code

[C F]

Huge thanks for mentioning what type of channel you are using.

On Tue, Dec 22, 2009 at 5:11 AM, Giorgio Incantalupo
 wrote:
> Hi C F,
>
> I solved the problem!! It was under my nose...
> If you are interested the solution is here:
> http://www.misdn.org/index.php/FAQ_chan_mISDN
> The right section is: "key pad elements"
>
> Giorgio Incantalupo
>
> C F wrote:
>> You would have to create a dialplan for it.
>> If your provider expects *67 (which is the case here with I/CLEC POTS)
>> then you would create something like:
>> exten => _*67[2-9]XX,1,Dial(Zap/g1/${EXTEN})
>> In the case of PRI you would use:
>> exten => _*67[2-9]XX,1,SetCallerPres(prohib)
>> exten => _*67[2-9]XX,2,Dial(Zap/g1/${EXTEN:3})
>>
>>
>>
>>
>> On Mon, Dec 21, 2009 at 6:19 AM, Giorgio Incantalupo
>>  wrote:
>>
>>> Hi all,
>>>
>>> does anybody know how to make on-demand anonymous calls? I've tried code
>>> *67# before the number to call but it is working with some providers only.
>>>
>>> Any hints?
>>>
>>> Thank you.
>>>
>>>
>>>
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>>>
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>
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Re: [asterisk-users] TDM 400 hardware(?) issue

[Greg Woods]

>  the machine will lock up because the TDM board or the Dahdi
> driver goes south. /var/log/messages starts filling up with repeated
> messages:
> 
> kernel: TDM PCI Master abort

Thank you to everyone who has taken the time to reply.

First I am going to apply the "fix what you know is broken" principle. I
have been having issues with the display as well. Since this machine is
primarily a server, and the only thing I use the console for is Amarok
(it is the house music player), the fact that the text consoles stop
working has never been a big issue. CTRL-ALT-F2 produces a blank screen,
but the console is really there because I can log in as root and type
commands that do get executed, although it's a bit tricky when I can't
see what I'm typing. So something is wrong with the display. The video
card is a PCI-E card. I don't know how directly the E bus connects to
the regular PCI bus, but it's entirely possible that a flaky video card
is the whole problem. So I replaced it. Too early to tell if that helps;
if I go a few days without any more issues, I'll know that was it. It
may not fix the issue, but it was easy and cheap and it needed to be
tried anyway for other reasons.

This is the price I pay for running a home system where cost is a major
issue. I can't afford to have special-purpose servers one for each use;
not only is money an issue, but also space and power, so my servers
fulfill multiple functions. I realize this is not ideal, and that it
would be much better to have the asterisk server dedicated to nothing
else, but that isn't a realistic option for me. Up until recently it has
always been rock solid.

Steve Totaro  wrote:


> In light of your budget issues, I would switch to quality SIP provider
> and have my numbers ported.

That is a possibility, but finding a quality SIP provider is one issue.
I use Teliax over IAX as a backup but have never gotten caller ID to
work properly and the sound quality just isn't as good as my POTS line.
The other issue is that this puts all my eggs in the Comcast basket. I
am reluctant to remove my Qwest land line and have everything depend on
Comcast. This also makes it impossible to use the phones without the
server machine being up (and asterisk being up), also undesirable
(although using the cell phones in that situation is also a
possibility). But it is an option. It is a more attractive option if the
alternative turns out to be replacing expensive hardware.

Lastly there is the WAF (Wife Acceptance Factor), that always has to be
considered for home projects. She is the one who doesn't like the sound
quality of Teliax. Another provider MIGHT be better but I have no way of
knowing, and she won't be happy knowing that we don't have an easy
workaround for the house phones if asterisk is down. So this provides
another motivation to keep the POTS line if I can do it without a major
expense. As far as she is concerned, we don't need asterisk. I like it
because I'm a geek and it's cool, and she's OK with that as long as it
works. She does enjoy having her messages e-mailed to her, having
separate mailboxes, etc., so she does understand the value of these geek
projects of mine (MythTV is cool too, a single button press to miss all
the annoying commercials), but first and foremost it has to be reliable
and at least fulfill the basic functionality. But the WAF has certainly
been dropping lately due to all the problems I am having.


> > Other options are going back to old versions of Asterisk.  What
> version
> > are you running? 

I am already running 1.4 because I have encountered this bug with 1.6:

https://issues.asterisk.org/view.php?id=15129

That pretty much prevents inbound calls from working, so I have already
had to go back to 1.4 . I am using the asterisk14-1.4.26.3-87 version
from ATrpms. I have thought about trying 1.6 with the old zaptel
drivers, but that isn't any better as a workaround than what I am
already doing, so I haven't gotten around to trying it yet.


Darrick Hartman  wrote:

> Why don't you contact Digium tech support? 

I have hesitated to do that because my card is fairly old now, but I
would certainly do that before permanently abandoning my POTS line or
replacing expensive hardware.

Steve Tatoro and Bruce Nik recommended Sangoma cards. That is only
important if it comes down to replacing the hardware at cost. I am
hoping to avoid that.

Tilghman Lesher  wrote:


> You could try purchasing just the base TDM410 card and move your old
> modules
> over from the old card to the new.  A little looking around has
> revealed
> somebody selling a "like new" card for $139:

Thank you for that information; I didn't know that was an option. My
fear with something like this is that if there is a hardware problem
that necessitates replacing the card, I don't really have any way of
knowing if the problem is in the base card or in one of the modules, so
I might end up doing a lot of work and not fixing the issue.


> Full disclosure, here:  I do work

Re: [asterisk-users] E1 R2 Congestion Status

[Moises Silva]

On Tue, Dec 22, 2009 at 9:08 AM, Khaled W Chehab wrote:

>   I have a 'CONGESTION' Status with R2 protocol.
>
> While testing this scenario sip GW--àAsterisk –Digium E1 R2 ProtocolàCisco
> E1 R2 protocolàsip Gw
>
> Find below my error and configuration ,where are the errors in my
> configuration ?
>

Typically you will be better off in the asterisk-r2 mailing list. Your
message is more easily spotted by R2 people.

First thing is to check you have green status in your E1, do you? and make
sure you have the right clock settings, I never configured a cisco but it
seems you configured the cisco to be a slave and I don't see any port in
DAHDI system.conf with master clock settings. (span=1,0,0...etc)

After that, enable R2 call logging using mfcr2_call_files=yes and pastebin
the generated call file (if any), if no file is generated it means the
problem is not at R2 but in your local trunk settings (may be dialing in the
wrong group or something).

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
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[asterisk-users] Account Code Inbound

[Peder]

I am trying to track inbound and outbound calls by user.  In sip.conf, I can
add an account code so that all outbound calls from user1 have that as the
accountcode in CDR, so that works fine.  For inbound, if someone calls user1
direct, I can set the account code in the dial plan like this and it works
fine:

exten => 700,1,Set(CDR(accountcode)=BOB)
exten => 700,2,Dial(SIP/BOB)

The problem is if a calls comes in and rings several phones at once, there
is no way to set the account code for each user that I can see.  Does
anybody have any idea on how to do this?  Here is an example:

exten => 799,1,Dial(SIP/BOB&SIP/MARY&SIP/DAVE&SIP/TOM)


To get the data I could just search by destination number of BOB, but in the
example above the destination number in the CDR is 799, not BOB since that
is the number called.  I could also search dstchannel as that will show what
I want, but I was hoping there was a more generic way to do it (my dialplan
is a lot more complex than that listed above and each user has 3-4 lines
like 799-BOB, 798-BOB, 797-BOB, so a query to find all of the calls for BOB
gets ugly very quickly).

Peder



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Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?

[Taylor, Jonn]

Dan Journo wrote:


I recommend you follow the detailed install guide in this book and 
install all the required support programs etc.


http://downloads.oreilly.com/books/9780596510480.pdf

 

 




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*From:* asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *hadi 
motamedi

*Sent:* 22 December 2009 10:47
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on 
CentOS 5.2?


 

 

On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen 
mailto:tzafrir.co...@xorcom.com>> wrote:


On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
> On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby > wrote:

>
> >  And what is the output of the ./configure?  Does it generate any 
errors?

> >
> >
> >
> > Thanks,
> > --Warren Selby
> >
> > On Dec 22, 2009, at 1:09 AM, hadi motamedi > wrote:

> >
> >
> >
> > On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby 
mailto:wcse...@selbytech.com>>wrote:

> >
> >>  On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi 
mailto:motamed...@gmail.com>>wrote:

> >>
> >>>
> >>> Please find below the error message that I got when issuing "make
> >>> install" :
> >>> [r...@mss-0 asterisk-1.4.26]# make install
> >>> make: -F.: Command not found
> >>> 
> >>>  The configure script must be executed before running 'make'.
> >>>    Please run "./configure".
> >>> 
> >>> make: *** [makeopts] Error 1
> >>>
> >>>
> >>
> >> And did you run ./configure like the error message says?
> >>
> >> --
> >> Thanks,
> >> --Warren Selby
> >> http://www.selbytech.com 
> >>
> >> ___
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> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> > Yes , I did .
> >
> >
> >
> >  ___
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 --

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> >
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 --

> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> Please find below the output of "./configure" :
>
> [r...@mss-0 asterisk-1.4.26]# ./configure
> checking build system type... i686-pc-linux-gnu
> checking host system type... i686-pc-linux-gnu
> checking for gcc... gcc
> checking for C compiler default output file name... a.out
> checking whether the C compiler works... yes
> checking whether we are cross compiling... no
> checking for suffix of executables...
> checking for suffix of object files... o
> checking whether we are using the GNU C compiler... yes
> checking whether gcc accepts -g... yes
> checking for gcc option to accept ISO C89... none needed
> checking how to run the C preprocessor... gcc -E
> checking for grep that handles long lines and -e... /bin/grep
> checking for egrep... /bin/grep -E
> checking for ANSI C header files... yes
> checking for sys/types.h... yes
> checking for sys/stat.h... yes
> checking for stdlib.h... yes
> checking for string.h... yes
> checking for memory.h... yes
> checking for strings.h... yes
> checking for inttypes.h... yes
> checking for stdint.h... yes
> checking for unistd.h... yes
> checking minix/config.h usabil

[asterisk-users] call queue with external numbers??

[Oguzhan Kayhan]

Hello,
Our asterisk is connected to an ericsson pbx by PRI.
What i want is the asterisk clients should call operator numbers by dialing 0

But, when a call is made to ericsson via number 0, it assumes that the
call is made from outside, so it doesnt allow to be dialed.
There are 3 real operator extensions which is grouped by ericsson for
operators. Lets assume  1112 1113.

What i want to know is, is there a way for me to create such group in
asterisk and add that external extension numbers which should be dialed by
order, or by 3 rings at a time etcso that i can create that operator
group on asterisk side also.

PS: I can call real extensions on ericsson without a problem.


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[asterisk-users] Asterisk Release Time Frames

[Russell Bryant]

Greetings,

Asterisk 1.6.2.0 was released last week.  It's time to revisit release 
plans for both current and future Asterisk releases.

For the past few months, there have been discussions regarding some 
updates to Asterisk release policies.  You can find my original -dev 
list post on this topic here [1].

I also spoke about this as part of my presentation at AstriCon, which 
you can find a text version of on the Asterisk project blog [2].

After much positive feedback, we have proceeded with implementing these 
policy updates.  I made some additional comments on this topic and noted 
plans for next major release yesterday [3].

The key things to note are that all current Asterisk releases now have a 
specified end of life date.  Future releases will have EOL dates from 
their initial release.  For additional details regarding maintenance 
time frames for Asterisk releases, please see the project web site [4].

Thank you all very much for your continued support of Asterisk!



[1] http://lists.digium.com/pipermail/asterisk-dev/2009-October/040082.html

[2] 
http://blogs.asterisk.org/2009/11/10/asterisk-project-update-astricon-2009/

[3] http://lists.digium.com/pipermail/asterisk-dev/2009-December/041336.html

[4] http://www.asterisk.org/asterisk-versions



-- 
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
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] Showing "name of extension" when calling

[Doug Lytle]

Kevin P. Fleming wrote:
>
> This is called Connected Party information display, and it will be in
> Asterisk 1.8.
>
>
Wasn't this scheduled for 1.6.2?

Doug



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Re: [asterisk-users] Showing "name of extension" when calling

[Kevin P. Fleming]

Magnus Benngård wrote:

> Is it possible, when placing a call that u see the name of the extension
> in your diplay?
> 
> For example, 2 sip.conf entries:
> [971]
> callerid="Stefan"<971>
> [975]
> callerid="Magnus"<975>
> 
> 975 calls 971 today 975 sees 971 in the display but would like to se:
> "Stefan <975>" or just "Stefan" or...

This is called Connected Party information display, and it will be in
Asterisk 1.8.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org


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[asterisk-users] E1 R2 Congestion Status

[Khaled W Chehab]

I have a 'CONGESTION' Status with R2 protocol.

While testing this scenario sip GW--àAsterisk –Digium E1 R2
ProtocolàCisco E1 R2 protocolàsip Gw

Find below my error and configuration ,where are the errors in my
configuration ?

 

=

Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on
rev-212-98-156-56 (pid = 3614)

Verbosity is at least 3

  == Using SIP RTP CoS mark 5

-- Executing [00223...@default:1] Dial("SIP/98.34.56.216-000e",
"DAHDI/g1/00223344") in new stack

[Dec 22 06:02:49] WARNING[4756]: app_dial.c:1745 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)

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

-- Auto fallthrough, channel 'SIP/98.34.56.216-000e' status is
'CONGESTION'

 

Cisco Gateway 

show controller

Controller E1 slot(0)/port(0) 

   E1 Link is UP

  No Alarm detected.

  Applique type is Channelized E1.

  Framing is CRC4, Line Code is HDB3.

  Signalling type is R2-MFC.

  0 Line Code Violations, 0 Framing Bit Errors

  0 Far End Block Errors, 0 CRC Errors

   signalling type = r2 

   clock source = slave 

   channel group 0 = 1-31 

  1 2 3

   allocated timeslots = YYYNYYY 

   outgoing barred channel group =  

   channel order = ascending 

   b-channel negotiation = exclusive 

   overlap receiving by forced = disabled 

   overlap sending by forced = disabled 

   protocol side = network 

   R2 get calling number = none 

   ISDN virtual connect = disabled 

   ISDN Layer 2 is DOWN

   ISDN Values

  ISDN Layer 2 values

 k= 7

 N200 = 3

 N201 = 260

 T200 = 1 seconds

 T203 = 10 seconds

  ISDN Layer 3 values

 T301 = 180 seconds

 T303 = 4 seconds

 T304 = 20 seconds

 T305 = 30 seconds

 T306 = 30 seconds

 T308 = 4 seconds

 T310 = 10 seconds

 T313 = 10 seconds

 T316 = 120 seconds

 T322 = 4 seconds

 T309 = 90 seconds

 N303 = 1

 

---

/etc/asterisk/chan_dahdi.conf

[trunkgroups]

signalling=mfcr2

mfcr2_variant=mx

trunkgroup => 1,16

spanmap => 1,1,1

 

[channels]

signalling=mfcr2

mfcr2_variant=mx

context=default

signalling=mfcr2

mfcr2_variant=mx

signalling=mfcr2

mfcr2_variant=mx

usecallerid=yes

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

canpark=yes

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=yes

 

group=1

callgroup=1

pickupgroup=1

callerid = asreceived

useincomingcalleridondahditransfer = yes

tonezone = 0 ; 0 is US

channel => 1-15,17-31

signalling=mfcr2

mfcr2_variant=itu

mfcr2_max_ani=7

mfcr2_max_dnis=8

mfcr2_get_ani_first=no

mfcr2_category=national_subscriber

mfcr2_logdir=span1

mfcr2_logging=all

 

;EOF

cat /etc/dahdi/system.conf

# Autogenerated by /usr/sbin/dahdi_genconf on Tue Dec 22 01:59:02 2009

# If you edit this file and execute /usr/sbin/dahdi_genconf again,

# your manual changes will be LOST.

# Dahdi Configuration File

#

# This file is parsed by the Dahdi Configurator, dahdi_cfg

#

# Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 RED

span=1,1,0,cas,hdb3,crc4

# termtype: te

#bchan=1-15,17-31

#dchan=16

cas=1-15:1101

dchan=16

cas=17-31:1101

echocanceller=mg2,1-15,17-31

 

# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS/CRC4 RED

span=2,2,0,ccs,hdb3,crc4

# termtype: te

bchan=32-46,48-62

dchan=47

echocanceller=mg2,32-46,48-62

 

# Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3" HDB3/CCS/CRC4 RED

span=3,3,0,ccs,hdb3,crc4

# termtype: te

bchan=63-77,79-93

dchan=78

echocanceller=mg2,63-77,79-93

 

# Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" HDB3/CCS/CRC4 RED

span=4,4,0,ccs,hdb3,crc4

# termtype: te

bchan=94-108,110-124

dchan=109

echocanceller=mg2,94-108,110-124

 

# Global data

 

loadzone= us

defaultzone = us

[default]

exten => _X.,1,Dial(DAHDI/g1/${EXTEN})

 

 



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[asterisk-users] Showing "name of extension" when calling

[Magnus Benngård]

Hi!

Is it possible, when placing a call that u see the name of the extension
in your diplay?

For example, 2 sip.conf entries:
[971]
callerid="Stefan"
[975]
 callerid="Magnus"

975 calls 971 today 975 sees 971 in the display but would like to se:
"Stefan " or just "Stefan" or...

/Magnus
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[asterisk-users] AsteriskNow and language

[Administrator TOOTAI]

Hi,

I installed AsteriskNow and upgraded FreePBX to 2.6.0. In a sip 
extension definition, when I set language, it is not reported in the 
extensions_custom.conf file (eg language=xx).

Am I missing something or is it not the right way to set language?

BTW, is this a valid place for AsteriskNow questions? Dedicated mailing 
list seems dead.

Thanks for answer

-- 
Daniel

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Re: [asterisk-users] Asterisk 1.2.14 - Play an audio or signal

[Juan David Diaz]

Thanks a lot Alec, I´ll check

2009/12/22 Alec Davis 

>  straight from our 1.6.1 dialplan, don't know about 1.2.14.
>
> exten => s,n,Set(LIMIT_WARNING_FILE=beep)
> exten => s,n,Set(LIMIT_TIMEOUT_FILE=call-terminated)
>
> ;terminate after 1 hour, start beep warnings at 10 minutes, every 5 minutes
> exten =>
> s,n,Dial(${AVAILCHAN_NOSESSION}/${ARG2}#,,rL(360:300:30))
>
>  --
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Juan David Diaz
> *Sent:* Tuesday, 22 December 2009 11:40 a.m.
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Asterisk 1.2.14 - Play an audio or signal
>
> Good Day List Users,
>
> Is there any way to play an audiofile or at least a beep into an
> established call, I want to do this event each 3 minutes in the call, for
> now I have a shell to get the call time and evaluate the 3 minutes.do
> you know any way to play that sound?
>
> I tried app_inject, it works really nice in asterisk 1.4.X releases; but
> my PBX runs 1.2.14 and It can´t be upgraded (policy reasons).
>
> Regards and Thanks every one.
>
>
> --
> Juan.
>
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-- 
Juan.
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Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?

[Dan Journo]

I recommend you follow the detailed install guide in this book and install all 
the required support programs etc.
http://downloads.oreilly.com/books/9780596510480.pdf



Thank you for contacting Kesher Communications Ltd.
IT Maintenance Clients can now receive a faster response by using our Live Chat 
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Company has taken reasonable precautions to ensure no viruses are present in 
this email, the company cannot accept responsibility for any loss or damage 
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From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi
Sent: 22 December 2009 10:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?


On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen 
mailto:tzafrir.co...@xorcom.com>> wrote:
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
> On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby 
> mailto:wcse...@selbytech.com>> wrote:
>
> >  And what is the output of the ./configure?  Does it generate any errors?
> >
> >
> >
> > Thanks,
> > --Warren Selby
> >
> > On Dec 22, 2009, at 1:09 AM, hadi motamedi 
> > mailto:motamed...@gmail.com>> wrote:
> >
> >
> >
> > On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby 
> > mailto:wcse...@selbytech.com>>wrote:
> >
> >>  On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi 
> >> mailto:motamed...@gmail.com>>wrote:
> >>
> >>>
> >>> Please find below the error message that I got when issuing "make
> >>> install" :
> >>> [r...@mss-0 asterisk-1.4.26]# make install
> >>> make: -F.: Command not found
> >>> 
> >>>  The configure script must be executed before running 'make'.
> >>>    Please run "./configure".
> >>> 
> >>> make: *** [makeopts] Error 1
> >>>
> >>>
> >>
> >> And did you run ./configure like the error message says?
> >>
> >> --
> >> Thanks,
> >> --Warren Selby
> >> http://www.selbytech.com
> >>
> >> ___
> >> -- Bandwidth and Colocation Provided by 
> >> http://www.api-digital.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> > Yes , I did .
> >
> >
> >
> >  ___
> > -- Bandwidth and Colocation Provided by 
> > http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by 
> > http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> Please find below the output of "./configure" :
>
> [r...@mss-0 asterisk-1.4.26]# ./configure
> checking build system type... i686-pc-linux-gnu
> checking host system type... i686-pc-linux-gnu
> checking for gcc... gcc
> checking for C compiler default output file name... a.out
> checking whether the C compiler works... yes
> checking whether we are cross compiling... no
> checking for suffix of executables...
> checking for suffix of object files... o
> checking whether we are using the GNU C compiler... yes
> checking whether gcc accepts -g... yes
> checking for gcc option to accept ISO C89... none needed
> checking how to run the C preprocessor... gcc -E
> checking for grep that handles long lines and -e... /bin/grep
> checking for egrep... /bin/grep -E
> checking for ANSI C header files... yes
> checking for sys/types.h... yes
> checking for sys/stat.h... yes
> checking for stdlib.h... yes
> checking for string.h... yes
> checking for memory.h... yes
> checking for strings.h... yes
> checking for inttypes.h... yes
> checking for stdint.h... yes
> checking for unistd.h... yes
> checking minix/config.h usability... no
> checking minix/config.h presence... no
> checking for minix/config.h... no
> checki

Re: [asterisk-users] Making a data connection with Asterisk

[Holger von Ameln]

Have a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS


On Tue, 22 Dec 2009 10:57:36 +, Will Payne 
wrote:
> Hi all,
> 
> We need to start obtaining some billing files from BT via a dial-up ISDN
> connection and I'm wondering if Asterisk is capable of doing this?
> 
> I need to make an ISDN dial-up CHAP connection and, once connected, grab
> some files over FTP. Currently, our Asterisk box is connected to an
ISDN30
> with a Zaptel card.
> 
> This may be a downright stupid question but I'm trying to find out if
> Asterisk can be of any use..
> 
> Any info or pointers in the right direction would be appreciated.
> 
> Thanks,
> Will
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[asterisk-users] Available Agent on Queue

[Daniel Stefanus]

Hi,
I have a problem..I have 3 agents active in my queue..How can I get list 
off all my active agents who was available when a call come?
in my extension.conf :
(1)exten = 6501,n,Queue(${EXTEN},ntT,,,1)
(2)exten = 6501,n,Queue(${EXTEN},ntT,,,25)
after the first one running, if i set like that in my extension.conf, in 
queue_log I can see ringing agents before entering the second queue
1261475067|1261475061.106|6501|IRMA|RINGNOANSWER|1000
1261475067|1261475061.106|6501|UMIE|RINGNOANSWER|1000
1261475067|1261475061.106|6501|EGA|RINGNOANSWER|1000
Is there any simple dialplan or is there any available option to get 
without exit the queue? I'm using Asterisk 1.4.26.2.

Best regards,
Daniel


__ Information from ESET NOD32 Antivirus, version of virus signature 
database 4615 (20091117) __

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com



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[asterisk-users] Making a data connection with Asterisk

[Will Payne]

Hi all,

We need to start obtaining some billing files from BT via a dial-up ISDN 
connection and I'm wondering if Asterisk is capable of doing this?

I need to make an ISDN dial-up CHAP connection and, once connected, grab some 
files over FTP. Currently, our Asterisk box is connected to an ISDN30 with a 
Zaptel card.

This may be a downright stupid question but I'm trying to find out if Asterisk 
can be of any use..

Any info or pointers in the right direction would be appreciated.

Thanks,
Will
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Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?

[hadi motamedi]

On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen wrote:

>  On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
> > On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby 
> wrote:
> >
> > >  And what is the output of the ./configure?  Does it generate any
> errors?
> > >
> > >
> > >
> > > Thanks,
> > > --Warren Selby
> > >
> > > On Dec 22, 2009, at 1:09 AM, hadi motamedi 
> wrote:
> > >
> > >
> > >
> > > On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby  >wrote:
> > >
> > >>  On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi <
> motamed...@gmail.com>wrote:
> > >>
> > >>>
> > >>> Please find below the error message that I got when issuing "make
> > >>> install" :
> > >>> [r...@mss-0 asterisk-1.4.26]# make install
> > >>> make: -F.: Command not found
> > >>> 
> > >>>  The configure script must be executed before running 'make'.
> > >>>    Please run "./configure".
> > >>> 
> > >>> make: *** [makeopts] Error 1
> > >>>
> > >>>
> > >>
> > >> And did you run ./configure like the error message says?
> > >>
> > >> --
> > >> Thanks,
> > >> --Warren Selby
> > >> http://www.selbytech.com
> > >>
> > >> ___
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> > >>
> > >> asterisk-users mailing list
> > >> To UNSUBSCRIBE or update options visit:
> > >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>
> > >
> > >
> > > Yes , I did .
> > >
> > >
> > >
> > >  ___
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> > >
> > > asterisk-users mailing list
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> > >
> > >
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> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> >
> > Please find below the output of "./configure" :
> >
> > [r...@mss-0 asterisk-1.4.26]# ./configure
> > checking build system type... i686-pc-linux-gnu
> > checking host system type... i686-pc-linux-gnu
> > checking for gcc... gcc
> > checking for C compiler default output file name... a.out
> > checking whether the C compiler works... yes
> > checking whether we are cross compiling... no
> > checking for suffix of executables...
> > checking for suffix of object files... o
> > checking whether we are using the GNU C compiler... yes
> > checking whether gcc accepts -g... yes
> > checking for gcc option to accept ISO C89... none needed
> > checking how to run the C preprocessor... gcc -E
> > checking for grep that handles long lines and -e... /bin/grep
> > checking for egrep... /bin/grep -E
> > checking for ANSI C header files... yes
> > checking for sys/types.h... yes
> > checking for sys/stat.h... yes
> > checking for stdlib.h... yes
> > checking for string.h... yes
> > checking for memory.h... yes
> > checking for strings.h... yes
> > checking for inttypes.h... yes
> > checking for stdint.h... yes
> > checking for unistd.h... yes
> > checking minix/config.h usability... no
> > checking minix/config.h presence... no
> > checking for minix/config.h... no
> > checking whether it is safe to define __EXTENSIONS__... yes
> > checking for uname... /bin/uname
> > checking for gcc... (cached) gcc
> > checking whether we are using the GNU C compiler... (cached) yes
> > checking whether gcc accepts -g... (cached) yes
> > checking for gcc option to accept ISO C89... (cached) none needed
> > checking for g++... no
> > checking for c++... no
> > checking for gpp... no
> > checking for aCC... no
> > checking for CC... no
> > checking for cxx... no
> > checking for cc++... no
> > checking for cl.exe... no
> > checking for FCC... no
> > checking for KCC... no
> > checking for RCC... no
> > checking for xlC_r... no
> > checking for xlC... no
> > checking whether we are using the GNU C++ compiler... no
> > checking whether g++ accepts -g... no
> > checking how to run the C preprocessor... gcc -E
> > checking how to run the C++ preprocessor... /lib/cpp
> > configure: error: in `/usr/local/asterisk-1.4.26':
> > configure: error: C++ preprocessor "/lib/cpp" fails sanity check
> > See `config.log' for more details.
>
> Do you have gcc and company installed? gxx, g++?
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
> ___
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



Yes , I have g++ installed .
__

Re: [asterisk-users] Every one Busy Problem

[ABBAS SHAKEEL]

Thanks Tzaffir,

Acutally i reinstalled DAHDI & Asterisk  and every thing seem to work fine
now.

i am using TDM800P  with 8 FXO ports. the Number wasnt busy and asterisk
server can recieve calls through that channel but cant use that channel to
dial out.
As the problem is solved :) so what left to explain

Cheers


On Tue, Dec 22, 2009 at 2:21 PM, Tzafrir Cohen wrote:

> On Tue, Dec 22, 2009 at 11:41:35AM +0500, ABBAS SHAKEEL wrote:
> > Hello
> >
> > When ever i try to use Dial DAHDI / SIP i get the following warning and
> > nothing happens and Asterisk moves to next instruction
> > Even i know that channel is free no one else is using it
> >
> > [Dec 22 12:43:39] WARNING[11915]: app_dial.c:1547 dial_exec_full: Unable
> to
> > create channel of type 'DAHDI' (cause 0 - Unknown)
> >   == Everyone is busy/congested at this time (1:0/0/1)
>
> What was the Dial line you used? What type of DAHDI device? FXS? FXO?
> PRI? Wasn't the number simply busy?
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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>



-- 
Best Regards
Shakeel Abbas
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Re: [asterisk-users] anonymous calls code

[Giorgio Incantalupo]

Hi C F,

I solved the problem!! It was under my nose...
If you are interested the solution is here:
http://www.misdn.org/index.php/FAQ_chan_mISDN
The right section is: "key pad elements"

Giorgio Incantalupo

C F wrote:
> You would have to create a dialplan for it.
> If your provider expects *67 (which is the case here with I/CLEC POTS)
> then you would create something like:
> exten => _*67[2-9]XX,1,Dial(Zap/g1/${EXTEN})
> In the case of PRI you would use:
> exten => _*67[2-9]XX,1,SetCallerPres(prohib)
> exten => _*67[2-9]XX,2,Dial(Zap/g1/${EXTEN:3})
>
>
>
>
> On Mon, Dec 21, 2009 at 6:19 AM, Giorgio Incantalupo
>  wrote:
>   
>> Hi all,
>>
>> does anybody know how to make on-demand anonymous calls? I've tried code
>> *67# before the number to call but it is working with some providers only.
>>
>> Any hints?
>>
>> Thank you.
>>
>>
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> 
>
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>   


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Re: [asterisk-users] asterisk & x-lite

[jonas kellens]

Where is your definition of codecs ??

On Tue, 2009-12-22 at 11:26 +0200, zehra yildiz wrote:

> Hello All,
> 
> I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone.
> The softphone can call the other one but I can' t hear any voice. My
> configuration files are below:
> 
> [r...@localhost asterisk]# cat sip.conf
> [general]
> canreinvite=yes
> 
> [1001]
> username=1001
> password=1001
> type=friend
> context=phones
> host=dynamic
> 
> [1002]
> callerid=1002
> username=1002
> password=1002
> type=friend
> context=phones
> host=dynamic
> 
> [r...@localhost asterisk]# cat extensions.conf
> [globals]
> 
> [general]
> autofallthrough=yes
> 
> [default]
> 
> [incoming_calls]
> 
> [phones]
> exten => _1XXX,1,NoOp()
> exten => _1XXX,n,Dial(SIP/${EXTEN},30)
> exten =>
> _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
> exten => _1XXX,n,Hangup()
> 
> 
> PS: My sip server and softphones are in the same network subnet. There
> are not any firewall or iptables rules. I tried the "nat=yes"
> parameter but no changes.
> 
> ___
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Re: [asterisk-users] New 1.4 system: registered, but not responding to invite?

[David Cunningham]

Doug,

It doesn't respond to the INVITE - the trace says "No response to the
INVITE?". If the phone doesn't even ring it's probably not getting anything,
which points to a problem with the router it's behind. How is the router set
up to deliver SIP and RTP to the phone?

On Tue, Dec 22, 2009 at 5:33 AM, Doug  wrote:

> At 00:46 12/21/2009, Alex Balashov wrote:
>  >A packet capture would be needed to illuminate the source of the problem.
>
> Thanks, Alex for your suggestion.
>
> Here is a link for the packet capture:
>
>   
> http://www.A7H.com/~stuph/TCPdump-2009-Dec-21-2304.txt
>
>
> I just don't see where the extension responds to
> the INVITE.  What would prevent that?
>
> By the way, I have a bunch of phones behind this
> same router that work just fine on our old v1.2
> system.
>
>
>
>
>
>  >
>  >On 12/21/2009 01:39 AM, Doug wrote:
>  >
>  >> I've turned on NAT everywhere I can think, but
>  >> even though I hear ringing on the calling
>  >> phone (different system) the called phone does
>  >> not ring.
>  >>
>  >> Has anyone bumped into this lately?
>
>
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-- 
David Cunningham
Voisonics
IVR development, VOIP consultancy
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3411 5024
Australia: +61 (0) 2 9037 2180
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Re: [asterisk-users] asterisk & x-lite

[BERGANZ François]

It is a nat problem

 

 

 

François BERGANZ

P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

 

De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé : mardi 22 décembre 2009 10:26
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] asterisk & x-lite

 

Hello All,

I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:

[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes

[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic

[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic

[r...@localhost asterisk]# cat extensions.conf
[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/${EXTEN},30)
exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _1XXX,n,Hangup()


PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the "nat=yes" parameter but no
changes.

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Re: [asterisk-users] asterisk & x-lite

[BERGANZ François]

Try tcpdump to see where RTP go from asterisk.

Configure your x-lite

Use stun server ?

 

 

 

P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

 

De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé : mardi 22 décembre 2009 10:26
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] asterisk & x-lite

 

Hello All,

I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:

[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes

[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic

[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic

[r...@localhost asterisk]# cat extensions.conf
[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/${EXTEN},30)
exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _1XXX,n,Hangup()


PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the "nat=yes" parameter but no
changes.

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[asterisk-users] asterisk & x-lite

[zehra yildiz]

Hello All,

I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:

[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes

[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic

[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic

[r...@localhost asterisk]# cat extensions.conf
[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/${EXTEN},30)
exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _1XXX,n,Hangup()


PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the "nat=yes" parameter but no
changes.
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Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?

[Tzafrir Cohen]

On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
> On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby  wrote:
> 
> >  And what is the output of the ./configure?  Does it generate any errors?
> >
> >
> >
> > Thanks,
> > --Warren Selby
> >
> > On Dec 22, 2009, at 1:09 AM, hadi motamedi  wrote:
> >
> >
> >
> > On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby wrote:
> >
> >>  On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi 
> >> wrote:
> >>
> >>>
> >>> Please find below the error message that I got when issuing "make
> >>> install" :
> >>> [r...@mss-0 asterisk-1.4.26]# make install
> >>> make: -F.: Command not found
> >>> 
> >>>  The configure script must be executed before running 'make'.
> >>>    Please run "./configure".
> >>> 
> >>> make: *** [makeopts] Error 1
> >>>
> >>>
> >>
> >> And did you run ./configure like the error message says?
> >>
> >> --
> >> Thanks,
> >> --Warren Selby
> >> http://www.selbytech.com
> >>
> >> ___
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> > Yes , I did .
> >
> >
> >
> >  ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> 
> Please find below the output of "./configure" :
> 
> [r...@mss-0 asterisk-1.4.26]# ./configure
> checking build system type... i686-pc-linux-gnu
> checking host system type... i686-pc-linux-gnu
> checking for gcc... gcc
> checking for C compiler default output file name... a.out
> checking whether the C compiler works... yes
> checking whether we are cross compiling... no
> checking for suffix of executables...
> checking for suffix of object files... o
> checking whether we are using the GNU C compiler... yes
> checking whether gcc accepts -g... yes
> checking for gcc option to accept ISO C89... none needed
> checking how to run the C preprocessor... gcc -E
> checking for grep that handles long lines and -e... /bin/grep
> checking for egrep... /bin/grep -E
> checking for ANSI C header files... yes
> checking for sys/types.h... yes
> checking for sys/stat.h... yes
> checking for stdlib.h... yes
> checking for string.h... yes
> checking for memory.h... yes
> checking for strings.h... yes
> checking for inttypes.h... yes
> checking for stdint.h... yes
> checking for unistd.h... yes
> checking minix/config.h usability... no
> checking minix/config.h presence... no
> checking for minix/config.h... no
> checking whether it is safe to define __EXTENSIONS__... yes
> checking for uname... /bin/uname
> checking for gcc... (cached) gcc
> checking whether we are using the GNU C compiler... (cached) yes
> checking whether gcc accepts -g... (cached) yes
> checking for gcc option to accept ISO C89... (cached) none needed
> checking for g++... no
> checking for c++... no
> checking for gpp... no
> checking for aCC... no
> checking for CC... no
> checking for cxx... no
> checking for cc++... no
> checking for cl.exe... no
> checking for FCC... no
> checking for KCC... no
> checking for RCC... no
> checking for xlC_r... no
> checking for xlC... no
> checking whether we are using the GNU C++ compiler... no
> checking whether g++ accepts -g... no
> checking how to run the C preprocessor... gcc -E
> checking how to run the C++ preprocessor... /lib/cpp
> configure: error: in `/usr/local/asterisk-1.4.26':
> configure: error: C++ preprocessor "/lib/cpp" fails sanity check
> See `config.log' for more details.

Do you have gcc and company installed? gxx, g++?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] anonymous calls code

[Giorgio Incantalupo]

Hi C F,

I tried but does not work. It seems that my telco (telecom) does not 
accept any number with a leading '*'.
Asterisk CLI returns busy:
empty_chan_in_stack: cannot empty channel 255
as if the channel were busy...but it works if I connect a normal phone 
(and it worked with the old analog PBX).
I'm a bit puzzled...

Giorgio

C F wrote:
> You would have to create a dialplan for it.
> If your provider expects *67 (which is the case here with I/CLEC POTS)
> then you would create something like:
> exten => _*67[2-9]XX,1,Dial(Zap/g1/${EXTEN})
> In the case of PRI you would use:
> exten => _*67[2-9]XX,1,SetCallerPres(prohib)
> exten => _*67[2-9]XX,2,Dial(Zap/g1/${EXTEN:3})
>
>
>
>
> On Mon, Dec 21, 2009 at 6:19 AM, Giorgio Incantalupo
>  wrote:
>   
>> Hi all,
>>
>> does anybody know how to make on-demand anonymous calls? I've tried code
>> *67# before the number to call but it is working with some providers only.
>>
>> Any hints?
>>
>> Thank you.
>>
>>
>>
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Re: [asterisk-users] Every one Busy Problem

[Tzafrir Cohen]

On Tue, Dec 22, 2009 at 11:41:35AM +0500, ABBAS SHAKEEL wrote:
> Hello
> 
> When ever i try to use Dial DAHDI / SIP i get the following warning and
> nothing happens and Asterisk moves to next instruction
> Even i know that channel is free no one else is using it
> 
> [Dec 22 12:43:39] WARNING[11915]: app_dial.c:1547 dial_exec_full: Unable to
> create channel of type 'DAHDI' (cause 0 - Unknown)
>   == Everyone is busy/congested at this time (1:0/0/1)

What was the Dial line you used? What type of DAHDI device? FXS? FXO?
PRI? Wasn't the number simply busy?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] Asterisk news :: Next release of Asterisk will be 1.8 Long Term Support

[Olle E. Johansson]

Dear Asterisk community,

Yesterday, Russell Bryant finally made up his mind and confirmed on the 
asterisk-dev mailing list that the next release of Asterisk will be 1.8, which 
will also be a Long Term Support (LTS) release. This also means that the 1.4 is 
now officially classed as a LTS release too.

I feel that this is a very good solution for the whole Asterisk community and 
that we all will benefit from it. I have personally not been happy with the 
1.6.x release schedule, which has been very misunderstood and has confused a 
large group of users. Hopefully, we can now continue with a release schedule 
that the world understands and that makes sense for everyone. While I 
understand the need for releasing quicker than we've done in the past, the 
detail about the naming, the actual release numbers (1.6.0, 1.6.1 etc) was very 
hard to explain to people. With years of experience of doing Asterisk and VoIP 
training, I have a lot of respect of the need of being able to easily explain 
things, from configuration details to release schedule...

Now we, the Asterisk community, need to focus quickly on the new release and 
plan what's going in there. If you have code for new features lying around (as 
I have tons of in various branches of my svn repository), now is the proper 
time to step forward, contribute it to the bug tracker and get it evaluated, 
discussed and maybe finally included in Asterisk. Whatever goes into 1.8 at 
release time, will be what we will have for production use for  a long time.

We also ask you to dedicate time during next year to help the Asterisk project 
with testing. You don't have to be a developer to test - and we need tests of 
everything from documentation to configuration and technichal issues. We don't 
have all of the equipment you have, we don't have your dialplans, we don't have 
all the applications you integrate Asterisk with. If Asterisk is important to 
your organization, please make sure that you dedicate time during the first 
half of 2010 to do regular testing of the new release betas and release 
candidates. We do need your help to make Asterisk 1.8 a good release, worthy to 
replace the 1.4 as a new LTS release. 

If you're a member of a Linux or Asterisk group, please help in organizing 
Asterisk 1.8 test-partys. If you need help with ideas, please contact our 
community liason, John Todd. Meeting other Asterisk users, testing stuff 
together is one of the best ways to expand your knowledge of Asterisk. Sharing 
ideas and how-to's in real time while setting up test labs and scenarious is 
really, really fun.

Here in Sweden, where I live, we have half a meter of snow and very cold 
weather. The days are very short and I've tried to brighten up the darkness by 
decorating my house with a large amount of blinking lamps. No, they're not SIP 
compliant using Subscribe/notify, sorry. That may be a project for a test - to 
see how many phones with subscriptions one Asterisk can carry. If that works 
out well, maybe my house's blinking lamps can be powered by SIP and Asterisk 
1.8 next year :-)

This is the final workday for me before Christmas. Tomorrow, I'll bake the 
traditional Swedish Christmas ham in the owen and then continue with the 
Christmas bread. I just love Christmas - the food, the family and friends, the 
gifts and, well, the food again :-)

I wish you all a Merry Christmas and a Happy New Year!

/Olle



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