Re: [asterisk-users] GSM codec is a good choice ???

[Tzafrir Cohen]

On Tue, Feb 24, 2009 at 11:16:51PM -0200, David fire wrote:
> out there is a free for educational and no commercial G729 lib for asterisk
> you can use it to test in a non-comercial system.

For personal use? Maybe. For educational use: not really. The licensing
of the Intel codec code are not that nice.

And naturally, if you wan ta good speech codec with a high quality and
yet good compression, and no extra bagage of patents, your first choice
should be Speex.

-- 
   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] HDLC Errors

[Daniel Johnson]

Hi,

Occasionally for a day our server will through lots of HDLC errors in
the log and the PRI card with go into alarm.
I have done some searching and this seems to mostly caused from
Hardware incompatibilities / IRQ sharing / Faulty card / Telco problem

The card will generally run for weeks at a time with no issues and then
it has a bad day... and then good for some weeks or so.
I am trying to find which one is the cause of my problems.

169: 1399921290 1667044798 1646568485 1544999181   IO-APIC-level 
wcte12xp0, wctdm24xxp0

Above is the line of my IRQ related to my cards (Full IRQ dump below).
Does the above mean that my two Digium cards share the 169 IRQ?
I take it this is bad?

We have these cards in our asterisk server:
AEX2451E (Analogue lines)
TE121B (PRI ISDN line)

Thanks for your help.

LOG:
[Feb 25 16:05:22] NOTICE[26570] chan_zap.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 1
[Feb 25 16:05:22] NOTICE[26570] chan_zap.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 1
[Feb 25 16:05:22] NOTICE[26570] chan_zap.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 1
[Feb 25 16:05:36] NOTICE[26570] chan_zap.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 1
[Feb 25 16:05:36] NOTICE[26570] chan_zap.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 1
[Feb 25 16:05:37] NOTICE[26570] chan_zap.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 1
[Feb 25 16:05:37] NOTICE[26570] chan_zap.c: PRI got event: HDLC Abort
(6) on Primary D-channel of span 1
[Feb 25 16:05:37] NOTICE[26570] chan_zap.c: PRI got event: HDLC Bad FCS
(8) on Primary D-channel of span 1
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
1: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 1
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
2: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 2
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
3: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 3
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
4: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 4
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
5: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 5
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
6: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 6
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
7: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 7
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
8: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 8
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
9: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 9
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Detected alarm on channel
10: Red Alarm
[Feb 25 16:05:39] WARNING[26571] chan_zap.c: Unable to disable echo
cancellation on channel 10
[Feb 25 16:05:39] NOTICE[26570] chan_zap.c: PRI got event: Alarm (4) on
Primary D-channel of span 1
[Feb 25 16:05:39] WARNING[26570] chan_zap.c: No D-channels available! 
Using Primary channel 16 as D-channel anyway!
[Feb 25 16:05:44] WARNING[26570] chan_zap.c: No D-channels available! 
Using Primary channel 16 as D-channel anyway!
[Feb 25 16:06:28] NOTICE[26571] chan_zap.c: Alarm cleared on channel 1
[Feb 25 16:06:28] NOTICE[26571] chan_zap.c: Alarm cleared on channel 2


asterisk:~# cat /proc/interrupts 
   CPU0   CPU1   CPU2   CPU3   
  0:   1472  743060701    114    112    IO-APIC-edge  timer
  1:    405  2  1  3    IO-APIC-edge  i8042
  8:  0  0  0  0    IO-APIC-edge  rtc
  9:  1  0  0  0   IO-APIC-level  acpi
 12:   1076    691  2  0    IO-APIC-edge  i8042
 14: 83  0  0  1    IO-APIC-edge  libata
 15:  0  0  0  0    IO-APIC-edge  libata
 74:  0  0  0  0   IO-APIC-level 
uhci_hcd:usb1, uhci_hcd:usb3, ehci_hcd:usb5
 90:  0  0  0  0   IO-APIC-level 
uhci_hcd:usb2, uhci_hcd:usb4
106:  115263410   86839619   38832711   22340356 PCI-MSI  eth0
114:   40956818   41973260   57827186   15209426 PCI-MSI  eth1
169: 1399921290 1667044798 1646568485 1544999181   IO-APIC-level 
wcte12xp0, wctdm24xxp0
177:    2024852    1865838 574172

[asterisk-users] Dropping RTP packets

[Jim Dickenson]

I have a SIP phone at home behind a NAT router registered with an * box at
my office with a routable static IP address running version
SVN-branch-1.6.0-r175638M.

If I make a call from my SIP phone out a PRI circuit to my cell phone
everything works as expected. I hear audio in both directions and all is
good.

If from the same SIP phone I make a call via our Veracity SIP account to my
cell phone I hear no audio in either direction.

In trying to find out what is wrong I used tcpdump to see if I could learn
anything. I can see the phone sending fixed length UDP packets on to my home
network heading to the IP address of the * box. If I run tcpdump on the *
box I do not see the packets being received. I do not see the * box sending
any packets to my home network either. I have not checked if the * box is
receiving packets from Veracity I only know that no audio packets are sent
to my home network.

If I use tcpdump to watch the SIP phone call via the PRI circuit I see
packets both on my home network and my * box.

If I use a SIP phone located in my office and make a call via Veracity
everything is okay. Also a co-worker has a vpn router on his home network
connected to the office vpn server and he can make calls from his SIP phone
via Veracity without problems.

I can also call his SIP phone from my SIP phone and packets pass as
expected.

It seems as if audio packets from my SIP phone disappear only if they are
involved with a call via Veracity.

Does anyone have some idea what I might look at to find what is causing this
problem?
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/




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[asterisk-users] Asterisk security between two servers

[arkda]

Hi,

I recently found someone was using one of my Asterisk servers to make
international calls via some SIP method that allowed them to bypass
authentication (running 1.4.21.1 so I'm not sure how they did this since the
major vulnerability for this was patched in 1.4.18.1). At any rate I caught
it the same day they started this, so I've blocked their IP range and put in
some monitoring solutions upstream.

I'd like to lock down the servers as far as Asterisk goes, so this is my
situation:

I'm running two Asterisk servers between two sites. They use DUNDi to route
calls between the two servers so there is no dialplan to route calls between
the two. Firewalls have been configured to allow SIP clients to connect from
the Internet. I'd like to change my sip.conf to allowguest=no without having
to recreate dialplans between the two. How is this accomplished? I can't
seem to find much on using allowguest=no in sip.conf.

Thanks in advance.
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Re: [asterisk-users] Incoming call

[David fire]

if is a codec problem start putting all the systems in the same codec and
disallow all other.
put for example all in alaw and disalaw all other includeing ulaw.

you can make calls from asterisk to the sip extencion registered in opensip?
check that you can start the call from a soft phone or you can use orginate
and send the call to open sip one side and music on hold in the other side.

David

2009/2/24 michel freiha 

> Dera All,
>
> I have the following scenario,
> A customer dial a DID number...The call is routed to a PSTN GW that send
> the call to asterisk...
> On asterisk I created an AGI Script that send the call to an extension
> registered on OpenSIPS server...
> The extension is ringing successfully, but as soon as I accept the call on
> OpenSIPS side the call is hangd up...
> I checked rhe SIP debug and it seems that I have a Codec issue as you can
> see on http://pastebin.com/m767a2172
>
> Need some help please
>
>
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[asterisk-users] Problem redirecting user running a Dynamic feature

[david]

Hello,

Here is my setup :

Telephone 1 ( GXP 2000 )
Telephone 2 ( SPA942 )
Asterisk 1.4.17 ( same behaviour on Asterisk 1.4.23.1 )

Scenario: I don't like the default asterisk transfer feature, so I am
trying to write my own.

What I did :

1. Added to dynamic features #3 with AGI pointing to my php script
2. PHP script asks the user to enter his/her extension
3. PHP connects to Asterisk Manager and sends a redirect command on the
other channel
4. PHP tells user the call has been transfered and wishes the user a
good day.
5. Script ends.

What should happen :
1. Telephone1 calls telephone2
2. Telephone2 enters #3
3. Telephone1 hears on hold music and Telephone2 enters the new extension
4. PHP calls Asterisk Manager to transfer Telephone1 to the new extension
5. Telephone2 hears a message wishing the user a good day
6. Telephone2 hangup

What actually happens :
1. Ok
2. Ok
3. Ok
4. PHP calls Asterisk Manager to transfer Telephone1, but Telephone one
does not transfer it just has dead air
5. Telephone2 hears a message wishing the user a good day
6. Telephone2 hangs up
7. Telephone1 disconnects
8. Telephone1 ( according to asterisk ) completes the transfer the the
new extension
9. Asterisk plays a file, realises that Telephone1 is gone and disconnects.

Here is part of the logs :

telephone2 = 081c5350
telephone1 = 081bfe98
I am transfering the caller to extension 860809864 in default context



  -- Called telepho...@voip.myserver.tld
-- SIP/voip.myserver.tld-081c5350 is ringing
-- SIP/voip.myserver.tld-081c5350 answered
SIP/voip.myserver.tld-081bfe98
--  Feature Found: df3 exten: df3
-- Launched AGI Script /opt/customTransfer.php
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager.d/blank.conf': Found
  == Manager 'theking' logged on from 127.0.0.1
-- PHP Log : Sending to SIP/voip.myserver.tld-081bfe98 to 860809864
  == Manager 'theking' logged off from 127.0.0.1
-- AGI Script /opt/customTransfer.php completed, returning 0
  == Spawn extension (default, 860809864, 0) exited non-zero on
'SIP/voip.myserver.tld-081bfe98'
-- Executing [860809...@default:1]
NoOp("SIP/voip.myserver.tld-081bfe98", "Receiving call to 860809864") in
new stack
-- Executing [860809...@default:2]
Set("SIP/voip.myserver.tld-081bfe98", "__TRANSFER_CONTEXT=default") in
new stack
-- Executing [860809...@default:3]
AGI("SIP/voip.myserver.tld-081bfe98", "/opt/itworks.php") in new stack
-- Launched AGI Script /opt/itworks.php
  == Spawn extension (default, 860809864, 3) exited non-zero on
'SIP/voip.myserver.tld-081bfe98'
cain*CLI>

=

If I run the customTransfer.php in bash while the telephone is NOT in
dynamic features, it works perfectly. It seems like that if telephone1
is held while telephone 2 is in a dynamic feature, the redirect fails.

Additionally, if I do it backwards and the calling user transfers the
called user using #3, it works perfectly without any problems.

What have I done wrong ? Is there a better way to implement a custom
transfer feature?

Thanks,

David




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Re: [asterisk-users] GSM codec is a good choice ???

[drew einhorn]

On Tue, Feb 24, 2009 at 6:39 PM, David fire  wrote:
> ?
> GSM is greate! there was a miss understood.
> if you transcode more than 90 in a core 2 duo you get a bad audio
> GSM to G729 or G711 to G729.
>
> G711 to GSM is ok, but you should test it.
>

The codec is fine, many multi core processors suck,
there are serious bottlenecks between the cores and the memory bus.

Everything is fine as long as there is room for all the data in the level n
cache, but if you have get it from level n+1 cache, or the memory bus,
by the time you get it, its too late.

Don't know how much memory is needed per call transcoded,
but when you multiply an increasing number of calls,
sooner or later you will exceed the limit.

If you want to see an amazing high school project that took
third place in the Siemens Competion,

http://www.cogito.org/Interviews/InterviewsDetail.aspx?ContentID=17633

These kids were wondering why they were having trouble getting the
expected performance from their model, and discovered it was a
limitation of their multi core processor.

-- 
Drew Einhorn

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Re: [asterisk-users] weird problem

[Michael]

On Wed, 25 Feb 2009 10:18:02 you wrote:
> use wireshark or somethink like it (tcpdump) and see if the "bye" is
> reaching asterisk.
> if this is the problem you can use rtptimeout option in the sip.conf or
> iax.conf.
> David
>
> 2009/2/23 Michael 
>
> > I am running Asterisk 1.4.22.2, though I have also found this problem
> > with 1.4.23.x
> >
> > Sometimes after I hang up the system continues to spew packets to my
> > phone causing it to become unusable until I restart Asterisk.
> >
> > Michael

It usually seems to happen after I use voice mail.

I have no 'rtptimeout' set, what should this be?

Michael

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Re: [asterisk-users] Polycom Spectralink 8002 Configuration

[Michael Graves]

It seems to me that based upon your comments you miss the point of the
product. It's design targets large commercial concerns, school
campuses, corporate parks, etc...not making free calls from Starbucks.

I had one under test for several months and it behaved really well on
my WLAN using a Netgear comsumer N type rouiter/AP with WMM. WMM is
essentially a wireless QoS mechanism. Without it you cannot assure
voice quality if there's anything else using the WLAN.

Granted, the phone is a bit fiddly to provision. In it's intended
target markets that's not a problem. If you want to make free calls
from hotspots you're far better of with trashy consumer oriented stuff
that has a built-in web browser. In many cases you need it to
authenticate against the hotspot.

The best option seems to be a SIP client on a dual mode cell phone. But
then, why use the wifi when you have a cell phone in your hand? Minutes
are cheap in either case.

Michael

On Tue, 24 Feb 2009 22:50:53 + (UTC), Jeff LaCoursiere wrote:

>I have one of these seemingly useless devices too.  Please let me know if 
>you get anywhere with it.  I bought it thinking it would be a good phone 
>to take around to various hotspots and keep my extension.  Turns out it 
>really wants to be only in its home "hotspot" and has some stringent 
>restrictions on wifi options (WMM ONLY?!?!) that will probably not be 
>present at Starbucks.  I'm pretty disgusted with it.  Bummer, too, because 
>otherwise Polycom has fantastic VoIP products.
>
>j
>
>On Mon, 23 Feb 2009, M Hulber wrote:
>
>> I have a new Polycom Spectralink 8002 and am having trouble with the
>> configuration or the unit but I can't see what's wrong.  The unit does
>> not seem to even attempt to register with the Asterisk proxy but I can
>> make calls to it.  I have viewed the syslog from the device which it
>> will actually write to the asterisk server so I know it can be reached.
>> I have also run a sip debug and see no registration traffic from the
>> unit.  It also pulls the configs from the tftp server on the asterisk
>> box ok.
>>
>> Does anyone have a sample set of configs that work?  I have samples for
>> the Polycom side but haven't seen the match on the asterisk side.  Since
>> I don't even see traffic, I can't think that it's even an authentication
>> issue.
>>
>> When I dial from the device it just sits there, basically.
>>
>> MARK.
>>
>> --
>>
>> sip_allusers.cfg:  (I've tried most variations on theses settings)
>>
>> ## FOR PROXY1_TYPE = ASTERISK
>>
>> #PROXY1_ADDR = 192.168.2.80:5060# replace the ip address with
>> the Asterisk Server's Address
>> PROXY1_ADDR = 192.168.2.80  # replace the ip address with the
>> Asterisk Server's Address
>> PROXY1_KEYPRESS_2833 = enable
>> PROXY1_KEYPRESS_INFO = enable
>> PROXY1_HOLD_IP0 = disable
>> PROXY1_PRACK = enable
>> #PROXY1_REREG_SECS=3600
>> PROXY1_REREG_SECS=35
>> PROXY1_KEEPALIVE_SECS=14
>> #PROXY1_DOMAIN = asterisk# Replace this with your SIP Domain's name
>> PROXY1_CALLID_PER_LINE = disable
>> PROXY1_MAIL_ACCESS = 864 # Put Your Voice Mail Sytem's
>> Pilot Number here
>>
>> sip_2000.cfg:
>>
>> LINE1 = 2000
>> LINE1_PROXY   = 1
>> LINE1_CALLID  = 2000
>> #LINE1_AUTH= 2000; 2000
>>
>> sip.conf:
>>
>> ; Polycom Spectralink 8002
>> [2000]
>>   type=friend
>>   host=192.168.3.123
>>   ;port=5060
>>   secret=2000
>>   username=2000
>>   ;fromuser=2000
>>   ;authuser=2000
>>   qualify=no   ; turned this off to stop asterisk side initiated traffic
>>   context=spectra_default
>>   dtmfmode=rfc2833
>>   disallow=all
>>   allow=ulaw
>>   mailbox...@default
>>   canreinvite=yes
>>   callgroup=1
>>   pickupgroup=1
>>   accountcode=Home
>>   nat=no
>>
>>
>> Syslog:
>>
>> Feb 23 20:25:06 192.168.3.123 Jan  1 00:18:24.57 0090.7a0a.13f3
>> (192.168.003.123) [0007] Call start, AP 0014.d1c2.70fe (-32 dBm)
>> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.87 0090.7a0a.13f3
>> (192.168.003.123) [0008] Number Abufs: 26
>> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.87 0090.7a0a.13f3
>> (192.168.003.123) [0009] Number Fbufs: 2
>> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.88 0090.7a0a.13f3
>> (192.168.003.123) [000a] Max Number Abufs: 359
>> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.88 0090.7a0a.13f3
>> (192.168.003.123) [000b] Max Number Fbufs: 33
>> Feb 23 20:25:11 192.168.3.123 Jan  1 00:18:29.57 0090.7a0a.13f3
>> (192.168.003.123) [000c] NStat: 0014.d1c2.70fe (-30 dBm), Tx 3704, Rx
>> 43841, BTx 2, BRx 2766, MTx 0, MRx 0, Tx Drop 3 (0.1%), Tx Retry 96
>> (2.7%), Rx Retry 19 (0.0%)
>> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:33.87 0090.7a0a.13f3
>> (192.168.003.123) [000d] Number Abufs: 46
>> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:33.87 0090.7a0a.13f3
>> (192.168.003.123) [000e] Number Fbufs: 3
>> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:34.57 0090.7a0a.13f3
>> (192.168.003.123) [000f] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3707, Rx
>> 43996, BTx 2, BRx 2773, MTx 0, MRx 0, Tx Drop 3 (0.0

Re: [asterisk-users] GSM codec is a good choice ???

[David fire]

?
GSM is greate! there was a miss understood.
if you transcode more than 90 in a core 2 duo you get a bad audio
GSM to G729 or G711 to G729.

G711 to GSM is ok, but you should test it.


2009/2/24 Alejandro Cabrera Obed 

> Do you think GSM codec has poor audio quality ???
>
> Because I've made some tests among softphones connected from different
> cities of my country and the audio was good to me.
>
> Maybe GSM is a good choice.
>
> On Tue, Feb 24, 2009 at 11:16 PM, David fire  wrote:
> > out there is a free for educational and no commercial G729 lib for
> asterisk
> > you can use it to test in a non-comercial system.
> > the digium lib is much better. if you have more than 30~60 phones
> > transcoding inst a very good idea.
> > i made my self a test on a core 2 duo 64 bits 2GB of ram a test
> transcoding
> > more than 90 calls the sound quality was BAD not poor BAD.
> >
> > the digium transcoder is GREATE 0 cpu was gone for transcoding.
> >
> > keep this in mind.
> >
> > David
> >
> > 2009/2/24 Kristian Kielhofner 
> >>
> >> On Mon, Feb 23, 2009 at 6:59 PM, Alejandro Cabrera Obed
> >>  wrote:
> >> > Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
> >> > with GSM sound files.
> >> >
> >> > The problem is I have IP phones Utopix HyperPhone 202 which support
> >> > only G.729a/u and G.723.1 high/low, but not GSM.
> >> >
> >> > If I choose G.729A the "pass-throu" calls among users are OK, but
> >> > Asterisk can't transcode GSM to G.729A in voicemail calls.
> >> >
> >> > My company doesn'y want to pay for a G.729 license, so I'm thinking to
> >> > buy new IP phones with GSM support, so I have no problem with the
> >> > voicemail system.
> >> >
> >> > Are the IP phone with GSM support a good choice for me ???
> >> >
> >> > (Maybe in the future I need to connect the Asterisk with the PSTN, GSM
> >> > doesn't matter at this point ???)
> >> >
> >> > Really thanks,
> >> >
> >> > Alejandro
> >> >
> >>
> >> Install the G.729 sound files and make app_voicemail record messages
> >> (format=g729) in G729.  As long as you don't need meetme or a few
> >> other apps that essentially require G.729 transcoding you don't need a
> >> license.
> >>
> >> --
> >> Kristian Kielhofner
> >> http://blog.krisk.org
> >> http://www.submityoursip.com
> >> http://www.astlinux.org
> >> http://www.star2star.com
> >>
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> >
> >
> >
> > --
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> > (")_(")signature to help him gain world domination.
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>
>
> --
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> aco1...@gmail.com
> www.alejandrocabrera.com.ar
>
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Re: [asterisk-users] multiple asterisks in a server

[Rilawich Ango]

Can you elaborate more how to use 2 IPs for 2 instances of asterisk?

On Tue, Feb 24, 2009 at 5:44 PM, Geraint Lee  wrote:
> Almost forgot, you need to make sure you bind each instance to either it's
> own IP address or different ports on the same ip, i used 2 IP's for it and
> never hda a problem.
>
> 2009/2/24 Geraint Lee 
>>
>> Yes it's possible..
>>
>> When you install use...
>> ./configure --prefix=/usr/local/asterisk2 or something like it.
>>
>> I had to change astrundir (in asterisk.conf) as well.
>>
>> One thing to watch out for is that if you run make samples it will
>> overwrite the ones stored in /etc/asterisk and not where you'd expect them
>> to be in /usr/local/asterisk2/etc/asterisk (or at least it di dwhen i did
>> it!).
>>
>> and for a helping hand i symlinked /usr/local/asterisk2/sbin/asterisk to
>> /usr/local/sbin/asterisk2 and /usr/local/asterisk2/sbin/safe_asterisk to
>> /usr/local/sbin/safe_asterisk2
>>
>> Cheers
>>
>> Geraint
>>
>> You will also need to look at asterisk.conf in the new installation
>> directory and as a quickfix to get it running, use a different location for
>> astrundir
>>
>> 2009/2/24 Rilawich Ango 
>> - Show quoted text -
>>>
>>> Hi all,
>>>  Is it possible to install more than 1 asterisk in a single server?
>>> If yes, what do I need to set and take care?
>>>
>>> Rgds,
>>> ango
>>>
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>
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Re: [asterisk-users] (no subject)

[C F]

Right

On Mon, Feb 23, 2009 at 9:07 PM, Lê Văn Hòa  wrote:
>
>
> ko gui nua
> --
>
>
>
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Re: [asterisk-users] multiple asterisks in a server

[Rilawich Ango]

It seems better to install once with multiple instances.  Do we need
to take care the port or IP of each instance?

On Wed, Feb 25, 2009 at 5:36 AM, Klaus Darilion
 wrote:
> Klaus Darilion wrote:
>> Rilawich Ango wrote:
>>> Hi all,
>>>   Is it possible to install more than 1 asterisk in a single server?
>>> If yes, what do I need to set and take care?
>>
>> Just to have several Asterisk instances on a single server you do not
>> need to install it multiple times. Install it once and start it multiple
>> times.
>>
>> Of course you have to have a dedicated configuration for each server, eg:
>> /etc/asterisk/instance1/*
>> /etc/asterisk/instance2/*
>> /etc/asterisk/instance3/*
>>
>> Then you start the Asterisk process and specify the location of the
>> asterisk.conf file.
>>
>> asterisk -C /etc/asterisk/instance1/asterisk.conf
>> asterisk -C /etc/asterisk/instance2/asterisk.conf
>> asterisk -C /etc/asterisk/instance3/asterisk.conf
>>
>> Further, in asterisk.conf specify for each asterisk instance a different
>> location of: spool directory, PID file, 
>
> btw: I use a common /var/lib/asterisk/ as I want to have the same
> "sounds" for all instances. This gives a problem when you use 1.4, as
> 1.4 can not configure the location of astdb. For these you have to apply
> this patch:
> http://bugs.digium.com/view.php?id=14257
>
> regards
> klaus
>
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Re: [asterisk-users] GSM codec is a good choice ???

[Alejandro Cabrera Obed]

Do you think GSM codec has poor audio quality ???

Because I've made some tests among softphones connected from different
cities of my country and the audio was good to me.

Maybe GSM is a good choice.

On Tue, Feb 24, 2009 at 11:16 PM, David fire  wrote:
> out there is a free for educational and no commercial G729 lib for asterisk
> you can use it to test in a non-comercial system.
> the digium lib is much better. if you have more than 30~60 phones
> transcoding inst a very good idea.
> i made my self a test on a core 2 duo 64 bits 2GB of ram a test transcoding
> more than 90 calls the sound quality was BAD not poor BAD.
>
> the digium transcoder is GREATE 0 cpu was gone for transcoding.
>
> keep this in mind.
>
> David
>
> 2009/2/24 Kristian Kielhofner 
>>
>> On Mon, Feb 23, 2009 at 6:59 PM, Alejandro Cabrera Obed
>>  wrote:
>> > Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
>> > with GSM sound files.
>> >
>> > The problem is I have IP phones Utopix HyperPhone 202 which support
>> > only G.729a/u and G.723.1 high/low, but not GSM.
>> >
>> > If I choose G.729A the "pass-throu" calls among users are OK, but
>> > Asterisk can't transcode GSM to G.729A in voicemail calls.
>> >
>> > My company doesn'y want to pay for a G.729 license, so I'm thinking to
>> > buy new IP phones with GSM support, so I have no problem with the
>> > voicemail system.
>> >
>> > Are the IP phone with GSM support a good choice for me ???
>> >
>> > (Maybe in the future I need to connect the Asterisk with the PSTN, GSM
>> > doesn't matter at this point ???)
>> >
>> > Really thanks,
>> >
>> > Alejandro
>> >
>>
>> Install the G.729 sound files and make app_voicemail record messages
>> (format=g729) in G729.  As long as you don't need meetme or a few
>> other apps that essentially require G.729 transcoding you don't need a
>> license.
>>
>> --
>> Kristian Kielhofner
>> http://blog.krisk.org
>> http://www.submityoursip.com
>> http://www.astlinux.org
>> http://www.star2star.com
>>
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>
>
>
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aco1...@gmail.com
www.alejandrocabrera.com.ar

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Re: [asterisk-users] GSM codec is a good choice ???

[David fire]

out there is a free for educational and no commercial G729 lib for asterisk
you can use it to test in a non-comercial system.
the digium lib is much better. if you have more than 30~60 phones
transcoding inst a very good idea.
i made my self a test on a core 2 duo 64 bits 2GB of ram a test transcoding
more than 90 calls the sound quality was BAD not poor BAD.

the digium transcoder is GREATE 0 cpu was gone for transcoding.

keep this in mind.

David

2009/2/24 Kristian Kielhofner 

> On Mon, Feb 23, 2009 at 6:59 PM, Alejandro Cabrera Obed
>  wrote:
> > Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
> > with GSM sound files.
> >
> > The problem is I have IP phones Utopix HyperPhone 202 which support
> > only G.729a/u and G.723.1 high/low, but not GSM.
> >
> > If I choose G.729A the "pass-throu" calls among users are OK, but
> > Asterisk can't transcode GSM to G.729A in voicemail calls.
> >
> > My company doesn'y want to pay for a G.729 license, so I'm thinking to
> > buy new IP phones with GSM support, so I have no problem with the
> > voicemail system.
> >
> > Are the IP phone with GSM support a good choice for me ???
> >
> > (Maybe in the future I need to connect the Asterisk with the PSTN, GSM
> > doesn't matter at this point ???)
> >
> > Really thanks,
> >
> > Alejandro
> >
>
> Install the G.729 sound files and make app_voicemail record messages
> (format=g729) in G729.  As long as you don't need meetme or a few
> other apps that essentially require G.729 transcoding you don't need a
> license.
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
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Re: [asterisk-users] GSM codec is a good choice ???

[Kristian Kielhofner]

On Mon, Feb 23, 2009 at 6:59 PM, Alejandro Cabrera Obed
 wrote:
> Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
> with GSM sound files.
>
> The problem is I have IP phones Utopix HyperPhone 202 which support
> only G.729a/u and G.723.1 high/low, but not GSM.
>
> If I choose G.729A the "pass-throu" calls among users are OK, but
> Asterisk can't transcode GSM to G.729A in voicemail calls.
>
> My company doesn'y want to pay for a G.729 license, so I'm thinking to
> buy new IP phones with GSM support, so I have no problem with the
> voicemail system.
>
> Are the IP phone with GSM support a good choice for me ???
>
> (Maybe in the future I need to connect the Asterisk with the PSTN, GSM
> doesn't matter at this point ???)
>
> Really thanks,
>
> Alejandro
>

Install the G.729 sound files and make app_voicemail record messages
(format=g729) in G729.  As long as you don't need meetme or a few
other apps that essentially require G.729 transcoding you don't need a
license.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] Aastra phones

[Greg Kennedy]

> Message: 13
> Date: Tue, 24 Feb 2009 15:13:41 -0500
> From: Mike 
> Subject: Re: [asterisk-users] Aastra phones
> To: asterisk-ad...@hulber.com, 'Asterisk Users Mailing List -
>   Non-Commercial  Discussion' 
> Message-ID: <001501c996bc$63001e60$29005b...@ca>
> Content-Type: text/plain; charset=iso-8859-1
> 
> I was hoping for a unattended firmware upgrade.  Let's say I have 50 phones
> out in the field; just changing the firmware file (i.e.: replacing 9143i.st)
> would upgrade all phones, without any intervention on my part.
> 
> That's what I do with my 200+ Polycom phones.
> 
> Possible or not?  
> 
> Mike
Mike,

I just login to each phone when i install them originally and set the tftp 
settings to be the right filename and server. I also store the config files on 
the server as well, much easier to make changes to all the phones or do 
upgrades, I just update the firmware and then blast out a sip notify out to the 
phones in wanna upgrade. Its a little more labor intensive on setup, but once 
done, it rocks. I do the same for my polycoms. I have a mix of both, across 12 
sites, and i remotely update them all all the time. Its just a pain to setup 
one time, then its cake!!
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Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

[Tilghman Lesher]

On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote:
> Barry L. Kline wrote:
> > that is supposed to gosub into the incoming extension at priority 1.
> > Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the
> > requested extension wasn't present in the incoming context.
>
> Really strange that Goto and Gosub behave different.

If Goto behaves that way, that's a bug.  As stated in a prior email, the
"i" extension should only be implicitly invoked when waiting for a new
extension and the typed extension does not match anything.

-- 
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Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

[Tilghman Lesher]

On Tuesday 24 February 2009 13:44:25 Barry L. Kline wrote:
> Here's one that may be of interest to any upgraders.  If you rely on the
> behavior of gosub you may want to make note of this change.
>
> I have an incoming call context:
>
> exten => _,n,GoSub(incoming,${EXTEN},1(${EXTEN}));
>
> that is supposed to gosub into the incoming extension at priority 1.
> Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the
> requested extension wasn't present in the incoming context.
>
> When I upgraded to 1.6.0.6 this behavior changed and I would simply get
> an error on the console that a matching extension was not found, and the
> dialplan would simply stop.  It was easy enough to add:
>
> [incoming]
> exten => _,1,Goto(i,1)
>
> to restore the previous behavior (I'm looking at four-digits from a PRI)
> which I should probably have done anyway.
>
> I don't know if this is a bug or WAD but just wanted to mention it.

It was a bug.  Gosub/Goto should NEVER go to the "i" extension, unless that
target is explicitly given.  The use of the "i" extension for invalid
extensions is limited to WaitExten/Background.

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Re: [asterisk-users] Polycom Spectralink 8002 Configuration

[Jeff LaCoursiere]

I have one of these seemingly useless devices too.  Please let me know if 
you get anywhere with it.  I bought it thinking it would be a good phone 
to take around to various hotspots and keep my extension.  Turns out it 
really wants to be only in its home "hotspot" and has some stringent 
restrictions on wifi options (WMM ONLY?!?!) that will probably not be 
present at Starbucks.  I'm pretty disgusted with it.  Bummer, too, because 
otherwise Polycom has fantastic VoIP products.

j

On Mon, 23 Feb 2009, M Hulber wrote:

> I have a new Polycom Spectralink 8002 and am having trouble with the
> configuration or the unit but I can't see what's wrong.  The unit does
> not seem to even attempt to register with the Asterisk proxy but I can
> make calls to it.  I have viewed the syslog from the device which it
> will actually write to the asterisk server so I know it can be reached.
> I have also run a sip debug and see no registration traffic from the
> unit.  It also pulls the configs from the tftp server on the asterisk
> box ok.
>
> Does anyone have a sample set of configs that work?  I have samples for
> the Polycom side but haven't seen the match on the asterisk side.  Since
> I don't even see traffic, I can't think that it's even an authentication
> issue.
>
> When I dial from the device it just sits there, basically.
>
> MARK.
>
> --
>
> sip_allusers.cfg:  (I've tried most variations on theses settings)
>
> ## FOR PROXY1_TYPE = ASTERISK
>
> #PROXY1_ADDR = 192.168.2.80:5060# replace the ip address with
> the Asterisk Server's Address
> PROXY1_ADDR = 192.168.2.80  # replace the ip address with the
> Asterisk Server's Address
> PROXY1_KEYPRESS_2833 = enable
> PROXY1_KEYPRESS_INFO = enable
> PROXY1_HOLD_IP0 = disable
> PROXY1_PRACK = enable
> #PROXY1_REREG_SECS=3600
> PROXY1_REREG_SECS=35
> PROXY1_KEEPALIVE_SECS=14
> #PROXY1_DOMAIN = asterisk# Replace this with your SIP Domain's name
> PROXY1_CALLID_PER_LINE = disable
> PROXY1_MAIL_ACCESS = 864 # Put Your Voice Mail Sytem's
> Pilot Number here
>
> sip_2000.cfg:
>
> LINE1 = 2000
> LINE1_PROXY   = 1
> LINE1_CALLID  = 2000
> #LINE1_AUTH= 2000; 2000
>
> sip.conf:
>
> ; Polycom Spectralink 8002
> [2000]
>   type=friend
>   host=192.168.3.123
>   ;port=5060
>   secret=2000
>   username=2000
>   ;fromuser=2000
>   ;authuser=2000
>   qualify=no   ; turned this off to stop asterisk side initiated traffic
>   context=spectra_default
>   dtmfmode=rfc2833
>   disallow=all
>   allow=ulaw
>   mailbox...@default
>   canreinvite=yes
>   callgroup=1
>   pickupgroup=1
>   accountcode=Home
>   nat=no
>
>
> Syslog:
>
> Feb 23 20:25:06 192.168.3.123 Jan  1 00:18:24.57 0090.7a0a.13f3
> (192.168.003.123) [0007] Call start, AP 0014.d1c2.70fe (-32 dBm)
> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.87 0090.7a0a.13f3
> (192.168.003.123) [0008] Number Abufs: 26
> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.87 0090.7a0a.13f3
> (192.168.003.123) [0009] Number Fbufs: 2
> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.88 0090.7a0a.13f3
> (192.168.003.123) [000a] Max Number Abufs: 359
> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.88 0090.7a0a.13f3
> (192.168.003.123) [000b] Max Number Fbufs: 33
> Feb 23 20:25:11 192.168.3.123 Jan  1 00:18:29.57 0090.7a0a.13f3
> (192.168.003.123) [000c] NStat: 0014.d1c2.70fe (-30 dBm), Tx 3704, Rx
> 43841, BTx 2, BRx 2766, MTx 0, MRx 0, Tx Drop 3 (0.1%), Tx Retry 96
> (2.7%), Rx Retry 19 (0.0%)
> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:33.87 0090.7a0a.13f3
> (192.168.003.123) [000d] Number Abufs: 46
> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:33.87 0090.7a0a.13f3
> (192.168.003.123) [000e] Number Fbufs: 3
> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:34.57 0090.7a0a.13f3
> (192.168.003.123) [000f] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3707, Rx
> 43996, BTx 2, BRx 2773, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96
> (0.0%), Rx Retry 19 (0.0%)
> Feb 23 20:25:21 192.168.3.123 Jan  1 00:18:39.57 0090.7a0a.13f3
> (192.168.003.123) [0010] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3708, Rx
> 44284, BTx 2, BRx 2792, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96
> (0.0%), Rx Retry 19 (0.0%)
> Feb 23 20:25:26 192.168.3.123 Jan  1 00:18:44.36 0090.7a0a.13f3
> (192.168.003.123) [0011] Call end, AP 0014.d1c2.70fe (-36 dBm)
>
>
>
>
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Re: [asterisk-users] asterisk -f and restart now

[Ex Vito]

>
> # rasterisk
> Connected to Asterisk 1.4.23 currently running on debian (pid = 17191)
> samuel*CLI> restart now
> samuel*CLI>
> Disconnected from Asterisk server
> Executing last minute cleanups
> Asterisk ending (0).
> # rasterisk
> Connected to Asterisk SVN-branch-1.4-r178373 currently running on debian
> (pid = 17191)
>
>
> This is interesting: Asterisk is running a new version, but still the
> same PID...
>

  man 3 exec

--
  exvito

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Re: [asterisk-users] multiple asterisks in a server

[Senad Jordanovic]

Danny Nicholas wrote:
> It is possible but not easy.  Virtualization isn't necessarily the answer
> because of sharing the physical device(s) - If you're a SIP-only
> environment, then that wouldn't be a problem, but most * installs (IMO) use
> some flavor of Zap/DAHDI which has to be addressed/locked.
> 



Maybe you can use SERVERware...

http://www.bicomsystems.com/products/


Senad



> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Tuesday, February 24, 2009 10:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] multiple asterisks in a server
> 
> On Tue, Feb 24, 2009 at 2:59 AM, Rilawich Ango 
> wrote:
>> Hi all,
>>  Is it possible to install more than 1 asterisk in a single server?
> 
> Can somebody help me understand why you would want to do this?
> 
> I suppose development versus production, but wouldn't you also want
> better separation, like virtualization?
> 
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[asterisk-users] Incoming call

[michel freiha]

Dera All,

I have the following scenario,
A customer dial a DID number...The call is routed to a PSTN GW that send the
call to asterisk...
On asterisk I created an AGI Script that send the call to an extension
registered on OpenSIPS server...
The extension is ringing successfully, but as soon as I accept the call on
OpenSIPS side the call is hangd up...
I checked rhe SIP debug and it seems that I have a Codec issue as you can
see on http://pastebin.com/m767a2172

Need some help please
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Re: [asterisk-users] asterisk -f and restart now

[Klaus Darilion]

Klaus Darilion wrote:
> Tzafrir Cohen wrote:
>> On Tue, Feb 24, 2009 at 10:55:40AM +0100, Klaus Darilion wrote:
>>> Hi!
>>>
>>> If I start Asterisk with -f (do not force) then on "restart now" the
>>> Asterisk process still has the same PID. 
>> On 'restart' Asterisk basically re-execs itself. Both with and without -f .
> 
> So, if I update Asterisk (make install) is it sufficient to call 
> "restart now" or do I have to stop and start Asterisk?

# rasterisk
Connected to Asterisk 1.4.23 currently running on debian (pid = 17191)
samuel*CLI> restart now
samuel*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk ending (0).
# rasterisk
Connected to Asterisk SVN-branch-1.4-r178373 currently running on debian 
(pid = 17191)


This is interesting: Asterisk is running a new version, but still the 
same PID...

regards
klaus

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Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

[Klaus Darilion]

Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Here's one that may be of interest to any upgraders.  If you rely on the
> behavior of gosub you may want to make note of this change.
> 
> I have an incoming call context:
> 
> exten => _,n,GoSub(incoming,${EXTEN},1(${EXTEN}));
> 
> that is supposed to gosub into the incoming extension at priority 1.
> Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the
> requested extension wasn't present in the incoming context.
> 
> When I upgraded to 1.6.0.6 this behavior changed and I would simply get
> an error on the console that a matching extension was not found, and the
> dialplan would simply stop.  It was easy enough to add:
> 
> [incoming]
> exten => _,1,Goto(i,1)
> 
> to restore the previous behavior (I'm looking at four-digits from a PRI)
> which I should probably have done anyway.
> 
> I don't know if this is a bug or WAD but just wanted to mention it.

That's probably related to a bug I reported. The i extension did match 
when the Gosub was executed normally, but now when the Gosub was 
executed from within a macro. I reported that they should fix the macro 
case - obviously they implemented the "bug" also in the normal case.

Nevertheless you can workaround it by using e.g. _[0-9]. instead of i

Really strange that Goto and Gosub behave different.

regards
klaus

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Re: [asterisk-users] receiving 1st digit from a variable

[Tamer Higazi]

Allmost :)

It is exactly:

exten => _[0-1]X.,1,Set(MSNCHOICE=${EXTEN:0:1})


Thank you for your great support.


Tamer




Danny Nicholas schrieb:
> Syntax should be
> exten => _[0-1]X.,1,Set(MSNCHOICE=${EXTEN:1:1})
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
> Sent: Tuesday, February 24, 2009 3:23 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] receiving 1st digit from a variable
>
> exten => _[0-1]X.,1,Set(MSNCHOICE=${EXTEN,1:1})
>
> brings me this output:
>
> Executing [1017649374...@officeie:1] Set("SIP/2000-007acf80",
> "MSNCHOICE=") in new stack
>
>
> and the result is always empty!
>
> even if I make ${EXTEN,1:3} or whaever
>
>
> Danny Nicholas schrieb:
>   
>> Exten => x,n,Set($NEWVAR=${EXTEN,1:1})
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
>> Sent: Tuesday, February 24, 2009 2:52 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] receiving 1st digit from a variable
>>
>> Hi people!
>> I want to save the 1st letter from the ${EXTEN} variable. I don't want
>> to trim it, I want to RESAVE it into a new variable.
>>
>> Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0
>>
>> I would thank you for all advises.
>>
>>
>>
>> Tamer
>>
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>> 
>
>
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Re: [asterisk-users] receiving 1st digit from a variable

[Klaus Darilion]

Tamer Higazi wrote:
> Hi people!
> I want to save the 1st letter from the ${EXTEN} variable. I don't want
> to trim it, I want to RESAVE it into a new variable.
> 
> Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0

Set(FIRSTDIGIT=${EXTEN:0:1})

substring options:
first number:  where to start (starts with zero)
second number: how many characters

regards
klaus

> I would thank you for all advises.
> 
> 
> 
> Tamer
> 
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[asterisk-users] Multiple SIPGate accounts.

[Razza]

Hi all,
I have two sipgate accounts (numbers), if I have both accounts register only
one will work for incoming calls (which is all i'm interested in). However
if I disable either account the other account will work perfectly. Am I
missing something obvious?

Thanks in advance,
Ray.

Excerpts from sip.conf -

[general]
8< SNIP! >8
Register => 1212121:a...@sipgate.co.uk/1212121
Register => 1313131:b...@sipgate.co.uk/1313131
8< SNIP! >8

[sipgate]
type=friend
username=1212121
secret=
host=sipgate.co.uk
fromuser=1212121
fromdomain=sipgate.co.uk
nat=yes
authuser=1212121
dtmfmode=rfc2833
context=infoline_SG
insecure=very
canreinvite=no
disallow=all
allow=alaw

[2sipgate2]
type=friend
username=1313131
secret=
host=sipgate.co.uk
fromuser=1313131
fromdomain=sipgate.co.uk
nat=yes
authuser=1313131
dtmfmode=rfc2833
context=infoline_config_SG
insecure=very
canreinvite=no
disallow=all
allow=alaw

Not that it really matters as these work when the other account is disabled,
Excerpts from extensions.conf -

8< SNIP! >8
[infoline_SG]
exten => 1212121,1,Goto(infoline,s,1)

[infoline]
exten => s,1,Answer
exten => s,2,DigitTimeout,5
exten => s,3,ResponseTimeout,10
exten => s,4,BackGround(/var/lib/asterisk/infolinesounds/welcomeHL)
8< SNIP! >8

[infoline_config_SG]
exten => 1313131,1,Answer
exten => 1313131,2,Background(/var/lib/asterisk/infolinesounds/welcomeHLC)
exten => 1313131,3,Authenticate(1234)
exten => 1313131,4,Goto(infoline_config,s,1)

[infoline_config]
exten => s,1,Answer
exten => s,2,DigitTimeout,5
exten => s,3,ResponseTimeout,10
exten => s,4,BackGround(/var/lib/asterisk/infolinesounds/welcomeHLC)
8< SNIP! >8
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Re: [asterisk-users] asterisk -f and restart now

[Klaus Darilion]

Tzafrir Cohen wrote:
> On Tue, Feb 24, 2009 at 10:55:40AM +0100, Klaus Darilion wrote:
>> Hi!
>>
>> If I start Asterisk with -f (do not force) then on "restart now" the
>> Asterisk process still has the same PID. 
> 
> On 'restart' Asterisk basically re-execs itself. Both with and without -f .

So, if I update Asterisk (make install) is it sufficient to call 
"restart now" or do I have to stop and start Asterisk?

thanks
klaus

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Re: [asterisk-users] Polycom Phones start to break up after being up a LONG time

[Jeff LaCoursiere]

I can second that.  I am using latest (4.1.2) bootrom and (3.1.2) sip with 
several hundred Polycom 501 and 601 phones.  Tastes great.  Less filling.

j

On Tue, 24 Feb 2009, Mike wrote:

> There was an issue with the early 2.x SIP with Asterisk IIRC, but that's
> been fixed for awhile.
>
>
>
> I'm using Asterisk 1.4.23.1, with plenty of 3.1.1 and 3.1.2 phones. YMMV,
> but it's possible your documentation is also out of date :-)
>
>
>
> Mike
>
>
>
>
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
> Sent: Tuesday, February 24, 2009 15:58
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Polycom Phones start to break up after being
> up a LONG time
>
>
>
> This is probably out of date, but my Polycom documentation recommends
> keeping the boot rom at less than 2.0 for use with Asterisk.
>
>
>
>  _
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
> Sent: Tuesday, February 24, 2009 2:54 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Polycom Phones start to break up after being
> up a LONG time
>
>
>
> That`s an old version, I've had plenty of issues (nothing like what you
> describe) since 2.2.0.  Try going to the latest and greatest (SIP 3.1.2), it
> seems stable (as far as I can tell with about 200 phones, including many
> 501).
>
>
>
> It's hard to troubleshoot something that hasn't been current for about a
> year.
>
>
>
> Mike
>
>
>
>
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry D.
> Hassler
> Sent: Tuesday, February 24, 2009 15:38
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Polycom Phones start to break up after being
> up a LONG time
>
>
>
> Folks, I haven't paid attention to these responses, sorry!
>
> This appears to be an issue primarily on calls with an EXTERNAL leg, I'm
> fairly certain it's on both inbound and outbound calls. The frequency of the
> reports from this client are increasing, and although I was intending to do
> a mass reboot of all the polycoms over the weekend, I have not yet, but will
> have to do it tonight (2 more reports today).
>
> All the external legs are via PRI.
>
> The phones are 501 and 601's, running bootrom 3.2.2.0019 and SIP 2.2.0.0047.
>
> There are many of these that ARE on POE switched (Cisco),
>
> On Fri, Feb 20, 2009 at 12:34 PM, Jeff LaCoursiere  wrote:
>
>
> You also don't mention if it is internal to internal or if there is an
> external leg involved, and if so what type.
>
> j
>
>
> On Fri, 20 Feb 2009, Asterisk Asterisk wrote:
>
>> That's interesting - I haven't noticed this with any of my installs. What
> version of firmware and SIP?
>>
>>
>>
>>
>> 
>> From: Barry D. Hassler 
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
>> Sent: Friday, February 20, 2009 8:41:33 AM
>> Subject: [asterisk-users] Polycom Phones start to break up after being up
> a LONG time
>>
>> Has anyone else encountered this? I have a fairly large installation (~50
> phones, almost all Polycom 501's and a handful of 601's. We're running into
> a number of phones on which the outbound voice (Polycom phone user doesn't
> hear any problems, but the other end does) is breaking up occasionally --
> enough to be noticeable and make you say "what?". In each case, rebooting
> the phone has resolved the symptoms, but I'd like to know if there is a
> known problem.
>>
>> most of these phones would be up for several months now (installed this
> past summer), and unless there are any power outages, would not be restarted
> specifically.
>>
>> I'm planning on restarting all the phones over the weekend, but as this is
> a 24-hour operation, we'd like to avoid interrupting phones at all.
>>
>> --
>> Barry D. Hassler
>> President, HCST
>>
>> http://www.hcst.net/
>> 937-427-9000
>>
>>
>>
>>
>
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>
>
>
> -- 
> Barry D. Hassler
> President, HCST
>
> http://www.hcst.net/
> 937-427-9000
>
>

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Re: [asterisk-users] what is the correct character to separate application parameters: , or |

[Klaus Darilion]

Eric Wieling, Asteria Solutions Group wrote:
> Versions before 0.65 I don't know about
>  From 0.65 to 1.4.x you can use either , or |
> In 1.6 you must use , (| was removed)

So I assume , is the way to go (to have a configuration which is 
compatible with 1.4 and 1.6).

thanks
klaus

> 
> 
> Klaus Darilion wrote:
>> Hi!
>>
>> I see lots of examples using , but core show application displays |
>>
>> So what is the correct character to use to separate parameters for 
>> application, functions and macros?
> 


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Re: [asterisk-users] multiple asterisks in a server

[Klaus Darilion]

Klaus Darilion wrote:
> Rilawich Ango wrote:
>> Hi all,
>>   Is it possible to install more than 1 asterisk in a single server?
>> If yes, what do I need to set and take care?
> 
> Just to have several Asterisk instances on a single server you do not 
> need to install it multiple times. Install it once and start it multiple 
> times.
> 
> Of course you have to have a dedicated configuration for each server, eg:
> /etc/asterisk/instance1/*
> /etc/asterisk/instance2/*
> /etc/asterisk/instance3/*
> 
> Then you start the Asterisk process and specify the location of the 
> asterisk.conf file.
> 
> asterisk -C /etc/asterisk/instance1/asterisk.conf
> asterisk -C /etc/asterisk/instance2/asterisk.conf
> asterisk -C /etc/asterisk/instance3/asterisk.conf
> 
> Further, in asterisk.conf specify for each asterisk instance a different 
> location of: spool directory, PID file, 

btw: I use a common /var/lib/asterisk/ as I want to have the same 
"sounds" for all instances. This gives a problem when you use 1.4, as 
1.4 can not configure the location of astdb. For these you have to apply 
this patch:
http://bugs.digium.com/view.php?id=14257

regards
klaus

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Re: [asterisk-users] receiving 1st digit from a variable

[Danny Nicholas]

Syntax should be
exten => _[0-1]X.,1,Set(MSNCHOICE=${EXTEN:1:1})

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: Tuesday, February 24, 2009 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] receiving 1st digit from a variable

exten => _[0-1]X.,1,Set(MSNCHOICE=${EXTEN,1:1})

brings me this output:

Executing [1017649374...@officeie:1] Set("SIP/2000-007acf80",
"MSNCHOICE=") in new stack


and the result is always empty!

even if I make ${EXTEN,1:3} or whaever


Danny Nicholas schrieb:
> Exten => x,n,Set($NEWVAR=${EXTEN,1:1})
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
> Sent: Tuesday, February 24, 2009 2:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] receiving 1st digit from a variable
>
> Hi people!
> I want to save the 1st letter from the ${EXTEN} variable. I don't want
> to trim it, I want to RESAVE it into a new variable.
>
> Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0
>
> I would thank you for all advises.
>
>
>
> Tamer
>
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Re: [asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card

[John Novack]

What does Sangoma have to say?
They have, in the past, given excellent support and their cards have a 5 
year warranty.


Best try there first.

John Novack



Ketema Harris wrote:
Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ 
FXS Analog AFT card set up properly.


The main issue is that the card has four ports and as far as I can  
tell Asterisk is only seeing two.  On the two that it recognizes the  
"Green" FXS ports are not green, they just are not lit.  The "RED" FXO  
ports are indeed red, but from what I have read your not supposed to  
plug the analog phone into those.  So from my standpoint its not  
"working"  below is some more info about my system.  Any help would be  
appreciated.


I am running the following:

http://wiki.sangoma.com/sangoma-hardware#A200

Asterisk 1.6.0.5
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
wanpipe-3.3.15.20
linux kernel 2.6.27

The card is detected, but I can only get two channels configured.   
Here is my chan_dahdi.conf


[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
progzone=us
tonezone=0
jbenable=yes

;Sangoma AFT-A200 [slot:4 bus:5 span:1]  
context=default
group=0
echocancel=yes
signalling=fxs_ks
channel => 1

context=default
group=0
echocancel=yes
signalling=fxs_ks
channel => 2

dahdi show version
DAHDI Version: 2.1.0.4 Echo Canceller: MG2

dahdi show status
Description  Alarms  IRQbpviol CRC4
Fra Codi Options  LBO
wrtdm Board 1OK  0  0  0   
CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)


dahdi show channels
Chan Extension  Context Language   MOH Interpret 
BlockedState
  pseudofrom-zaptel 
default In Service
   1default 
default In Service
   2default 
default In Service


Whenever asterisk starts or chan_dahdi.so is reloaded I get the  
following:


[Feb 24 15:23:41] WARNING[10206]: chan_dahdi.c:14304 process_dahdi:  
Ignoring signalling at line 38.

 -- Reconfigured channel 1, FXS Kewlstart signalling
[Feb 24 15:23:41] WARNING[10206]: chan_dahdi.c:14304 process_dahdi:  
Ignoring signalling at line 44.

 -- Reconfigured channel 2, FXS Kewlstart signalling

My goal is:

1) Get Dialtone for my analog phone.
2) Be able to send and make calls from my analog phone
3) Be able to receive and send faxes from my fax machine

My service provider is VOIP, so I will never be getting or sending  
calls from/to the traditional telco.  But I do believe I should be  
able to send and receive calls from this device out to my SIP provider.


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Re: [asterisk-users] Polycom Phones start to break up after being up a LONG time

[Mike]

There was an issue with the early 2.x SIP with Asterisk IIRC, but that's
been fixed for awhile.

 

I'm using Asterisk 1.4.23.1, with plenty of 3.1.1 and 3.1.2 phones. YMMV,
but it's possible your documentation is also out of date :-)

 

Mike

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, February 24, 2009 15:58
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phones start to break up after being
up a LONG time

 

This is probably out of date, but my Polycom documentation recommends
keeping the boot rom at less than 2.0 for use with Asterisk.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, February 24, 2009 2:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phones start to break up after being
up a LONG time

 

That`s an old version, I've had plenty of issues (nothing like what you
describe) since 2.2.0.  Try going to the latest and greatest (SIP 3.1.2), it
seems stable (as far as I can tell with about 200 phones, including many
501).

 

It's hard to troubleshoot something that hasn't been current for about a
year.

 

Mike

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry D.
Hassler
Sent: Tuesday, February 24, 2009 15:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Phones start to break up after being
up a LONG time

 

Folks, I haven't paid attention to these responses, sorry!

This appears to be an issue primarily on calls with an EXTERNAL leg, I'm
fairly certain it's on both inbound and outbound calls. The frequency of the
reports from this client are increasing, and although I was intending to do
a mass reboot of all the polycoms over the weekend, I have not yet, but will
have to do it tonight (2 more reports today).

All the external legs are via PRI.

The phones are 501 and 601's, running bootrom 3.2.2.0019 and SIP 2.2.0.0047.

There are many of these that ARE on POE switched (Cisco),

On Fri, Feb 20, 2009 at 12:34 PM, Jeff LaCoursiere  wrote:


You also don't mention if it is internal to internal or if there is an
external leg involved, and if so what type.

j


On Fri, 20 Feb 2009, Asterisk Asterisk wrote:

> That's interesting - I haven't noticed this with any of my installs. What
version of firmware and SIP?
>
>
>
>
> 
> From: Barry D. Hassler 
> To: Asterisk Users Mailing List - Non-Commercial Discussion

> Sent: Friday, February 20, 2009 8:41:33 AM
> Subject: [asterisk-users] Polycom Phones start to break up after being up
a LONG time
>
> Has anyone else encountered this? I have a fairly large installation (~50
phones, almost all Polycom 501's and a handful of 601's. We're running into
a number of phones on which the outbound voice (Polycom phone user doesn't
hear any problems, but the other end does) is breaking up occasionally --
enough to be noticeable and make you say "what?". In each case, rebooting
the phone has resolved the symptoms, but I'd like to know if there is a
known problem.
>
> most of these phones would be up for several months now (installed this
past summer), and unless there are any power outages, would not be restarted
specifically.
>
> I'm planning on restarting all the phones over the weekend, but as this is
a 24-hour operation, we'd like to avoid interrupting phones at all.
>
> --
> Barry D. Hassler
> President, HCST
>
> http://www.hcst.net/
> 937-427-9000
>
>
>
>

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President, HCST

http://www.hcst.net/
937-427-9000

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Re: [asterisk-users] receiving 1st digit from a variable

[Tamer Higazi]

exten => _[0-1]X.,1,Set(MSNCHOICE=${EXTEN,1:1})

brings me this output:

Executing [1017649374...@officeie:1] Set("SIP/2000-007acf80",
"MSNCHOICE=") in new stack


and the result is always empty!

even if I make ${EXTEN,1:3} or whaever


Danny Nicholas schrieb:
> Exten => x,n,Set($NEWVAR=${EXTEN,1:1})
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
> Sent: Tuesday, February 24, 2009 2:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] receiving 1st digit from a variable
>
> Hi people!
> I want to save the 1st letter from the ${EXTEN} variable. I don't want
> to trim it, I want to RESAVE it into a new variable.
>
> Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0
>
> I would thank you for all advises.
>
>
>
> Tamer
>
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>
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Re: [asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card

[Tzafrir Cohen]

On Tue, Feb 24, 2009 at 03:55:30PM -0500, Ketema Harris wrote:
> I forgot to add that I have tried changing the chan_dahdi.conf file by  
> adding:
> 
> context=default
> group=1
> echocancel=yes
> signalling=fxo_ks
> channel => 3
> 
> context=default
> group=1
> echocancel=yes
> signalling=fxo_ks
> channel => 4
> 
> This addition simply caused dahdi not to load at all:
> 
> [Feb 24 15:54:52] WARNING[11026]: chan_dahdi.c:14304 process_dahdi:  
> Ignoring signalling at line 38.
>  -- Reconfigured channel 1, FXS Kewlstart signalling
> [Feb 24 15:54:52] WARNING[11026]: chan_dahdi.c:14304 process_dahdi:  
> Ignoring signalling at line 44.
>  -- Reconfigured channel 2, FXS Kewlstart signalling
> [Feb 24 15:54:52] WARNING[11026]: chan_dahdi.c:14304 process_dahdi:  
> Ignoring signalling at line 50.
> [Feb 24 15:54:52] ERROR[11026]: chan_dahdi.c:13459 build_channels:  
> Unable to reconfigure channel '3'
> [Feb 24 15:54:52] WARNING[11026]: chan_dahdi.c:14667 reload: Reload of  
> chan_dahdi.so is unsuccessful!

Use 'dahdi restart' instead of 'reload'

-- 
   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] weird problem

[David fire]

use wireshark or somethink like it (tcpdump) and see if the "bye" is
reaching asterisk.
if this is the problem you can use rtptimeout option in the sip.conf or
iax.conf.
David

2009/2/23 Michael 

> I am running Asterisk 1.4.22.2, though I have also found this problem with
> 1.4.23.x
>
> Sometimes after I hang up the system continues to spew packets to my phone
> causing it to become unusable until I restart Asterisk.
>
> Michael
>
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Re: [asterisk-users] Polycom Phones start to break up after being up a LONG time

[Danny Nicholas]

This is probably out of date, but my Polycom documentation recommends
keeping the boot rom at less than 2.0 for use with Asterisk.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, February 24, 2009 2:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Polycom Phones start to break up after being
up a LONG time

 

That`s an old version, I've had plenty of issues (nothing like what you
describe) since 2.2.0.  Try going to the latest and greatest (SIP 3.1.2), it
seems stable (as far as I can tell with about 200 phones, including many
501).

 

It's hard to troubleshoot something that hasn't been current for about a
year.

 

Mike

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry D.
Hassler
Sent: Tuesday, February 24, 2009 15:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Phones start to break up after being
up a LONG time

 

Folks, I haven't paid attention to these responses, sorry!

This appears to be an issue primarily on calls with an EXTERNAL leg, I'm
fairly certain it's on both inbound and outbound calls. The frequency of the
reports from this client are increasing, and although I was intending to do
a mass reboot of all the polycoms over the weekend, I have not yet, but will
have to do it tonight (2 more reports today).

All the external legs are via PRI.

The phones are 501 and 601's, running bootrom 3.2.2.0019 and SIP 2.2.0.0047.

There are many of these that ARE on POE switched (Cisco),

On Fri, Feb 20, 2009 at 12:34 PM, Jeff LaCoursiere  wrote:


You also don't mention if it is internal to internal or if there is an
external leg involved, and if so what type.

j


On Fri, 20 Feb 2009, Asterisk Asterisk wrote:

> That's interesting - I haven't noticed this with any of my installs. What
version of firmware and SIP?
>
>
>
>
> 
> From: Barry D. Hassler 
> To: Asterisk Users Mailing List - Non-Commercial Discussion

> Sent: Friday, February 20, 2009 8:41:33 AM
> Subject: [asterisk-users] Polycom Phones start to break up after being up
a LONG time
>
> Has anyone else encountered this? I have a fairly large installation (~50
phones, almost all Polycom 501's and a handful of 601's. We're running into
a number of phones on which the outbound voice (Polycom phone user doesn't
hear any problems, but the other end does) is breaking up occasionally --
enough to be noticeable and make you say "what?". In each case, rebooting
the phone has resolved the symptoms, but I'd like to know if there is a
known problem.
>
> most of these phones would be up for several months now (installed this
past summer), and unless there are any power outages, would not be restarted
specifically.
>
> I'm planning on restarting all the phones over the weekend, but as this is
a 24-hour operation, we'd like to avoid interrupting phones at all.
>
> --
> Barry D. Hassler
> President, HCST
>
> http://www.hcst.net/
> 937-427-9000
>
>
>
>

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President, HCST

http://www.hcst.net/
937-427-9000

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Re: [asterisk-users] receiving 1st digit from a variable

[Danny Nicholas]

Exten => x,n,Set($NEWVAR=${EXTEN,1:1})

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: Tuesday, February 24, 2009 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] receiving 1st digit from a variable

Hi people!
I want to save the 1st letter from the ${EXTEN} variable. I don't want
to trim it, I want to RESAVE it into a new variable.

Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0

I would thank you for all advises.



Tamer

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Re: [asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card

[Ketema Harris]

I forgot to add that I have tried changing the chan_dahdi.conf file by  
adding:

context=default
group=1
echocancel=yes
signalling=fxo_ks
channel => 3

context=default
group=1
echocancel=yes
signalling=fxo_ks
channel => 4

This addition simply caused dahdi not to load at all:

[Feb 24 15:54:52] WARNING[11026]: chan_dahdi.c:14304 process_dahdi:  
Ignoring signalling at line 38.
 -- Reconfigured channel 1, FXS Kewlstart signalling
[Feb 24 15:54:52] WARNING[11026]: chan_dahdi.c:14304 process_dahdi:  
Ignoring signalling at line 44.
 -- Reconfigured channel 2, FXS Kewlstart signalling
[Feb 24 15:54:52] WARNING[11026]: chan_dahdi.c:14304 process_dahdi:  
Ignoring signalling at line 50.
[Feb 24 15:54:52] ERROR[11026]: chan_dahdi.c:13459 build_channels:  
Unable to reconfigure channel '3'
[Feb 24 15:54:52] WARNING[11026]: chan_dahdi.c:14667 reload: Reload of  
chan_dahdi.so is unsuccessful!

On Feb 24, 2009, at 3:41 PM, Ketema Harris wrote:

> Hi I have been having a rough time getting a Sangoma A200/Remora FXO/
> FXS Analog AFT card set up properly.
>
> The main issue is that the card has four ports and as far as I can
> tell Asterisk is only seeing two.  On the two that it recognizes the
> "Green" FXS ports are not green, they just are not lit.  The "RED" FXO
> ports are indeed red, but from what I have read your not supposed to
> plug the analog phone into those.  So from my standpoint its not
> "working"  below is some more info about my system.  Any help would be
> appreciated.
>
> I am running the following:
>
> http://wiki.sangoma.com/sangoma-hardware#A200
>
> Asterisk 1.6.0.5
> dahdi-linux-2.1.0.4
> dahdi-tools-2.1.0.2
> wanpipe-3.3.15.20
> linux kernel 2.6.27
>
> The card is detected, but I can only get two channels configured.
> Here is my chan_dahdi.conf
>
> [trunkgroups]
>
> [channels]
> context=default
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> progzone=us
> tonezone=0
> jbenable=yes
>
> ;Sangoma AFT-A200 [slot:4 bus:5 span:1]  
> context=default
> group=0
> echocancel=yes
> signalling=fxs_ks
> channel => 1
>
> context=default
> group=0
> echocancel=yes
> signalling=fxs_ks
> channel => 2
>
> dahdi show version
> DAHDI Version: 2.1.0.4 Echo Canceller: MG2
>
> dahdi show status
> Description  Alarms  IRQbpviol CRC4
> Fra Codi Options  LBO
> wrtdm Board 1OK  0  0  0
> CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)
>
> dahdi show channels
>Chan Extension  Context Language   MOH Interpret
> BlockedState
>  pseudofrom-zaptel
> default In Service
>   1default
> default In Service
>   2default
> default In Service
>
> Whenever asterisk starts or chan_dahdi.so is reloaded I get the
> following:
>
> [Feb 24 15:23:41] WARNING[10206]: chan_dahdi.c:14304 process_dahdi:
> Ignoring signalling at line 38.
> -- Reconfigured channel 1, FXS Kewlstart signalling
> [Feb 24 15:23:41] WARNING[10206]: chan_dahdi.c:14304 process_dahdi:
> Ignoring signalling at line 44.
> -- Reconfigured channel 2, FXS Kewlstart signalling
>
> My goal is:
>
> 1) Get Dialtone for my analog phone.
> 2) Be able to send and make calls from my analog phone
> 3) Be able to receive and send faxes from my fax machine
>
> My service provider is VOIP, so I will never be getting or sending
> calls from/to the traditional telco.  But I do believe I should be
> able to send and receive calls from this device out to my SIP  
> provider.
>
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Re: [asterisk-users] Polycom Phones start to break up after being up a LONG time

[Mike]

That`s an old version, I've had plenty of issues (nothing like what you
describe) since 2.2.0.  Try going to the latest and greatest (SIP 3.1.2), it
seems stable (as far as I can tell with about 200 phones, including many
501).

 

It's hard to troubleshoot something that hasn't been current for about a
year.

 

Mike

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry D.
Hassler
Sent: Tuesday, February 24, 2009 15:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Phones start to break up after being
up a LONG time

 

Folks, I haven't paid attention to these responses, sorry!

This appears to be an issue primarily on calls with an EXTERNAL leg, I'm
fairly certain it's on both inbound and outbound calls. The frequency of the
reports from this client are increasing, and although I was intending to do
a mass reboot of all the polycoms over the weekend, I have not yet, but will
have to do it tonight (2 more reports today).

All the external legs are via PRI.

The phones are 501 and 601's, running bootrom 3.2.2.0019 and SIP 2.2.0.0047.

There are many of these that ARE on POE switched (Cisco),

On Fri, Feb 20, 2009 at 12:34 PM, Jeff LaCoursiere  wrote:


You also don't mention if it is internal to internal or if there is an
external leg involved, and if so what type.

j


On Fri, 20 Feb 2009, Asterisk Asterisk wrote:

> That's interesting - I haven't noticed this with any of my installs. What
version of firmware and SIP?
>
>
>
>
> 
> From: Barry D. Hassler 
> To: Asterisk Users Mailing List - Non-Commercial Discussion

> Sent: Friday, February 20, 2009 8:41:33 AM
> Subject: [asterisk-users] Polycom Phones start to break up after being up
a LONG time
>
> Has anyone else encountered this? I have a fairly large installation (~50
phones, almost all Polycom 501's and a handful of 601's. We're running into
a number of phones on which the outbound voice (Polycom phone user doesn't
hear any problems, but the other end does) is breaking up occasionally --
enough to be noticeable and make you say "what?". In each case, rebooting
the phone has resolved the symptoms, but I'd like to know if there is a
known problem.
>
> most of these phones would be up for several months now (installed this
past summer), and unless there are any power outages, would not be restarted
specifically.
>
> I'm planning on restarting all the phones over the weekend, but as this is
a 24-hour operation, we'd like to avoid interrupting phones at all.
>
> --
> Barry D. Hassler
> President, HCST
>
> http://www.hcst.net/
> 937-427-9000
>
>
>
>

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President, HCST

http://www.hcst.net/
937-427-9000

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[asterisk-users] receiving 1st digit from a variable

[Tamer Higazi]

Hi people!
I want to save the 1st letter from the ${EXTEN} variable. I don't want
to trim it, I want to RESAVE it into a new variable.

Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0

I would thank you for all advises.



Tamer

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Re: [asterisk-users] Aastra phones

[Mike]

Sure, but I need FTP (phones are not on the LAN and TFTP is too unsecured).

I'll try updating to a more recent (with a 2-step update) version and see if
that enables FTP firmware upgrade.


Mike

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Ira
> Sent: Tuesday, February 24, 2009 15:22
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Aastra phones
> 
> At 12:13 PM 2/24/2009, you wrote:
> >I was hoping for a unattended firmware upgrade.  Let's say I have 50
> phones
> >out in the field; just changing the firmware file (i.e.: replacing
> 9143i.st)
> >would upgrade all phones, without any intervention on my part.
> >
> >That's what I do with my 200+ Polycom phones.
> >
> >Possible or not?
> 
> 
> That's what I do on my 3 Aastra phones. Install new firmware in the
> TFTP folder and reset my phone, if it all goes well, I pull the plug
> on the switch and they all update.
> 
> Ira
> 
> 
> 
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[asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Here's one that may be of interest to any upgraders.  If you rely on the
behavior of gosub you may want to make note of this change.

I have an incoming call context:

exten => _,n,GoSub(incoming,${EXTEN},1(${EXTEN}));

that is supposed to gosub into the incoming extension at priority 1.
Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the
requested extension wasn't present in the incoming context.

When I upgraded to 1.6.0.6 this behavior changed and I would simply get
an error on the console that a matching extension was not found, and the
dialplan would simply stop.  It was easy enough to add:

[incoming]
exten => _,1,Goto(i,1)

to restore the previous behavior (I'm looking at four-digits from a PRI)
which I should probably have done anyway.

I don't know if this is a bug or WAD but just wanted to mention it.

Barry



-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

iD8DBQFJpE4ZCFu3bIiwtTARAlELAKCKFKpIsUGf44yZBcx/kpYnzSpelACgoOqB
iYIg4keZ5EIL35rrLwCRdTU=
=fvE0
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[asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card

[Ketema Harris]

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ 
FXS Analog AFT card set up properly.

The main issue is that the card has four ports and as far as I can  
tell Asterisk is only seeing two.  On the two that it recognizes the  
"Green" FXS ports are not green, they just are not lit.  The "RED" FXO  
ports are indeed red, but from what I have read your not supposed to  
plug the analog phone into those.  So from my standpoint its not  
"working"  below is some more info about my system.  Any help would be  
appreciated.

I am running the following:

http://wiki.sangoma.com/sangoma-hardware#A200

Asterisk 1.6.0.5
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
wanpipe-3.3.15.20
linux kernel 2.6.27

The card is detected, but I can only get two channels configured.   
Here is my chan_dahdi.conf

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
progzone=us
tonezone=0
jbenable=yes

;Sangoma AFT-A200 [slot:4 bus:5 span:1]  
context=default
group=0
echocancel=yes
signalling=fxs_ks
channel => 1

context=default
group=0
echocancel=yes
signalling=fxs_ks
channel => 2

dahdi show version
DAHDI Version: 2.1.0.4 Echo Canceller: MG2

dahdi show status
Description  Alarms  IRQbpviol CRC4
Fra Codi Options  LBO
wrtdm Board 1OK  0  0  0   
CAS Unk  YEL  0 db (CSU)/0-133 feet (DSX-1)

dahdi show channels
Chan Extension  Context Language   MOH Interpret 
BlockedState
  pseudofrom-zaptel 
default In Service
   1default 
default In Service
   2default 
default In Service

Whenever asterisk starts or chan_dahdi.so is reloaded I get the  
following:

[Feb 24 15:23:41] WARNING[10206]: chan_dahdi.c:14304 process_dahdi:  
Ignoring signalling at line 38.
 -- Reconfigured channel 1, FXS Kewlstart signalling
[Feb 24 15:23:41] WARNING[10206]: chan_dahdi.c:14304 process_dahdi:  
Ignoring signalling at line 44.
 -- Reconfigured channel 2, FXS Kewlstart signalling

My goal is:

1) Get Dialtone for my analog phone.
2) Be able to send and make calls from my analog phone
3) Be able to receive and send faxes from my fax machine

My service provider is VOIP, so I will never be getting or sending  
calls from/to the traditional telco.  But I do believe I should be  
able to send and receive calls from this device out to my SIP provider.

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Re: [asterisk-users] Polycom Phones start to break up after being up a LONG time

[Barry D. Hassler]

Folks, I haven't paid attention to these responses, sorry!

This appears to be an issue primarily on calls with an EXTERNAL leg, I'm
fairly certain it's on both inbound and outbound calls. The frequency of the
reports from this client are increasing, and although I was intending to do
a mass reboot of all the polycoms over the weekend, I have not yet, but will
have to do it tonight (2 more reports today).

All the external legs are via PRI.

The phones are 501 and 601's, running bootrom 3.2.2.0019 and SIP 2.2.0.0047.

There are many of these that ARE on POE switched (Cisco),

On Fri, Feb 20, 2009 at 12:34 PM, Jeff LaCoursiere  wrote:

>
> You also don't mention if it is internal to internal or if there is an
> external leg involved, and if so what type.
>
> j
>
> On Fri, 20 Feb 2009, Asterisk Asterisk wrote:
>
> > That's interesting - I haven't noticed this with any of my installs. What
> version of firmware and SIP?
> >
> >
> >
> >
> > 
> > From: Barry D. Hassler 
> > To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> > Sent: Friday, February 20, 2009 8:41:33 AM
> > Subject: [asterisk-users] Polycom Phones start to break up after being up
> a LONG time
> >
> > Has anyone else encountered this? I have a fairly large installation (~50
> phones, almost all Polycom 501's and a handful of 601's. We're running into
> a number of phones on which the outbound voice (Polycom phone user doesn't
> hear any problems, but the other end does) is breaking up occasionally --
> enough to be noticeable and make you say "what?". In each case, rebooting
> the phone has resolved the symptoms, but I'd like to know if there is a
> known problem.
> >
> > most of these phones would be up for several months now (installed this
> past summer), and unless there are any power outages, would not be restarted
> specifically.
> >
> > I'm planning on restarting all the phones over the weekend, but as this
> is a 24-hour operation, we'd like to avoid interrupting phones at all.
> >
> > --
> > Barry D. Hassler
> > President, HCST
> >
> > http://www.hcst.net/
> > 937-427-9000
> >
> >
> >
> >
>
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-- 
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000
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Re: [asterisk-users] Aastra phones

[Ira]

At 12:13 PM 2/24/2009, you wrote:
>I was hoping for a unattended firmware upgrade.  Let's say I have 50 phones
>out in the field; just changing the firmware file (i.e.: replacing 9143i.st)
>would upgrade all phones, without any intervention on my part.
>
>That's what I do with my 200+ Polycom phones.
>
>Possible or not?


That's what I do on my 3 Aastra phones. Install new firmware in the 
TFTP folder and reset my phone, if it all goes well, I pull the plug 
on the switch and they all update.

Ira



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Re: [asterisk-users] Aastra phones

[Mike]

I was hoping for a unattended firmware upgrade.  Let's say I have 50 phones
out in the field; just changing the firmware file (i.e.: replacing 9143i.st)
would upgrade all phones, without any intervention on my part.

That's what I do with my 200+ Polycom phones.

Possible or not?  

Mike



> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of M Hulber
> Sent: Tuesday, February 24, 2009 12:45
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Aastra phones
> 
> On my other Aastra phone there is a config in the phone menu to specify
> the name of the firmware file under Advanced Settings / Firmware
> Update.  If you have access to that you might see what name it's looking
> for.  I suppose you would be able to set this name in the config files
> also but I don't know the setting.
> 
> Mike wrote:
> >
> > Hi,
> >
> >
> >
> > I`ve been toying with an Aastra phone (9143i) wondering if it could be
> > a good alternative to to the more expensive Polycom phones.
> >
> >
> >
> > One thing which I can't figure out, although it certainly looks
> > simple, is to update the firmware though FTP (not TFTP).  I have set
> > the ftp provisioning server in the Aastra phone, and put the firmware
> > file 9143i.st in the root folder where the login/password pair ends
> > up. Everything is entered correctly, or so it seems (works fine with
> > my Polycoms).
> >
> >
> >
> > When I reboot the phone from the Web UI, it doesn't seem to take in
> > the new firmware.  But it does seem to download the (empty) aastra.cfg
> > file (proving that the provisoning server settings are correct).
> >
> >
> >
> > What am I missing?
> >
> >
> >
> >
> >
> >
> >
> > 
> >
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Re: [asterisk-users] chan_sip and database integration

[John Todd]

On Feb 24, 2009, at 12:33 PM, Klaus Darilion wrote:
> Johansson Olle E wrote:
>> 24 feb 2009 kl. 11.31 skrev Michiel van Baak:
>>> On 10:56, Tue 24 Feb 09, Klaus Darilion wrote:
 Hi!

 I tried to understand how chan_sip can be configured by means of a
 database. I found these 2 different approaches (please correct me
 if I
 am wrong):

 static configuration: the sip.conf file is mapped to a database
 table.
 The table contains one line for each line in sip.conf.

 realtime configuration: the peers/users are stored in the database
 using
 a single line for each peer/user.


 "Static" does not eases provisioning as configuring a SIP peer/user
 using this approach is really complicated - it is just a method to
 store
 .conf files in database.

 "realtime" really eases provisioning of SIP peers/users. You only
 have
 to insert/update/delete a single line. But functionality is
 different -
 there are limitations as these objects are not stored in memory
 (can be
 cached), for example device status information.


 What I am looking for is a method to provision peers/users with a
 single
 line in the database, but without limitations. Thus, the peers need
 not
 to be realtime but are loaded on "sip reload".

 So is there a possiblity to have static peer/users configuration
 using a
 nice and easy way?
>>> Store them in a database and use a combination of cron and some
>>> scripting to generate the configuration files.
>>>
>>> Some advice: keep track if an update has been done to the database
>>> since
>>> last reload and only regen files and issue a reload when this is  
>>> true.
>>
>> I think this is what FreePBX does.
>
> But wouldn't it be great for Asterisk to support the realtime DB  
> schema
> also for static peers/users?
>
> klaus


Of course, it would be great!  It sounds like this is of some value to  
you, and I expect it would be useful to others, as well.  So there is  
a high incentive for someone to code it - maybe you!  :-)

Another solution, though I don't think it does what you want, would be  
to use the "#exec" command in your configuration files, which then  
calls a script to build the config file.

Honestly, I don't know if #exec gets called every time that the  
dialplan is reloaded - I'd assume the answer to this is "yes" but I  
can't test - see PPS below.

PS: Remember to enable #exec mode in /etc/asterisk/asterisk.conf - it  
is disabled by default.

PPS: It seems that #exec is broken in SVN-TRUNK.  I've posted a bug. 
http://bugs.digium.com/view.php?id=14542

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] multiple asterisks in a server

[Geraint Lee]

should have thought of that one lol

Cheers for the tip... will be changing my setup to this lol

2009/2/24 Klaus Darilion 

> Rilawich Ango wrote:
> > Hi all,
> >   Is it possible to install more than 1 asterisk in a single server?
> > If yes, what do I need to set and take care?
>
> Just to have several Asterisk instances on a single server you do not
> need to install it multiple times. Install it once and start it multiple
> times.
>
> Of course you have to have a dedicated configuration for each server, eg:
> /etc/asterisk/instance1/*
> /etc/asterisk/instance2/*
> /etc/asterisk/instance3/*
>
> Then you start the Asterisk process and specify the location of the
> asterisk.conf file.
>
> asterisk -C /etc/asterisk/instance1/asterisk.conf
> asterisk -C /etc/asterisk/instance2/asterisk.conf
> asterisk -C /etc/asterisk/instance3/asterisk.conf
>
> Further, in asterisk.conf specify for each asterisk instance a different
> location of: spool directory, PID file, 
>
> regards
> klaus
>
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Re: [asterisk-users] what is the correct character to separate application parameters: , or |

[Eric Wieling, Asteria Solutions Group]

Versions before 0.65 I don't know about
 From 0.65 to 1.4.x you can use either , or |
In 1.6 you must use , (| was removed)


Klaus Darilion wrote:
> Hi!
> 
> I see lots of examples using , but core show application displays |
> 
> So what is the correct character to use to separate parameters for 
> application, functions and macros?

-- 
Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com

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Re: [asterisk-users] what is the correct character to separate application parameters: , or |

[Danny Nicholas]

You code it in extensions.conf as "," then when it is interpreted, the ","
is changed to a "|".  Probably some C anomaly.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion
Sent: Tuesday, February 24, 2009 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] what is the correct character to separate
application parameters: , or |

Hi!

I see lots of examples using , but core show application displays |

So what is the correct character to use to separate parameters for 
application, functions and macros?

thanks
klaus

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Re: [asterisk-users] Swichted digits on received number from fax on an fxs port

[Loic Didelot]

Hello,
I do not mean the caller ID, I am talking about the dialed extension.

Best regards,
Loïc Didelot.

On Tue, 2009-02-24 at 13:34 +0200, Tzafrir Cohen wrote:
> On Tue, Feb 24, 2009 at 10:15:27AM +0100, Loic Didelot wrote:
> > Hello,
> > I have connected a fax machine to a xorcom fxs port (signalling=fxo_ks).
> > But when I try to dial (send a fax) the number that I receive in
> > asterisk is wrong. Quite often a few digits are wrong but sometime is
> > correct. It looks like it works 2 times out of 10.
> > 
> > Examples: 
> > 2090 becomes 2999 or 2000
> > 1234567890 becomes 1234566790
> 
> On fxo_ks the phone does not get to set the caller ID. Where do you set
> the caller ID? Where do you read it?
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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[asterisk-users] what is the correct character to separate application parameters: , or |

[Klaus Darilion]

Hi!

I see lots of examples using , but core show application displays |

So what is the correct character to use to separate parameters for 
application, functions and macros?

thanks
klaus

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Re: [asterisk-users] multiple asterisks in a server

[Klaus Darilion]

Rilawich Ango wrote:
> Hi all,
>   Is it possible to install more than 1 asterisk in a single server?
> If yes, what do I need to set and take care?

Just to have several Asterisk instances on a single server you do not 
need to install it multiple times. Install it once and start it multiple 
times.

Of course you have to have a dedicated configuration for each server, eg:
/etc/asterisk/instance1/*
/etc/asterisk/instance2/*
/etc/asterisk/instance3/*

Then you start the Asterisk process and specify the location of the 
asterisk.conf file.

asterisk -C /etc/asterisk/instance1/asterisk.conf
asterisk -C /etc/asterisk/instance2/asterisk.conf
asterisk -C /etc/asterisk/instance3/asterisk.conf

Further, in asterisk.conf specify for each asterisk instance a different 
location of: spool directory, PID file, 

regards
klaus

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Re: [asterisk-users] Aastra phones

[Greg Kennedy]

> 
> Hi,
> 
>  
> 
> I`ve been toying with an Aastra phone (9143i) wondering if it could be a
> good alternative to to the more expensive Polycom phones.
> 
>  
> 
> One thing which I can't figure out, although it certainly looks simple, is
> to update the firmware though FTP (not TFTP).  I have set the ftp
> provisioning server in the Aastra phone, and put the firmware file 9143i.st
> in the root folder where the login/password pair ends up. Everything is
> entered correctly, or so it seems (works fine with my Polycoms).
> 
>  
> 
> When I reboot the phone from the Web UI, it doesn't seem to take in the new
> firmware.  But it does seem to download the (empty) aastra.cfg file (proving
> that the provisoning server settings are correct).
> 
>  
> 
> What am I missing?
**

The phones by default look for a file called firmware.st, make sure you update 
the file name in the phone config to grab that 9143.st, or rename the file on 
your ftp server.

If you are running any modern redhat based linux distro you can also do the 
following:
yum install tftp-server
edit /etc/xinet.d/tftp
change the enabled=false to true

/etc/init.d/xinetd restart
copy the firmware file to the tftproot directory as defined in the above tftp 
file, normally /tftpboot and you should be all set.

Hope that helps.

Greg
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Re: [asterisk-users] chan_sip and database integration

[Klaus Darilion]

Johansson Olle E wrote:
> 24 feb 2009 kl. 11.31 skrev Michiel van Baak:
> 
>> On 10:56, Tue 24 Feb 09, Klaus Darilion wrote:
>>> Hi!
>>>
>>> I tried to understand how chan_sip can be configured by means of a
>>> database. I found these 2 different approaches (please correct me  
>>> if I
>>> am wrong):
>>>
>>> static configuration: the sip.conf file is mapped to a database  
>>> table.
>>> The table contains one line for each line in sip.conf.
>>>
>>> realtime configuration: the peers/users are stored in the database  
>>> using
>>> a single line for each peer/user.
>>>
>>>
>>> "Static" does not eases provisioning as configuring a SIP peer/user
>>> using this approach is really complicated - it is just a method to  
>>> store
>>> .conf files in database.
>>>
>>> "realtime" really eases provisioning of SIP peers/users. You only  
>>> have
>>> to insert/update/delete a single line. But functionality is  
>>> different -
>>> there are limitations as these objects are not stored in memory  
>>> (can be
>>> cached), for example device status information.
>>>
>>>
>>> What I am looking for is a method to provision peers/users with a  
>>> single
>>> line in the database, but without limitations. Thus, the peers need  
>>> not
>>> to be realtime but are loaded on "sip reload".
>>>
>>> So is there a possiblity to have static peer/users configuration  
>>> using a
>>> nice and easy way?
>> Store them in a database and use a combination of cron and some
>> scripting to generate the configuration files.
>>
>> Some advice: keep track if an update has been done to the database  
>> since
>> last reload and only regen files and issue a reload when this is true.
> 
> I think this is what FreePBX does.

But wouldn't it be great for Asterisk to support the realtime DB schema 
also for static peers/users?

klaus

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Re: [asterisk-users] building asterisk-1.6.0.6 failed!

[Tamer Higazi]

System: Gentoo Linux 2008.0, 64 Bit


About package content (libsrtp 1.4.4):
/*
 * err.h
 *
 * error status codes
 *
 * David A. McGrew
 * Cisco Systems, Inc.
 */
/*


and I opested at digium bugs something:
http://bugs.digium.com/view.php?id=14535

the supporter wants me to rename "/usr/local" => "/usr/local2" what
sounds for me more then strange (to take a whole partition out.

Tamer

Tzafrir Cohen schrieb:
> On Tue, Feb 24, 2009 at 09:31:21AM +0100, Tamer Higazi wrote:
>   
>> I did the same thing, without the prefix stuff!
>>
>> The same error!
>>
>>[CC] extconf.c -> extconf.o
>> In file included from /usr/local/include/datatypes.h:50,
>>  from /usr/local/include/err.h:49,
>>  from extconf.c:45:
>> /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file
>> or directory
>> In file included from /usr/local/include/datatypes.h:50,
>>  from /usr/local/include/err.h:49,
>>  from extconf.c:45:
>> /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t'
>> /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here
>> make[1]: *** [extconf.o] Error 1
>> make: *** [utils] Error 2
>> ta...@tux /tmp/asterisk-1.6.0.6 $
>> 
>
> What system is it, exactly?
>
> If Linux: what distribution? What version?
> If not: what OS? What version?
>
> In what package (or whatever) is /usr/local/include/err.h included?
>
>   


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Re: [asterisk-users] Aastra phones

[Ira]

At 09:33 AM 2/24/2009, you wrote:

>When I reboot the phone from the Web UI, it doesn't seem to take in 
>the new firmware.  But it does seem to download the (empty) 
>aastra.cfg file (proving that the provisoning server settings are correct).
>
>
>What am I missing?

I have three 480iCTs and it's always worked fine but I think I have 
to rename the downloaded files to what the phone is looking for. I've 
no clue to tell you what that might be, but I'd guess if you figure 
it out, all will start working as expected.

Ira 


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Re: [asterisk-users] building asterisk-1.6.0.6 failed!

[M Hulber]

Thanks Leif.  That worked.

Leif Madsen wrote:
> Most likely you don't need app_dahdiras, so you can just disable it from 
> menuselect under the Applications section.
>
> In addition, Terry Wilson is working on getting this resolved, perhaps 
> sometime 
> today, so any fixes for it will be in the next release.
>
> Thanks!
> Leif Madsen.
>
> M Hulber wrote:
>   
>> I'm having a different problem building this release:
>>
>> [CC] app_dahdiras.c -> app_dahdiras.o
>> In file included from app_dahdiras.c:50:
>> /usr/include/dahdi/user.h:736: error: expected specifier-qualifier-list 
>> before ‘__s32’
>> /usr/include/dahdi/user.h:939: error: expected specifier-qualifier-list 
>> before ‘__s32’
>> make[1]: *** [app_dahdiras.o] Error 1
>> make: *** [apps] Error 2
>> 
>
>
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Re: [asterisk-users] Aastra phones

[M Hulber]

On my other Aastra phone there is a config in the phone menu to specify 
the name of the firmware file under Advanced Settings / Firmware 
Update.  If you have access to that you might see what name it's looking 
for.  I suppose you would be able to set this name in the config files 
also but I don't know the setting.

Mike wrote:
>
> Hi,
>
>  
>
> I`ve been toying with an Aastra phone (9143i) wondering if it could be 
> a good alternative to to the more expensive Polycom phones.
>
>  
>
> One thing which I can't figure out, although it certainly looks 
> simple, is to update the firmware though FTP (not TFTP).  I have set 
> the ftp provisioning server in the Aastra phone, and put the firmware 
> file 9143i.st in the root folder where the login/password pair ends 
> up. Everything is entered correctly, or so it seems (works fine with 
> my Polycoms).
>
>  
>
> When I reboot the phone from the Web UI, it doesn't seem to take in 
> the new firmware.  But it does seem to download the (empty) aastra.cfg 
> file (proving that the provisoning server settings are correct).
>
>  
>
> What am I missing?
>
>  
>
>  
>
>  
>
> 
>
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Re: [asterisk-users] Aastra phones

[John Millican]

Mike wrote:
> Hi,
> 
>  
> 
> I`ve been toying with an Aastra phone (9143i) wondering if it could be a
> good alternative to to the more expensive Polycom phones.
> 
>  
> 
> One thing which I can't figure out, although it certainly looks simple,
> is to update the firmware though FTP (not TFTP).  I have set the ftp
> provisioning server in the Aastra phone, and put the firmware file
> 9143i.st in the root folder where the login/password pair ends up.
> Everything is entered correctly, or so it seems (works fine with my
> Polycoms).
> 
>  
> 
> When I reboot the phone from the Web UI, it doesn't seem to take in the
> new firmware.  But it does seem to download the (empty) aastra.cfg file
> (proving that the provisoning server settings are correct).
> 
>  
> 
> What am I missing?
> 

I believe that the older firmware for the Aastra phone will only update
from TFTP.  I am not sure what rev level this changed at though.
JohnM


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Re: [asterisk-users] building asterisk-1.6.0.6 failed!

[Leif Madsen]

In addition, I just found the bug that contains the same error you mention. It 
has been assigned and will be resolved for the next Asterisk release.

http://bugs.digium.com/view.php?id=14516

Thanks!
Leif Madsen.

M Hulber wrote:
> I'm having a different problem building this release:
> 
> [CC] app_dahdiras.c -> app_dahdiras.o
> In file included from app_dahdiras.c:50:
> /usr/include/dahdi/user.h:736: error: expected specifier-qualifier-list 
> before ‘__s32’
> /usr/include/dahdi/user.h:939: error: expected specifier-qualifier-list 
> before ‘__s32’
> make[1]: *** [app_dahdiras.o] Error 1
> make: *** [apps] Error 2


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[asterisk-users] Aastra phones

[Mike]

Hi,

 

I`ve been toying with an Aastra phone (9143i) wondering if it could be a
good alternative to to the more expensive Polycom phones.

 

One thing which I can't figure out, although it certainly looks simple, is
to update the firmware though FTP (not TFTP).  I have set the ftp
provisioning server in the Aastra phone, and put the firmware file 9143i.st
in the root folder where the login/password pair ends up. Everything is
entered correctly, or so it seems (works fine with my Polycoms).

 

When I reboot the phone from the Web UI, it doesn't seem to take in the new
firmware.  But it does seem to download the (empty) aastra.cfg file (proving
that the provisoning server settings are correct).

 

What am I missing?

 

 

 

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Re: [asterisk-users] multiple asterisks in a server

[Geraint Lee]

I do it for CDR, when using the originate command via the manager and
initiate a call to a phone which then connects to an agi script upon answer,
the cdr stops at the point of answer and there is no other created, which of
course is useless for billing customers - there may very well be a way to
make the cdr continue after it seems to stop logging, or is it a bug? either
way, the quickest solution for me was to install a second copy and send all
calls out on a second installation with accurate cdr logging.

2009/2/24 David Backeberg 

> On Tue, Feb 24, 2009 at 2:59 AM, Rilawich Ango 
> wrote:
> > Hi all,
> >  Is it possible to install more than 1 asterisk in a single server?
>
> Can somebody help me understand why you would want to do this?
>
> I suppose development versus production, but wouldn't you also want
> better separation, like virtualization?
> - Show quoted text -
>
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Re: [asterisk-users] building asterisk-1.6.0.6 failed!

[Leif Madsen]

Most likely you don't need app_dahdiras, so you can just disable it from 
menuselect under the Applications section.

In addition, Terry Wilson is working on getting this resolved, perhaps sometime 
today, so any fixes for it will be in the next release.

Thanks!
Leif Madsen.

M Hulber wrote:
> I'm having a different problem building this release:
> 
> [CC] app_dahdiras.c -> app_dahdiras.o
> In file included from app_dahdiras.c:50:
> /usr/include/dahdi/user.h:736: error: expected specifier-qualifier-list 
> before ‘__s32’
> /usr/include/dahdi/user.h:939: error: expected specifier-qualifier-list 
> before ‘__s32’
> make[1]: *** [app_dahdiras.o] Error 1
> make: *** [apps] Error 2


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Re: [asterisk-users] Polycom Spectralink 8002 Configuration

[M Hulber]

Since I have a partial answer for completeness:

When I moved (duplicated) the PROXYn_ADDRESS definition to the user 
specific config it then registers with the proxy.  It seems like a 
defect to me.

I'd still like to see how people have set up different users/lines using 
LINEn and LINEn_AUTH.  I'm struggling a bit with the relationship 
between the user registrations in the phone admin and the lines/users in 
the sip_.cfg and sip.conf.

M Hulber wrote:
> I have a new Polycom Spectralink 8002 and am having trouble with the 
> configuration or the unit but I can't see what's wrong.  The unit does 
> not seem to even attempt to register with the Asterisk proxy but I can 
> make calls to it.  I have viewed the syslog from the device which it 
> will actually write to the asterisk server so I know it can be reached.  
> I have also run a sip debug and see no registration traffic from the 
> unit.  It also pulls the configs from the tftp server on the asterisk 
> box ok.
>
> Does anyone have a sample set of configs that work?  I have samples for 
> the Polycom side but haven't seen the match on the asterisk side.  Since 
> I don't even see traffic, I can't think that it's even an authentication 
> issue.
>
> When I dial from the device it just sits there, basically.
>
> MARK.
>
> -- 
>
> sip_allusers.cfg:  (I've tried most variations on theses settings)
>
> ## FOR PROXY1_TYPE = ASTERISK
>
> #PROXY1_ADDR = 192.168.2.80:5060# replace the ip address with 
> the Asterisk Server's Address  
> PROXY1_ADDR = 192.168.2.80  # replace the ip address with the 
> Asterisk Server's Address  
> PROXY1_KEYPRESS_2833 = enable
> PROXY1_KEYPRESS_INFO = enable
> PROXY1_HOLD_IP0 = disable
> PROXY1_PRACK = enable
> #PROXY1_REREG_SECS=3600
> PROXY1_REREG_SECS=35
> PROXY1_KEEPALIVE_SECS=14
> #PROXY1_DOMAIN = asterisk# Replace this with your SIP Domain's name
> PROXY1_CALLID_PER_LINE = disable
> PROXY1_MAIL_ACCESS = 864 # Put Your Voice Mail Sytem's 
> Pilot Number here
>
> sip_2000.cfg:
>
> LINE1 = 2000
> LINE1_PROXY   = 1
> LINE1_CALLID  = 2000
> #LINE1_AUTH= 2000; 2000
>
> sip.conf:
>
> ; Polycom Spectralink 8002
> [2000]
>type=friend
>host=192.168.3.123
>;port=5060
>secret=2000
>username=2000
>;fromuser=2000
>;authuser=2000
>qualify=no   ; turned this off to stop asterisk side initiated traffic
>context=spectra_default
>dtmfmode=rfc2833
>disallow=all
>allow=ulaw
>mailbox...@default
>canreinvite=yes
>callgroup=1
>pickupgroup=1
>accountcode=Home
>nat=no
>
>
> Syslog:
>
> Feb 23 20:25:06 192.168.3.123 Jan  1 00:18:24.57 0090.7a0a.13f3 
> (192.168.003.123) [0007] Call start, AP 0014.d1c2.70fe (-32 dBm)
> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.87 0090.7a0a.13f3 
> (192.168.003.123) [0008] Number Abufs: 26
> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.87 0090.7a0a.13f3 
> (192.168.003.123) [0009] Number Fbufs: 2
> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.88 0090.7a0a.13f3 
> (192.168.003.123) [000a] Max Number Abufs: 359
> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.88 0090.7a0a.13f3 
> (192.168.003.123) [000b] Max Number Fbufs: 33
> Feb 23 20:25:11 192.168.3.123 Jan  1 00:18:29.57 0090.7a0a.13f3 
> (192.168.003.123) [000c] NStat: 0014.d1c2.70fe (-30 dBm), Tx 3704, Rx 
> 43841, BTx 2, BRx 2766, MTx 0, MRx 0, Tx Drop 3 (0.1%), Tx Retry 96 
> (2.7%), Rx Retry 19 (0.0%)
> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:33.87 0090.7a0a.13f3 
> (192.168.003.123) [000d] Number Abufs: 46
> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:33.87 0090.7a0a.13f3 
> (192.168.003.123) [000e] Number Fbufs: 3
> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:34.57 0090.7a0a.13f3 
> (192.168.003.123) [000f] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3707, Rx 
> 43996, BTx 2, BRx 2773, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96 
> (0.0%), Rx Retry 19 (0.0%)
> Feb 23 20:25:21 192.168.3.123 Jan  1 00:18:39.57 0090.7a0a.13f3 
> (192.168.003.123) [0010] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3708, Rx 
> 44284, BTx 2, BRx 2792, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96 
> (0.0%), Rx Retry 19 (0.0%)
> Feb 23 20:25:26 192.168.3.123 Jan  1 00:18:44.36 0090.7a0a.13f3 
> (192.168.003.123) [0011] Call end, AP 0014.d1c2.70fe (-36 dBm)
>
>
>
>
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Re: [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5

[Leif Madsen]

This doesn't really answer the initial question you asked, but 1.6.0.6 was 
released yesterday, and it contains some fixes specifically for the Asterisk 
GUI.

Check the release announcement for the two issues that were fixed. I don't 
believe they affect getting the GUI running, but rather tend to creep up over a 
period of time. At least as is my understanding.

Thanks!
Leif.

Tamer Higazi wrote:
> Hi people!
> I am not getting really smart. I get the SVN Edition of asterisk GUI
> interface, compiled and love to get it to run, what won't work. What am
> I doing wrong?!


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Re: [asterisk-users] multiple asterisks in a server

[Danny Nicholas]

It is possible but not easy.  Virtualization isn't necessarily the answer
because of sharing the physical device(s) - If you're a SIP-only
environment, then that wouldn't be a problem, but most * installs (IMO) use
some flavor of Zap/DAHDI which has to be addressed/locked.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Tuesday, February 24, 2009 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] multiple asterisks in a server

On Tue, Feb 24, 2009 at 2:59 AM, Rilawich Ango 
wrote:
> Hi all,
>  Is it possible to install more than 1 asterisk in a single server?

Can somebody help me understand why you would want to do this?

I suppose development versus production, but wouldn't you also want
better separation, like virtualization?

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Re: [asterisk-users] multiple asterisks in a server

[David Backeberg]

On Tue, Feb 24, 2009 at 2:59 AM, Rilawich Ango  wrote:
> Hi all,
>  Is it possible to install more than 1 asterisk in a single server?

Can somebody help me understand why you would want to do this?

I suppose development versus production, but wouldn't you also want
better separation, like virtualization?

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Re: [asterisk-users] astdb and Debian : can't use db4.5_dump

[Olivier]

2009/2/24 Philipp Kempgen 

> Olivier schrieb:
> > 2009/2/24 Steve Howes 
> >> On 24 Feb 2009, at 12:37, Olivier wrote:
>
> >> > On Lenny, I typed "apt-get install db4.5-util " then (as root) :
> >> >
> >> > # db4.5_dump /var/lib/asterisk/astdb
> >> > db4.5_dump: /var/lib/asterisk/astdb: unexpected file type or format
> >> > db4.5_dump: open: /var/lib/asterisk/astdb: Invalid argument
> >> >
> >> > # file /var/lib/asterisk/astdb
> >> > /var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3,
> >> > native byte-order)
> >> >
> >> > Is db4.5_dump appropriate to dump an Asterisk database ?
> >>
> >> Apparently not... its a Berkeley DB version 1
>
> > So the concensus is db4.5_dump is not appropriate.
> >
> > Which is the appropriate tool ?
> > In another thread, I mentioned I couldn't find any db_dump185.c file in
> my
> > system though /usr/src/bristuff../asterisk/main/db1-ast/Makefile
> explicitely
> > mentions a db_dump185 target coming from this db_dump185.c.
>
> Somewhat related to this thread:
> http://lists.digium.com/pipermail/asterisk-users/2009-January/225950.html


Yes, that's a follow up on this thread !
As I discovered  db4.5-util , I thought it could be worth trying ...

It seems to me, now,  it's not the case  I should try
http://search.cpan.org/dist/DB_File/DB_File.pm

which I wanted to avoid as I'm not used to play with Perl (nobody's perfect
anyway ;)))


>
>
>Philipp Kempgen
> --
> AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>
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Re: [asterisk-users] GSM codec is a good choice ???

[Alejandro Cabrera Obed]

Yes, there is a WAN among the hardphones and my Asterisk server.

I know the GSM bitrate is about 31 Kbps.

Thanks

On Tue, Feb 24, 2009 at 1:41 PM, Olivier  wrote:
>
>
> 2009/2/24 Alejandro Cabrera Obed 
>>
>> Thanks for your comment about codecsI tell you I can't use G.711
>> because I use a WAN link, and this is a wide band codec.
>>
>> Is GSM codec totally free (avoid to pay for any license) ???
>
> yes !
> Is there a WAN between your hardphones and Asterisk ?
>>
>>
>> Thnks again.
>>
>> On Tue, Feb 24, 2009 at 11:50 AM, Tiago Durante 
>> wrote:
>> > I'd use alaw/ulaw for everything that's local, gsm or g729 only for
>> > remote extensions.
>> >
>> > On 2/24/09, Philipp Kempgen  wrote:
>> >> Alejandro Cabrera Obed schrieb:
>> >>> Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
>> >>> with GSM sound files.
>> >>>
>> >>> The problem is I have IP phones Utopix HyperPhone 202 which support
>> >>> only G.729a/u and G.723.1 high/low, but not GSM.
>> >>
>> >> http://www.utopixnetworks.com.ar/ip_phones_hiperphone_202.php
>> >> According to the web site the Utopix HiperPhone 202 and 112
>> >> support G.711a/u (alaw/ulaw) as well.
>> >> So why not use G.711a for everything?
>> >>
>> >>
>> >> Philipp Kempgen
>> >> --
>> >> AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
>> >> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
>> >> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
>> >> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
>> >> --
>> >>
>> >> ___
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>> >> To UNSUBSCRIBE or update options visit:
>> >>http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >
>> > --
>> > Sent from my mobile device
>> >
>> > Tiago Durante
>> >
>> > ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,
>> > Perseverance is the hard work you do after you
>> > get tired of doing the hard work you already did.
>> > -- Newt Gingrich
>> >
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>> >
>>
>>
>>
>> --
>> Alejandro Cabrera Obed
>> aco1...@gmail.com
>> www.alejandrocabrera.com.ar
>>
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-- 
Alejandro Cabrera Obed
aco1...@gmail.com
www.alejandrocabrera.com.ar

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Re: [asterisk-users] GSM codec is a good choice ???

[Eric Wieling, Asteria Solutions Group]

Tiago Durante wrote:
> On Tue, Feb 24, 2009 at 10:41 AM, Olivier  wrote:
>>
>> 2009/2/24 Alejandro Cabrera Obed 
>>> Thanks for your comment about codecsI tell you I can't use G.711
>>> because I use a WAN link, and this is a wide band codec.
>>>
>>> Is GSM codec totally free (avoid to pay for any license) ???
>> yes !
>> Is there a WAN between your hardphones and Asterisk ?
> 
> GSM is a great and free codec. However if your phones doesn't have it,
> you'll have to buy the g729 licenses...

How much would new phones cost?  If they cost more than $10 each then 
you will spend more money by buying new phones rather than buying a G729 
license.  Remember G729 is licensed on a per simultaneous channel basis. 
  i.e. if you buy 10 licenses you can have up to 10 G729 channels in use 
at any one time.  You do not need one license per phone.

-- 
Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com

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Re: [asterisk-users] GSM codec is a good choice ???

[Tiago Durante]

On Tue, Feb 24, 2009 at 10:41 AM, Olivier  wrote:
>
>
> 2009/2/24 Alejandro Cabrera Obed 
>>
>> Thanks for your comment about codecsI tell you I can't use G.711
>> because I use a WAN link, and this is a wide band codec.
>>
>> Is GSM codec totally free (avoid to pay for any license) ???
>
> yes !
> Is there a WAN between your hardphones and Asterisk ?

GSM is a great and free codec. However if your phones doesn't have it,
you'll have to buy the g729 licenses...

regards,


-- 
Tiago Durante

,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,
Perseverance is the hard work you do after you
get tired of doing the hard work you already did.
-- Newt Gingrich

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Re: [asterisk-users] GSM codec is a good choice ???

[Olivier]

2009/2/24 Alejandro Cabrera Obed 

> Thanks for your comment about codecsI tell you I can't use G.711
> because I use a WAN link, and this is a wide band codec.
>
> Is GSM codec totally free (avoid to pay for any license) ???

yes !
Is there a WAN between your hardphones and Asterisk ?

>
>
> Thnks again.
>
> On Tue, Feb 24, 2009 at 11:50 AM, Tiago Durante 
> wrote:
> > I'd use alaw/ulaw for everything that's local, gsm or g729 only for
> > remote extensions.
> >
> > On 2/24/09, Philipp Kempgen  wrote:
> >> Alejandro Cabrera Obed schrieb:
> >>> Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
> >>> with GSM sound files.
> >>>
> >>> The problem is I have IP phones Utopix HyperPhone 202 which support
> >>> only G.729a/u and G.723.1 high/low, but not GSM.
> >>
> >> http://www.utopixnetworks.com.ar/ip_phones_hiperphone_202.php
> >> According to the web site the Utopix HiperPhone 202 and 112
> >> support G.711a/u (alaw/ulaw) as well.
> >> So why not use G.711a for everything?
> >>
> >>
> >> Philipp Kempgen
> >> --
> >> AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
> >> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> >> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> >> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> >> --
> >>
> >> ___
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> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> > --
> > Sent from my mobile device
> >
> > Tiago Durante
> >
> > ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,
> > Perseverance is the hard work you do after you
> > get tired of doing the hard work you already did.
> > -- Newt Gingrich
> >
> > ___
> > -- 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
> >
>
>
>
> --
> Alejandro Cabrera Obed
> aco1...@gmail.com
> www.alejandrocabrera.com.ar
>
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Re: [asterisk-users] astdb and Debian : can't use db4.5_dump

[Philipp Kempgen]

Olivier schrieb:
> 2009/2/24 Steve Howes 
>> On 24 Feb 2009, at 12:37, Olivier wrote:

>> > On Lenny, I typed "apt-get install db4.5-util " then (as root) :
>> >
>> > # db4.5_dump /var/lib/asterisk/astdb
>> > db4.5_dump: /var/lib/asterisk/astdb: unexpected file type or format
>> > db4.5_dump: open: /var/lib/asterisk/astdb: Invalid argument
>> >
>> > # file /var/lib/asterisk/astdb
>> > /var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3,
>> > native byte-order)
>> >
>> > Is db4.5_dump appropriate to dump an Asterisk database ?
>>
>> Apparently not... its a Berkeley DB version 1

> So the concensus is db4.5_dump is not appropriate.
> 
> Which is the appropriate tool ?
> In another thread, I mentioned I couldn't find any db_dump185.c file in my
> system though /usr/src/bristuff../asterisk/main/db1-ast/Makefile explicitely
> mentions a db_dump185 target coming from this db_dump185.c.

Somewhat related to this thread:
http://lists.digium.com/pipermail/asterisk-users/2009-January/225950.html


Philipp Kempgen
-- 
AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] building asterisk-1.6.0.6 failed!

[M Hulber]

I'm having a different problem building this release:

[CC] app_dahdiras.c -> app_dahdiras.o
In file included from app_dahdiras.c:50:
/usr/include/dahdi/user.h:736: error: expected specifier-qualifier-list 
before ‘__s32’
/usr/include/dahdi/user.h:939: error: expected specifier-qualifier-list 
before ‘__s32’
make[1]: *** [app_dahdiras.o] Error 1
make: *** [apps] Error 2


Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 
EST 2009 x86_64 x86_64 x86_64 GNU/Linux


Tzafrir Cohen wrote:
> On Tue, Feb 24, 2009 at 09:31:21AM +0100, Tamer Higazi wrote:
>   
>> I did the same thing, without the prefix stuff!
>>
>> The same error!
>>
>>[CC] extconf.c -> extconf.o
>> In file included from /usr/local/include/datatypes.h:50,
>>  from /usr/local/include/err.h:49,
>>  from extconf.c:45:
>> /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file
>> or directory
>> In file included from /usr/local/include/datatypes.h:50,
>>  from /usr/local/include/err.h:49,
>>  from extconf.c:45:
>> /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t'
>> /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here
>> make[1]: *** [extconf.o] Error 1
>> make: *** [utils] Error 2
>> ta...@tux /tmp/asterisk-1.6.0.6 $
>> 
>
> What system is it, exactly?
>
> If Linux: what distribution? What version?
> If not: what OS? What version?
>
> In what package (or whatever) is /usr/local/include/err.h included?
>
>   

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Re: [asterisk-users] astdb and Debian : can't use db4.5_dump

[Olivier]

2009/2/24 Steve Howes 

>
> On 24 Feb 2009, at 12:37, Olivier wrote:
>
> > Hi,
> >
> > On Lenny, I typed "apt-get install db4.5-util " then (as root) :
> >
> > # db4.5_dump /var/lib/asterisk/astdb
> > db4.5_dump: /var/lib/asterisk/astdb: unexpected file type or format
> > db4.5_dump: open: /var/lib/asterisk/astdb: Invalid argument
> >
> > # file /var/lib/asterisk/astdb
> > /var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3,
> > native byte-order)
> >
> > Is db4.5_dump appropriate to dump an Asterisk database ?
>
> Apparently not... its a Berkeley DB version 1
>
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>

So the concensus is db4.5_dump is not appropriate.

Which is the appropriate tool ?
In another thread, I mentioned I couldn't find any db_dump185.c file in my
system though /usr/src/bristuff../asterisk/main/db1-ast/Makefile explicitely
mentions a db_dump185 target coming from this db_dump185.c.
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Re: [asterisk-users] GSM codec is a good choice ???

[Alejandro Cabrera Obed]

Thanks for your comment about codecsI tell you I can't use G.711
because I use a WAN link, and this is a wide band codec.

Is GSM codec totally free (avoid to pay for any license) ???

Thnks again.

On Tue, Feb 24, 2009 at 11:50 AM, Tiago Durante  wrote:
> I'd use alaw/ulaw for everything that's local, gsm or g729 only for
> remote extensions.
>
> On 2/24/09, Philipp Kempgen  wrote:
>> Alejandro Cabrera Obed schrieb:
>>> Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
>>> with GSM sound files.
>>>
>>> The problem is I have IP phones Utopix HyperPhone 202 which support
>>> only G.729a/u and G.723.1 high/low, but not GSM.
>>
>> http://www.utopixnetworks.com.ar/ip_phones_hiperphone_202.php
>> According to the web site the Utopix HiperPhone 202 and 112
>> support G.711a/u (alaw/ulaw) as well.
>> So why not use G.711a for everything?
>>
>>
>> Philipp Kempgen
>> --
>> AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
>> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
>> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
>> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
>> --
>>
>> ___
>> -- 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
>>
>
> --
> Sent from my mobile device
>
> Tiago Durante
>
> ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,
> Perseverance is the hard work you do after you
> get tired of doing the hard work you already did.
> -- Newt Gingrich
>
> ___
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Alejandro Cabrera Obed
aco1...@gmail.com
www.alejandrocabrera.com.ar

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Re: [asterisk-users] astdb and Debian : can't use db4.5_dump

[Steve Howes]

On 24 Feb 2009, at 12:37, Olivier wrote:

> Hi,
>
> On Lenny, I typed "apt-get install db4.5-util " then (as root) :
>
> # db4.5_dump /var/lib/asterisk/astdb
> db4.5_dump: /var/lib/asterisk/astdb: unexpected file type or format
> db4.5_dump: open: /var/lib/asterisk/astdb: Invalid argument
>
> # file /var/lib/asterisk/astdb
> /var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3,  
> native byte-order)
>
> Is db4.5_dump appropriate to dump an Asterisk database ?

Apparently not... its a Berkeley DB version 1

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Re: [asterisk-users] API hangup command

[Danny Nicholas]

You can attach the two ends of the call to a conference room and record
that.  Call manager would be good for this.  Here's the scenario:
Caller A dials Caller B
Caller A is put into conference XXX
Caller B is called and put into conference XXX
Call actually ends when both callers have hung up (so there would be a risk
if Caller A ended the call for some reason, leaving Caller B in the
conference room).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, February 24, 2009 7:49 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] API hangup command

If a call is established to a destination device Phone to other 
device (phone or something).
Then I issue the "monitor" command to record the person speaking.
Now I want to STOP the end device call --- BUT  I want to continue 
to record the
person speaking and sometime later deliver the entire message - is this 
possible

Looking at the hangup command its going to shutdown or stop my monitor 
command.
I pretty certain about that...

Is there a way to "redirect" the phone to other device to perhaps a 
local channel or something
like that so I can continue to record the call for use later.

Any ideas?

Jerry

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[asterisk-users] COSTA RICA - E1

[Luis Morales]

Does any have experience with  E1 telephony support plus asterisk in
costa rica ?


Regards,

Luis Morales

--
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-



-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
"Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible"

Leonardo Da'Vinci
-

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Re: [asterisk-users] GSM codec is a good choice ???

[Tiago Durante]

I'd use alaw/ulaw for everything that's local, gsm or g729 only for
remote extensions.

On 2/24/09, Philipp Kempgen  wrote:
> Alejandro Cabrera Obed schrieb:
>> Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
>> with GSM sound files.
>>
>> The problem is I have IP phones Utopix HyperPhone 202 which support
>> only G.729a/u and G.723.1 high/low, but not GSM.
>
> http://www.utopixnetworks.com.ar/ip_phones_hiperphone_202.php
> According to the web site the Utopix HiperPhone 202 and 112
> support G.711a/u (alaw/ulaw) as well.
> So why not use G.711a for everything?
>
>
> Philipp Kempgen
> --
> AMOOCON 2009, May 4-5, Rostock / Germany   ->  http://www.amoocon.de
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>
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>

-- 
Sent from my mobile device

Tiago Durante

,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,
Perseverance is the hard work you do after you
get tired of doing the hard work you already did.
-- Newt Gingrich

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[asterisk-users] API hangup command

[Jerry Geis]

If a call is established to a destination device Phone to other 
device (phone or something).
Then I issue the "monitor" command to record the person speaking.
Now I want to STOP the end device call --- BUT  I want to continue 
to record the
person speaking and sometime later deliver the entire message - is this 
possible

Looking at the hangup command its going to shutdown or stop my monitor 
command.
I pretty certain about that...

Is there a way to "redirect" the phone to other device to perhaps a 
local channel or something
like that so I can continue to record the call for use later.

Any ideas?

Jerry

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[asterisk-users] db_dump185.c missing if Asterisk 1.4 source file

[Olivier]

Hi,

In http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.22.tar.gz,
you can read db_dump185 among main/astdb/Makefile targets, though this
target is commented out from all target :
...
LIBDBSO=libdb.so.$(SOVER)
PROG=db_dump185
...
SHOBJS=$(patsubst %.o,%.os,$(OBJS))

include $(ASTTOPDIR)/Makefile.rules

all: $(LIBDB) #$(LIBDBSO) $(PROG)

db_dump185.o: db_dump185.c
$(CL) -o $@ $<


If you run "make ASTTOPDIR=/usr/src/bristuff .../asterisk db_dump185", it
replies db_dump185.c is missing.

Is using this makefile the way to build db_dump185 ?
If negative, which is the way to get a running db_dump185 ?
If positive, where should db_dump185.c come from ?

Regards
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Re: [asterisk-users] astdb and Debian : can't use db4.5_dump

[Tzafrir Cohen]

On Tue, Feb 24, 2009 at 01:37:07PM +0100, Olivier wrote:
> Hi,
> 
> On Lenny, I typed "apt-get install db4.5-util " then (as root) :
> 
> # db4.5_dump /var/lib/asterisk/astdb
> db4.5_dump: /var/lib/asterisk/astdb: unexpected file type or format
> db4.5_dump: open: /var/lib/asterisk/astdb: Invalid argument
> 
> # file /var/lib/asterisk/astdb
> /var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3, native
> byte-order)
> 
> Is db4.5_dump appropriate to dump an Asterisk database ?

Right. It is a different format. 

-- 
   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] HDD FULLL

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

David @ULC wrote:
> When I am trying to delete voice logs, 
> 
> [r...@vicidialnow monitor]# rm * -r -f
> -bash: /bin/rm: Argument list too long
> [r...@vicidialnow monitor]#
> 
> Argument list too long is coming as a road block.
> 
> Now way to forcefully delete files ?
> 


Use:
cd /path/to/monitor
find . -type f | xargs rm


-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

iD8DBQFJo+4sCFu3bIiwtTARAmtlAJ9ZSHjMUTFogxjV1+R3SVai46PxtQCgifkJ
m18j5pNazt3YBytO3rUV/NU=
=djs8
-END PGP SIGNATURE-

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Re: [asterisk-users] chan_sip and database integration

[Johansson Olle E]

24 feb 2009 kl. 11.31 skrev Michiel van Baak:

> On 10:56, Tue 24 Feb 09, Klaus Darilion wrote:
>> Hi!
>>
>> I tried to understand how chan_sip can be configured by means of a
>> database. I found these 2 different approaches (please correct me  
>> if I
>> am wrong):
>>
>> static configuration: the sip.conf file is mapped to a database  
>> table.
>> The table contains one line for each line in sip.conf.
>>
>> realtime configuration: the peers/users are stored in the database  
>> using
>> a single line for each peer/user.
>>
>>
>> "Static" does not eases provisioning as configuring a SIP peer/user
>> using this approach is really complicated - it is just a method to  
>> store
>> .conf files in database.
>>
>> "realtime" really eases provisioning of SIP peers/users. You only  
>> have
>> to insert/update/delete a single line. But functionality is  
>> different -
>> there are limitations as these objects are not stored in memory  
>> (can be
>> cached), for example device status information.
>>
>>
>> What I am looking for is a method to provision peers/users with a  
>> single
>> line in the database, but without limitations. Thus, the peers need  
>> not
>> to be realtime but are loaded on "sip reload".
>>
>> So is there a possiblity to have static peer/users configuration  
>> using a
>> nice and easy way?
>
> Store them in a database and use a combination of cron and some
> scripting to generate the configuration files.
>
> Some advice: keep track if an update has been done to the database  
> since
> last reload and only regen files and issue a reload when this is true.

I think this is what FreePBX does.

/O

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Re: [asterisk-users] multiple asterisks in a server

[Olivier]

2009/2/24 Geraint Lee 

> Almost forgot, you need to make sure you bind each instance to either it's
> own IP address or different ports on the same ip, i used 2 IP's for it and
> never hda a problem.
>
> 2009/2/24 Geraint Lee 
>
>> Yes it's possible..
>>
>> When you install use...
>> ./configure --prefix=/usr/local/asterisk2 or something like it.
>>
>> I had to change astrundir (in asterisk.conf) as well.
>>
>> One thing to watch out for is that if you run make samples it will
>> overwrite the ones stored in /etc/asterisk and not where you'd expect them
>> to be in /usr/local/asterisk2/etc/asterisk (or at least it di dwhen i did
>> it!).
>>
>> and for a helping hand i symlinked /usr/local/asterisk2/sbin/asterisk to
>> /usr/local/sbin/asterisk2 and /usr/local/asterisk2/sbin/safe_asterisk to
>> /usr/local/sbin/safe_asterisk2
>>
>> Cheers
>>
>> Geraint
>>
>> You will also need to look at asterisk.conf in the new installation
>> directory and as a quickfix to get it running, use a different location for
>> astrundir
>>
>> 2009/2/24 Rilawich Ango 
>> - Show quoted text -
>>
>> Hi all,
>>>  Is it possible to install more than 1 asterisk in a single server?
>>> If yes, what do I need to set and take care?
>>>
>>> Rgds,
>>> ango
>>>
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>>
>>
>
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Alternatively, VServer or OpenVZ might help ...
(Beware to DHCP is your Asterisk also have to act as DHCP servers)
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[asterisk-users] astdb and Debian : can't use db4.5_dump

[Olivier]

Hi,

On Lenny, I typed "apt-get install db4.5-util " then (as root) :

# db4.5_dump /var/lib/asterisk/astdb
db4.5_dump: /var/lib/asterisk/astdb: unexpected file type or format
db4.5_dump: open: /var/lib/asterisk/astdb: Invalid argument

# file /var/lib/asterisk/astdb
/var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3, native
byte-order)

Is db4.5_dump appropriate to dump an Asterisk database ?

Regards
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Re: [asterisk-users] asterisk -f and restart now

[Klaus Darilion]

Tzafrir Cohen wrote:
> On Tue, Feb 24, 2009 at 10:55:40AM +0100, Klaus Darilion wrote:
>> Hi!
>>
>> If I start Asterisk with -f (do not force) then on "restart now" the
>> Asterisk process still has the same PID. 
> 
> On 'restart' Asterisk basically re-execs itself. Both with and without -f .

But without -f the new Asterisk gets a new PID, and with -f the PID is 
the same. Thus I thought there should be a diference.

klaus

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Re: [asterisk-users] Managing the spiralling costs

[BJ Weschke]

Vikas wrote:
> I have been using the inbound 800 services from vitelity. Slowly the
> usage has been rising and in the month of Jan the bill was for $650. I
> am currently on a 1.9 cents a minute plan. Am I paying too much ?
>
> Some suggestions my team generated to reduce the toll free incoming
> call bill were:
>
> 1. When people call in on the 800 number take the local number they
> are calling from and then call them back from our unlimited outgoing
> account from broadvoice.
>
> 2. Find a vendor with a better rate.
>
> Any idea what we can do to better manage the 800 cost.
>
> Thanks for your time,
>
> Vikas
>
>   
 As far as a better rate, that really depends where your callers are coming 
from. If they're calling in primarily from off-net areas, you'll find that 
$0.019/min as a blended rate is actually fairly competitive and your provider 
is probably losing money on your business! 

 If they're calling in from on-net areas, you may find another provider willing 
to give you a better per min rate, but my experience has been that the monthly 
revenue commitment usually starts in around the $5k/month range before the rate 
comes down below what you're already paying. 

 All that being said, before you go seeking a better rate, make sure you've 
done a good amount of due dilligence on a different provider to make sure that 
you're still going to receive the best level of service even though you may 
have a better rate.

--
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Re: [asterisk-users] Swichted digits on received number from fax on an fxs port

[Tzafrir Cohen]

On Tue, Feb 24, 2009 at 10:15:27AM +0100, Loic Didelot wrote:
> Hello,
> I have connected a fax machine to a xorcom fxs port (signalling=fxo_ks).
> But when I try to dial (send a fax) the number that I receive in
> asterisk is wrong. Quite often a few digits are wrong but sometime is
> correct. It looks like it works 2 times out of 10.
> 
> Examples: 
> 2090 becomes 2999 or 2000
> 1234567890 becomes 1234566790

On fxo_ks the phone does not get to set the caller ID. Where do you set
the caller ID? Where do you read it?

-- 
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Re: [asterisk-users] asterisk -f and restart now

[Tzafrir Cohen]

On Tue, Feb 24, 2009 at 10:55:40AM +0100, Klaus Darilion wrote:
> Hi!
> 
> If I start Asterisk with -f (do not force) then on "restart now" the
> Asterisk process still has the same PID. 

On 'restart' Asterisk basically re-execs itself. Both with and without -f .

-- 
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+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] HDD FULLL

[Tzafrir Cohen]

On Tue, Feb 24, 2009 at 02:40:35AM +, Jeff LaCoursiere wrote:
> 
> On Mon, 23 Feb 2009, David fire wrote:
> 
> [snip]
> 
> >
> > cd //
> > rm * -r -f
> >
> > PLEASE DONT DO THIS AT THE ROOT DIR OR YOU WILL ERASE ALL THE DISK.
> > CD TO THE TARGET DIRECTORY OR YOU WILL DESTROY YOUR SERVER.
> >
> 
> Nah, you will only get as far as some shared libraries before the system 
> crashes :)

Reminder: what rm does is actually unlink(2). That is: it doesn't remove
a file from the disk. It merely removes one link from the file system to
it. A file is removed only when its reference count is down to 0.

When a program uses a library, the copy of that library is still kept on
the disk. Even if it is unlinked from the filesystem already.

$ echo whatever >file
$ sleep 1000  
> Odd syntax you have there.  Most would destroy things with rm -rf * .

The '.' in the end adds some extra power. ;-)


So the bad news are that there is nothing to stop an rm -rf //#. Not 
even when system libraries are deleted.

-- 
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http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] building asterisk-1.6.0.6 failed!

[Tzafrir Cohen]

On Tue, Feb 24, 2009 at 09:31:21AM +0100, Tamer Higazi wrote:
> I did the same thing, without the prefix stuff!
> 
> The same error!
> 
>[CC] extconf.c -> extconf.o
> In file included from /usr/local/include/datatypes.h:50,
>  from /usr/local/include/err.h:49,
>  from extconf.c:45:
> /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file
> or directory
> In file included from /usr/local/include/datatypes.h:50,
>  from /usr/local/include/err.h:49,
>  from extconf.c:45:
> /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t'
> /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here
> make[1]: *** [extconf.o] Error 1
> make: *** [utils] Error 2
> ta...@tux /tmp/asterisk-1.6.0.6 $

What system is it, exactly?

If Linux: what distribution? What version?
If not: what OS? What version?

In what package (or whatever) is /usr/local/include/err.h included?

-- 
   Tzafrir Cohen
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Re: [asterisk-users] chan_sip and database integration

[Michiel van Baak]

On 10:56, Tue 24 Feb 09, Klaus Darilion wrote:
> Hi!
> 
> I tried to understand how chan_sip can be configured by means of a
> database. I found these 2 different approaches (please correct me if I
> am wrong):
> 
> static configuration: the sip.conf file is mapped to a database table.
> The table contains one line for each line in sip.conf.
> 
> realtime configuration: the peers/users are stored in the database using
> a single line for each peer/user.
> 
> 
> "Static" does not eases provisioning as configuring a SIP peer/user
> using this approach is really complicated - it is just a method to store
> .conf files in database.
> 
> "realtime" really eases provisioning of SIP peers/users. You only have
> to insert/update/delete a single line. But functionality is different -
> there are limitations as these objects are not stored in memory (can be
> cached), for example device status information.
> 
> 
> What I am looking for is a method to provision peers/users with a single
> line in the database, but without limitations. Thus, the peers need not
> to be realtime but are loaded on "sip reload".
> 
> So is there a possiblity to have static peer/users configuration using a
> nice and easy way?

Store them in a database and use a combination of cron and some
scripting to generate the configuration files.

Some advice: keep track if an update has been done to the database since
last reload and only regen files and issue a reload when this is true.

> 
> thanks
> klaus
> 
> 
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-- 

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"Why is it drug addicts and computer aficionados are both called users?"


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[asterisk-users] chan_sip and database integration

[Klaus Darilion]

Hi!

I tried to understand how chan_sip can be configured by means of a
database. I found these 2 different approaches (please correct me if I
am wrong):

static configuration: the sip.conf file is mapped to a database table.
The table contains one line for each line in sip.conf.

realtime configuration: the peers/users are stored in the database using
a single line for each peer/user.


"Static" does not eases provisioning as configuring a SIP peer/user
using this approach is really complicated - it is just a method to store
.conf files in database.

"realtime" really eases provisioning of SIP peers/users. You only have
to insert/update/delete a single line. But functionality is different -
there are limitations as these objects are not stored in memory (can be
cached), for example device status information.


What I am looking for is a method to provision peers/users with a single
line in the database, but without limitations. Thus, the peers need not
to be realtime but are loaded on "sip reload".

So is there a possiblity to have static peer/users configuration using a
nice and easy way?

thanks
klaus


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[asterisk-users] asterisk -f and restart now

[Klaus Darilion]

Hi!

If I start Asterisk with -f (do not force) then on "restart now" the
Asterisk process still has the same PID. Thus, what happens really in -f
mode? Is the "restart now" just a reload of all modules?

thanks
klaus


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