Re: [asterisk-users] setvar for outgoing SIP channels?

[Klaus Darilion]

I answer myself: since Asterisk 1.6 you can use the SIPPEER function to 
retrieve the peer's setvar variables.

regards
klaus

Klaus Darilion schrieb:
 Hi!
 
 Using setvar in a peer configuration (sip.conf) I can set the channel 
 variables for the incoming channel. Is there a similar method which 
 allows me to load these variables also for outgoing channels (e.g. to 
 load callee preferences)?
 
 thanks
 klaus
 
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[asterisk-users] asterisk console: quit is twice in history

[Klaus Darilion]

Hi!

When I enter the Asterisk console and press the up key I get the 
command history. But the quit is twice in the history. Why?

thanks
klaus

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Re: [asterisk-users] SIP request send me 482 error

[remi . druilhe]

Hi,

I tried to apply both patchs, but there isn't improvements.

Finally, I have implemented the idea of Stefan. I use another B2B
equipment which create a new session after the first passage in
Asterisk and before the second passage. It's not very clean but it
works ...

Thanks for help :)

Regards

--
Rmi Druilhe

Stefan Gofferje a crit:

  Hi,

[EMAIL PROTECTED] schrieb:
  
  
In fact, after entering in Asterisk for the first time, my call is
redirected to an other component of my system. This other equiment
redirect the same call to Asterisk a second time.

  
  
Hm, I suppose, your "equipment" is using reinvites for that redirection.
The only idea to solve this I can think of would be having your
"equipment" stay in the media path, i.e. making that redirection a brand
new call. Then * shouldn't complain.
But in my opinion that would be pretty ugly by means of scalability and
ressources.
Maybe a redesign of your callflow in general would be a better option.

Terve,
Stefan

  






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Re: [asterisk-users] No route to destination error

[Martin Seebach]

Hi, thanks for the reply , 

- Original Message - 
From: Philipp Kempgen [EMAIL PROTECTED] 

 Maybe something is broken in recent versions of chan_iax2.c? 
 http://lists.digium.com/pipermail/asterisk-users/2008-September/218560.html 
 Not the same issue though. 

I doubt it. It has been working fine for a while, and others report IAX2 
working fine. 

- Martin 
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[asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

[Nathan Dennis]

We have been using asterisk for a while now but have recently needed to
install a second server in a remote office and set up a iax trunk
between the 2 servers. The dial plan seems to work well when I tested it
on the same LAN. However this afternoon I connected the system at the
remote office and made some calls. All the calls connect and work fine,
voice quality is great no really couldn't have hoped for better. Hang up
the call and tried to make another call and nothing, the link was not
responding, after much trouble shooting I have found that after the call
is hung up the 2 asterisk servers seem to go into some kind of loop
sending each other message. I have pasted a debug for both servers below
that include everything from the start of the call to after hangup. I
have cut them short at the VNAK and Hangup cycle just continues for
30seconds or so flooding the link completely.
 
Any help you may be able to provide would be greatly appreciated
 
 
IAX Debug on first server
 
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
   Timestamp: 00014ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
   Timestamp: 3ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
   AUTHMETHODS : 3
   CHALLENGE   : 147710225
   USERNAME: cairns
 
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
   Timestamp: 00100ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
   MD5 RESULT  : 028e5753e0c82ce337ad4ba898a80bcf
 
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
   Timestamp: 00100ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACCEPT
   Timestamp: 00088ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
   FORMAT  : 256
 
-- Call accepted by 10.10.51.22 (format g729)
-- Format for call is g729
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
ACK
   Timestamp: 00088ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass:
ANSWER
   Timestamp: 00091ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass:
ACK
   Timestamp: 00091ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
-- IAX2/brisbane-16384 answered SIP/1406-b7b2b530
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: VOICE   Subclass:
136
   Timestamp: 00100ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass:
ACK
   Timestamp: 00100ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 004 Type: VOICE   Subclass:
136
   Timestamp: 00288ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass:
ACK
   Timestamp: 00288ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass:
(255?)
   Timestamp: 03123ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass:
ACK
   Timestamp: 03123ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
POKE
   Timestamp: 5ms  SCall: 08676  DCall: 0 [10.10.51.22:4569]
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
PONG
   Timestamp: 5ms  SCall: 7  DCall: 08676 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
ACK
   Timestamp: 5ms  SCall: 08676  DCall: 7 [10.10.51.22:4569]
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
POKE
   Timestamp: 00018ms  SCall: 1  DCall: 0 [10.10.51.22:4569]
Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
PONG
   Timestamp: 0ms  SCall: 1  DCall: 1 [10.10.51.22:4569]
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
   Timestamp: 00078ms  SCall: 1  DCall: 1 [10.10.51.22:4569]
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass:
LAGRQ
   Timestamp: 10015ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass:
LAGRP
   Timestamp: 10015ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 006 Type: IAX Subclass:
ACK
   Timestamp: 10015ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 004 Type: IAX Subclass:
LAGRQ
   Timestamp: 09855ms  SCall: 06840  DCall: 16384 [10.10.51.22:4569]
Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 007 Type: IAX Subclass:
LAGRP
   Timestamp: 09855ms  SCall: 16384  DCall: 06840 [10.10.51.22:4569]
Rx-Frame Retry[ No] -- OSeqno: 007 ISeqno: 005 Type: IAX Subclass:
ACK

Re: [asterisk-users] AGI and prepaid billing + Radius

[Philippe Sultan]

Hi Bilal,

On Tue, Sep 23, 2008 at 11:11 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
 Dear Philippe;

 Thanks a lot for ur kindly answer.

 How can I use the Radius with CDR (Accounting)?

Here is the documentation :
http://svn.digium.com/view/asterisk/branches/1.4/doc/radius.txt?view=markup


 About PortaOne's billing systems: Do u mean I can use the PortaOne's billing 
 systems Radius client (to be fixed at Asterisk side), and customize this 
 client to be used with any RADIUS based billing system?

Yep. This client is written in PERL, and uses the Authen::Radius API.
You can integrate it with Asterisk (see the doc in the link I sent),
and adapt it to make it work with any RADIUS server.

Regards,

Philippe

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Re: [asterisk-users] Asterisk mysql CDR

[Rizwan Hisham]

You can use the ResetCDR() application with the w flag in it after you get
the unavailable, busy or etc message from the callee. It will store the cdr
of that call and after forwarding to mobile, that cdr will be dumped again.

On Wed, Sep 24, 2008 at 8:26 AM, Nhadie [EMAIL PROTECTED] wrote:

 hi,

 i'm using this macro to dial an extension and forward to a mobile if
 unavailable,busy or noanswer

 exten = 100,1,Macro(dial-ext|SIP/${EXTEN}|vm-100|moh-100)
 exten = 100,2,Goto(100-${DIALSTATUS}|1)
 exten = 100-BUSY,1,Macro(dialout-local-mobile|91234567)
 exten = 100-BUSY,2,Voicemail([EMAIL PROTECTED]|u)
 exten = 100-CONGESTION,1,Macro(dialout-local-mobile|91234567)
 exten = 100-CONGESTION,2,Voicemail([EMAIL PROTECTED]|u)
 exten = 100-NOANSWER,1,Macro(dialout-local-mobile|91234567)
 exten = 100-NOANSWER,2,Voicemail([EMAIL PROTECTED]|u)

 my prob is on the CDR, from extension 500 i called 100, 100 is not
 online so it should forward it to my mobile

 but on the cdr it shows like this:

  FromTo
 500 100-CHANUNAVAIL

 should it be like

  FromTo
 500 91234567

 or

  FromTo
 100 91234567

 any idea how to fix those?

 regards,
 nhadie

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Re: [asterisk-users] Fax with asterisk

[Rizwan Hisham]

Hi all,
Sorry to interrupt. I need some help regarding fax passthru mode.

We are trying to configure fax passthru mode in asterisk using sip. For out
of network calls/fax we use trunk configuration. i am using asterisk
1.4.2.The user has to use fax machine connected to their ata and dial
the callee
number, the call is originated just like a regular voice call. have not
defined any special context for sending faxes. Have enabled t38 and
canreinvite in peer/user and trunk configuration. But the fax is not going
thru. Our service provider does support fax passthru. Following is the trunk
and user/peer configuration:

TRUNK CONF
[TRUNK-OUT]
type=peer
host=XXX
port=5060
context=default
country=us
dtmfmode=rfc2833
restrictcid=no
canreinvite=yes
insecure=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
promiscredir=yes
t38_udptl=yes

USER/PEER

[abc]
username=abc
type=friend
secret=123
qualify=25000
nat=yes
mailbox=12129339037
insecure=port,invite
incominglimit=2
outgoinglimit=2
intl_trunk=TRUNK-OUT
local_trunk=TRUNK-OUT
host=dynamic
dtmfmode=inband
context=uscan
canreinvite=yes
callerid=Rizwan Qureshi 122
accountcode=1:0:abc
amaflags=default
disallow=all
allow=ulaw
allow=alaw
allow=gsm
t38_udptl=yes


Any solutions?

On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote:

 On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro
 [EMAIL PROTECTED] wrote:
  ATAs work OK I guess, just make sure to use a loss less codec such as
 ULAW.

 Since the OP stated he is using E1 lines then he should probably be
 using alaw instead.

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Re: [asterisk-users] extension definition

[Rizwan Hisham]

You maybe using wrong username. If the user is defined in sip, you should be
able to register using the correct username and password. Also, see if
asterisk is listening on a defferent sip port instead of default 5060. If
its different use that port.

On Wed, Sep 24, 2008 at 3:32 AM, michel freiha [EMAIL PROTECTED] wrote:

 Hello Eric,
 i didwhat you asked me to do but i'm getting Notfound sip message when
 trying to register

 regrads



 On Tue, Sep 23, 2008 at 9:56 PM, Eric ManxPower Wieling [EMAIL 
 PROTECTED]wrote:

 This is done in sip.conf, iax.conf, etc, not in extensions.conf.  By the
 time a call gets to extensions.conf it must already be authenticated.

 Assume the username is robertdobbs and the ip is 209.17.71.61

 In sip.conf you would have something like this:

 [robertdobbs]
 deny=0.0.0.0/0
 permit=209.17.71.61 http://0.0.0.0/0permit=209.17.71.61
 rest of the options here



 michel freiha wrote:
  Hi all,
  I need please the exact extension definition under extensions.conf that
  accepts any call coming from an appropriate username and Ip
 address...This
  mean that the authentication should be done on username and IP address
 
  Regards
 
 
 
  
 
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 --
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 T-1, PRI, Frame Relay, Linux, and network design.  Based near
 Birmingham, AL.  Now accepting clients worldwide.

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-- 
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Rizwan Hisham
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Re: [asterisk-users] No route to destination error

[Steve Totaro]

On Wed, Sep 24, 2008 at 4:45 AM, Martin Seebach [EMAIL PROTECTED] wrote:
 Hi, thanks for the reply,

 - Original Message -
 From: Philipp Kempgen [EMAIL PROTECTED]

 Maybe something is broken in recent versions of chan_iax2.c?

 http://lists.digium.com/pipermail/asterisk-users/2008-September/218560.html
 Not the same issue though.

 I doubt it. It has been working fine for a while, and others report IAX2
 working fine.

 - Martin


Do a show codecs.  Maybe IAX2 is not loaded.  Did you build and load
ztdummy (assuming you have no Zaptel/Dahdi cards?

http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe

Thanks,
Steve Totaro

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Re: [asterisk-users] AGI and prepaid billing

[Rizwan Hisham]

We have done it too. www.axvoice.com

On Tue, Sep 23, 2008 at 3:39 PM, Benjamin Jacob [EMAIL PROTECTED]wrote:


 Hi Bilal,
 Yes it is definitely possible. And I've done it myself for a couple of our
 clients.
 Does that answer your two questions?

 cheers
 - Ben.



 --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote:

  From: bilal ghayyad [EMAIL PROTECTED]
  Subject: [asterisk-users] AGI and prepaid billing
  To: asterisk-users@lists.digium.com
  Date: Tuesday, September 23, 2008, 9:52 AM
  Hi All;
 
  Did anyone do an prepaid billing application via AGI? I
  would like to know if that is possible.
 
  Regards
  Bilal
 
 
 
 
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Re: [asterisk-users] wad happen if there is nothing wrong in conf but still can't make calls?

[Rizwan Hisham]

you must share your configuration with us. otherwise we cant even make a
wild guess.

On Mon, Sep 22, 2008 at 7:48 PM, Cindy Tan [EMAIL PROTECTED] wrote:

  may i noe wad can i do because my asterisk is working fine but the calls
 cannot proceed between 2 asterisk servers.
 hope anyone can help me solve this major problem.

 thanks a lot in advance

 Regards

 --
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Re: [asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

[Tony Mountifield]

In article [EMAIL PROTECTED],
Nathan Dennis [EMAIL PROTECTED] wrote:
 We have been using asterisk for a while now but have recently needed to
 install a second server in a remote office and set up a iax trunk
 between the 2 servers. The dial plan seems to work well when I tested it
 on the same LAN. However this afternoon I connected the system at the
 remote office and made some calls. All the calls connect and work fine,
 voice quality is great no really couldn't have hoped for better. Hang up
 the call and tried to make another call and nothing, the link was not
 responding, after much trouble shooting I have found that after the call
 is hung up the 2 asterisk servers seem to go into some kind of loop
 sending each other message. I have pasted a debug for both servers below
 that include everything from the start of the call to after hangup. I
 have cut them short at the VNAK and Hangup cycle just continues for
 30seconds or so flooding the link completely.
  
 Any help you may be able to provide would be greatly appreciated

I can't help with your problem, sorry, but anyone who can help will need
to know exactly what version of Asterisk you have at each end.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] No route to destination error

[Martin Seebach]

Hi, 
- Steve Totaro wrote: 
 Do a show codecs. 
It looks right. ulaw is loaded, and that's the only thing I allow, on both SIP 
and IAX. 

 Maybe IAX2 is not loaded. 
Looks like it: 
filserver*CLI module show like iax2 
Module Description Use Count 
chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 
1 modules loaded 

 Did you build and load ztdummy (assuming you have no Zaptel/Dahdi cards? 
No - but i don't use MeetMe? 

Thanks, 
Martin 
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[asterisk-users] Voicemail cutting out after about 30 seconds

[Chris Bagnall]

Greetings list,

I've had problems on a few of our asterisk boxes where voicemail tends to cut 
out after about 30 seconds, despite the maximum message length being set at 
240s (4m).

I've tried reducing the silence detection threshold from 128 down to 32, which 
has helped, but not resolved the issue entirely.

The calls in question are being delivered to the boxes via IAX or SIP, so it's 
not a Zaptel gain issue.

Has anyone else had similar problems? Is it safe to drop the silence 
detection threshold even lower?

Thanks in advance.

Regards,

Chris



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Re: [asterisk-users] Asterisk 1.4 or 1.6

[Artem Makhutov]

Hi,

On Tue, Sep 23, 2008 at 01:05:17PM -0600, Joseph wrote:
 I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo portage 
 but I think this version has a problem with RFC2833 DTMF signaling and I 
 don't think there 
 will be any newer version available anytime soon on portage.
 
 I need stable version, I'm using Asterisk mostly with ATA adapter (Linksys 
 and Sipura);  should I go to 1.6 or 1.4?

You can use the gentoo voip overlay. Asterisk 1.4.21.2 is included in
the overlay.

# emerge layman
# layman -a voip

Regards, Artem

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Re: [asterisk-users] chan_misdn troubles

[Thanos Koukoulis]

On Tue, Sep 23, 2008 at 1:48 PM, Thanos Koukoulis [EMAIL PROTECTED] wrote:



 On Tue, Sep 23, 2008 at 1:19 PM, Gergo Csibra [EMAIL PROTECTED] wrote:

 Tuesday, September 23, 2008, 11:57:00 AM, Thanos wrote:

  Hello

  I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine.
  I am using the OpenVox B200P ISDN card.

  My problem is that even though chan_misdn module seems to be loaded
  correctly with
  Asterisk (I can see it using 'module show' command) the misdn commands
 are
  not available
  to me in the CLI so I cannot tell if my box is correctly interfacing
 with
  the ISDN card

  Any ideas what can be going wrong ?

 ...

cd ../mISDN-1_1_7_2/

 What kernel version you use? Newer linux kernels (2.6.24) works only
 with new (and beta) 1.1.8 misdn.

 --
 Best regards,
  Gergomailto:[EMAIL PROTECTED]


 Using 2.6.18-92.1.10.el5.centos.plus kernel so I suppose that should be
 OKs.
 The modules are correctly loading

I did manage (with a little help) to get misdn to load properly
misdn show stacks returns :
BEGIN STACK_LIST:
  * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0  Debug:4
  * Port 2 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0  Debug:4


Unfortunately I can't make any calls. Whenever I try to make an external
call I get all circuits are busy msg.

Using the debug msgs the problem seems to be
 MGMT: SSTATUS: L1_DEACTIVATED

I edited misdn.conf and set pmp_l1_check=no but that does not seem to help.
I am attaching my configuration files in case someone has a better idea of
what to try :


asterisk.tar.gz
Description: GNU Zip compressed data
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Re: [asterisk-users] Voicemail cutting out after about 30 seconds

[Steven Howes]

Hi,

We saw this between Asterisk and an Audiocodes gateway. Whilst the  
voicemail is being recorded asterisk is not sending *ANY* rtp. Silence  
detection will always detect silence if it listens to this side of the  
conversation. Adjusting the threshold wont work, you need to find the  
timeout for the gateway which is doing it (not always easy...)

Steve

On 24 Sep 2008, at 12:30, Chris Bagnall wrote:

 Greetings list,

 I've had problems on a few of our asterisk boxes where voicemail  
 tends to cut out after about 30 seconds, despite the maximum message  
 length being set at 240s (4m).

 I've tried reducing the silence detection threshold from 128 down to  
 32, which has helped, but not resolved the issue entirely.

 The calls in question are being delivered to the boxes via IAX or  
 SIP, so it's not a Zaptel gain issue.

 Has anyone else had similar problems? Is it safe to drop the  
 silence detection threshold even lower?

 Thanks in advance.

 Regards,

 Chris



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Re: [asterisk-users] Asterisk 1.4 or 1.6

[Chris Bagnall]

 You can use the gentoo voip overlay. Asterisk 1.4.21.2 is included in
 the overlay.
 # emerge layman
 # layman -a voip

You may need to modify some of the .ebuild files, or your 
/etc/portage/packages.unmask depending on your asterisk build.

Regards,

Chris



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[asterisk-users] Asterisk is covering the peers IP address in SIP and SDP messages

[Arno Scholz]

Hello,

I'm implementing a VoIP client and using Asterisk 1.4. The RTP transfer 
should be handled in a direct connection from client to client. But the 
Asterisk server does not reveal the IP address of the peer in the 
contact header field of a SIP request nor in the connection header field 
of the SDP message. Instead he always writes its own address.
So the clients are forced to handle the RTP transfer over the Asterisk 
server.

Is there a possibility to configure the Asterisk server that he does not 
replace the peer IP address with his own?

I hope I could describe my problem properly.

Regards,

Arno

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Re: [asterisk-users] Debug dropped calls

[Mark Engelhardt]

Hello,

I have nearly the same issue. Does anyone have a suggestion as to how  
to find and fix this problem?

Mark

On Jul 16, 2008, at 10:59 AM, Mike (Asterisk) wrote:

 [zaptel]

 span=1,0,0,esf,b8zs

 At least one of your spans should be getting it's timing from your
 service provider.  It looks like that would be span one, this should
 read:

 span=1,1,0,esf,b8zs


 I checked my config files from before my upgrade, and I do have  
 span 1
 setup as you indicated. Silly oversight on my part. I will make the
 change and restart Asterisk/zaptel when I can. However, I experienced
 dropped calls before the upgrade, but I'll make this change and go
 from
 there.


 [zapata]

 faxdetect=incoming

 In the past, faxdetect has been known to cause problems.


 I'll change zaptel.conf first so I only change one thing at a time.

 I was finally able to change my configuration for span 1 from my
 provider to by primary clock source and span 2 to be a secondary  
 source.
 Aside from the configuration file, I'm unable to locate a way from
 within asterisk to confirm that the change took effect. 'pri show span
 1' shows:  (2 shows the same, just the D channel as 48)

 Primary D-channel: 24
 Status: Provisioned, Up, Active
 Switchtype: National ISDN
 Type: CPE
 Window Length: 0/7
 Sentrej: 0
 SolicitFbit: 0
 Retrans: 0
 Busy: 0
 Overlap Dial: 0
 T200 Timer: 1000
 T203 Timer: 1
 T305 Timer: 3
 T308 Timer: 4000
 T309 Timer: -1
 T313 Timer: 4000
 N200 Counter: 3


 Is there some text in pri intense debug that would confirm the clock
 source?

 -Mike

 -
 The information contained in this e-mail message is confidential and/ 
 or
 privileged and is intended only for the use of the individual or  
 entity
 named above. Please notify the sender immediately by email if you have
 received this email by mistake and delete this email from your system.

 If you are not the intended recipient you are hereby notified that any
 unauthorized disclosure, copying, distributing or taking any action in
 reliance of contents of this information is strictly prohibited.
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[asterisk-users] dundi and regcontext

[ronald ramos]

hi,



when a user register on my asterisk i can see it adding Noop for that 
extension, but after awhile i won't see it anymore:



what are the reasons for it being removed on the dynamic context?

one thing i found when i unregister it's removed.



dialplan show myregcontext

[ Context 'myregcontext' created by 'SIP' ]

  '100500' =   1. Noop(100500)   [SIP]

  '112802' =   1. Noop(112802)   [SIP]



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



[ Context 'pfingobizsip' created by 'SIP' ]



-= 0 extensions (0 priorities) in 1 context. =-



my prob is when it's removed dundi cant find it anymore so a user 
calling from server 1 cannot call user that is in server 2.



i've set re-registration to very low (1 minute) to monitor if my phone 
re-register and to see if it will be added again on the regcontext.

but i don't even see it unregistering after 1 minute i only 
unregistering when i am using x-lite and closing x-lite, i dont see 
x-lite re-registering if i just leave the softphone open. any idea?



regards,

ron





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Re: [asterisk-users] asterisk console: quit is twice in history

[Gordon Henderson]

On Wed, 24 Sep 2008, Klaus Darilion wrote:

 Hi!

 When I enter the Asterisk console and press the up key I get the
 command history. But the quit is twice in the history. Why?

Who knows - I suspect because it's always been like that and no-ones 
bothered to report it :)

Gordon

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Re: [asterisk-users] Voicemail cutting out after about 30 seconds

[Gordon Henderson]

On Wed, 24 Sep 2008, Chris Bagnall wrote:

 Greetings list,

 I've had problems on a few of our asterisk boxes where voicemail tends 
 to cut out after about 30 seconds, despite the maximum message length 
 being set at 240s (4m).

 I've tried reducing the silence detection threshold from 128 down to 32, 
 which has helped, but not resolved the issue entirely.

 The calls in question are being delivered to the boxes via IAX or SIP, 
 so it's not a Zaptel gain issue.

 Has anyone else had similar problems? Is it safe to drop the silence 
 detection threshold even lower?

Try:

In /etc/asterisk/asterisk.conf, under

   [options]

add

   transmit_silence_during_record = yes

That stopped a remote site I was taking VoIP calls from hanging up after 
30 seconds in a custom recording setup...

Gordon

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Re: [asterisk-users] Asterisk is covering the peers IP address in SIP and SDP messages

[Vinícius Fontes]

canreinvite=yes



Atenciosamente,

Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
 
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000

- Arno Scholz [EMAIL PROTECTED] escreveu:

 Hello,
 
 I'm implementing a VoIP client and using Asterisk 1.4. The RTP
 transfer 
 should be handled in a direct connection from client to client. But
 the 
 Asterisk server does not reveal the IP address of the peer in the 
 contact header field of a SIP request nor in the connection header
 field 
 of the SDP message. Instead he always writes its own address.
 So the clients are forced to handle the RTP transfer over the Asterisk
 
 server.
 
 Is there a possibility to configure the Asterisk server that he does
 not 
 replace the peer IP address with his own?
 
 I hope I could describe my problem properly.
 
 Regards,
 
 Arno
 
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[asterisk-users] Timeout question

[Vadim Lebedev]

I wonder which timeout will apply here: the one in master context or one from
the slave context?

[master]
exten=100,1,Dial(Local/[EMAIL PROTECTED], 20)

[slave]
exten=100,1,Dial(SIP/100, 30)



Thanks
Vadim


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Re: [asterisk-users] Fax with asterisk

[C F]

On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham [EMAIL PROTECTED] wrote:
 Hi all,
 Sorry to interrupt. I need some help regarding fax passthru mode.

 We are trying to configure fax passthru mode in asterisk using sip. For out
 of network calls/fax we use trunk configuration. i am using asterisk 1.4.2.
 The user has to use fax machine connected to their ata and dial the callee
 number, the call is originated just like a regular voice call. have not
 defined any special context for sending faxes. Have enabled t38 and
 canreinvite in peer/user and trunk configuration. But the fax is not going
 thru. Our service provider does support fax passthru. Following is the trunk
 and user/peer configuration:

They support passthru, and the originating send fax is what? PSTN? or
VoIP ATA with t38 support?
There has to one that does the t38, if the point where it gets
converted to VoIP does not support t38 then passthru will not help
you.


 TRUNK CONF
 [TRUNK-OUT]
 type=peer
 host=XXX
 port=5060
 context=default
 country=us
 dtmfmode=rfc2833
 restrictcid=no
 canreinvite=yes
 insecure=no
 disallow=all
 allow=ulaw
 allow=alaw
 allow=g729
 allow=gsm
 promiscredir=yes
 t38_udptl=yes

 USER/PEER

 [abc]
 username=abc
 type=friend
 secret=123
 qualify=25000
 nat=yes
 mailbox=12129339037
 insecure=port,invite
 incominglimit=2
 outgoinglimit=2
 intl_trunk=TRUNK-OUT
 local_trunk=TRUNK-OUT
 host=dynamic
 dtmfmode=inband
 context=uscan
 canreinvite=yes
 callerid=Rizwan Qureshi 122
 accountcode=1:0:abc
 amaflags=default
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 t38_udptl=yes


 Any solutions?

 On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen [EMAIL PROTECTED]
 wrote:

 On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro
 [EMAIL PROTECTED] wrote:
  ATAs work OK I guess, just make sure to use a loss less codec such as
  ULAW.

 Since the OP stated he is using E1 lines then he should probably be
 using alaw instead.

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 --
 Best Regards
 Rizwan Hisham


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[asterisk-users] Astricon 08 Videos interviews Voiceroute twitters on astricon

[Ming Yong]

Dear Asterisk Users,

We have uploaded a bunch of videos on Astricon 08 Day 1
Interview with Mark Spencer on big announcement  Astricon
http://www.youtube.com/watch?v=nzEeIEuQvf4

Interview with Allison Smith (See allison in a white dress with Asterisk 
shoes with asterisk on it :)
http://www.youtube.com/watch?v=fvhTKbpUP30

Interviews with various exhibitors like Digium, Xorcom, Pika, Redfone,
Counterpath at
http://www.youtube.com/voiceroute

Follow our twitters at
http://twitter.com/voiceroute

Send us email on what you would like covered for the rest of the conference.

Ming

On Wed, Sep 24, 2008 at 2:21 PM, Steve Sokol [EMAIL PROTECTED] wrote:

 AstriCon Wednesday Update

 Great to see such a good turnout!  Here's a quick update for participants
 at
 AstriCon for today.

 - If you're a Twitter user, post your updates @astricon and you'll show up
 on
 our http://community.digium.com/ site!

 - Conference changes:  We have two changes to the show schedule that didn't
 make
 it to the last minute guide - please find the deletions and replacements
 below.  Thanks to our back pocket talk speakers for agreeing to fill in
 at the
 last minute.

 DELETED:
 Wednesday 2:45 - 3:30 Cira A
Alex Kurganov: Using Asterisk in a Carrier Scale
Unified Messaging Platform

 REPLACED WITH:
   Wednesday 2:45 - 3:30 Cira A
   Jose Landivar:  Elastix: An Integrated IP Communications Platform

 DELETED:
 Thursday 2:00 - 2:45 PM Solana C/D
M. Mobeen Kahn: Start to use Speech IVR
Applications with your Asterisk System in Minutes

 REPLACED WITH:
 Thursday 2:00 - 2:45 Solana C/D
Tim Panton: The Jigsaw-Shaped Future of Mobile 2.0

 - The All-Conference Party:  Make sure to schedule your plans for the
 evening
 around the All-Conference party!  It's from 7:00 - 11:00 PM, in the Solana
 Ballroom E, which is the same room where lunch is served.

 - Errors:

 I've also been informed that we've horribly mangled the bio of  at least
 one of
 our speakers.  :-)  Our apologies to Mark Vince - please find the correct
 data
 here:

 http://www.astricon.net/2008/glendale/web/confSpeakers.php

 --
 John Todd
 [EMAIL PROTECTED]+1-256-428-6083
 Asterisk Open Source Community Director





-- 
Ming Yong
CEO, www.voiceroute.org
Druid - Open Source Unified Communications
DID: +1-877-242-3704
Office: +1-866-915-2407 ext 301
SIP/email: [EMAIL PROTECTED]
--
Attend Vikram's talk at ASTRICON 2008, 23-25 Sept 08, Glendale Arizona
http://www.astricon.net/2008/glendale/web/confTracks.php#t193

Meet us at WEB 2.0 EXPO, 17-18 Sept 08, Booth #17 in Long Tail Pavilion,
Javits Center, NYC
http://druidweb20.eventbrite.com

DruidCON 1-2 Oct Atlanta GA (The leading UC 2.0 Application platform)
http://www.voiceroute.org/druidcon

VoiceCON 08 San Francisco
10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco, CA
http://druidvoicecon.eventbrite.com

UC 2.0 Video - Mozilla Ubiquity + Druid
http://www.youtube.com/watch?v=f-5rDBPuGRc
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Re: [asterisk-users] Asterisk mysql CDR

[Nhadie]

Thank you for your reply Sir

i tried inserting ResetCDR(w) almost everywhere but i still end up with 
this on the cdr:

 FromTo:
500 100-CHANUNAVAIL

This is my current setting hat produces that CDR:

exten = 100,1,Macro(dial-ext-cf|SIP/${EXTEN}|vm-100|moh-100)
exten = 100,2,Goto(100-${DIALSTATUS}|1)
exten = 100-BUSY,1,Macro(dialout-local-mobile|91234567)
exten = 100-BUSY,2,Voicemail([EMAIL PROTECTED]|u)
exten = 100-CONGESTION,1,Macro(dialout-local-mobile|91234567)
exten = 100-CONGESTION,2,Voicemail([EMAIL PROTECTED]|u)
exten = 100-NOANSWER,1,Macro(dialout-local-mobile|91234567)
exten = 100-NOANSWER,2,Voicemail([EMAIL PROTECTED]|u)

[macro-dialout-local-mobile]
exten = s,1,Wait(1)
exten = s,n,ResetCDR(w)
exten = s,n,Dial(IAX2/trunk-100-1000/91234567|30|t)

[macro-dial-ext-cf]
exten = s,1,SetMusicOnHold(${ARG3})
exten = s,2,ResetCDR(w)
exten = s,3,Dial(SIP/100|30|t|M(setmusiconhold,100))

where do you think should i put the ResetCDR(w)?

thanks again.

regards,
nhadie

Rizwan Hisham wrote:
 You can use the ResetCDR() application with the w flag in it after you 
 get the unavailable, busy or etc message from the callee. It will store 
 the cdr of that call and after forwarding to mobile, that cdr will be 
 dumped again.
 
 On Wed, Sep 24, 2008 at 8:26 AM, Nhadie [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED] wrote:
 
 hi,
 
 i'm using this macro to dial an extension and forward to a mobile if
 unavailable,busy or noanswer
 
 exten = 100,1,Macro(dial-ext|SIP/${EXTEN}|vm-100|moh-100)
 exten = 100,2,Goto(100-${DIALSTATUS}|1)
 exten = 100-BUSY,1,Macro(dialout-local-mobile|91234567)
 exten = 100-BUSY,2,Voicemail([EMAIL PROTECTED]|u)
 exten = 100-CONGESTION,1,Macro(dialout-local-mobile|91234567)
 exten = 100-CONGESTION,2,Voicemail([EMAIL PROTECTED]|u)
 exten = 100-NOANSWER,1,Macro(dialout-local-mobile|91234567)
 exten = 100-NOANSWER,2,Voicemail([EMAIL PROTECTED]|u)
 
 my prob is on the CDR, from extension 500 i called 100, 100 is not
 online so it should forward it to my mobile
 
 but on the cdr it shows like this:
 
  FromTo
 500 100-CHANUNAVAIL
 
 should it be like
 
  FromTo
 500 91234567
 
 or
 
  FromTo
 100 91234567
 
 any idea how to fix those?
 
 regards,
 nhadie
 
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 -- 
 Best Regards
 Rizwan Hisham
 
 
 
 
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Re: [asterisk-users] Asterisk 1.4 or 1.6

[Joseph]

On 09/24/08 14:12, Chris Bagnall wrote:
 You can use the gentoo voip overlay. Asterisk 1.4.21.2 is included in
 the overlay.
 # emerge layman
 # layman -a voip

You may need to modify some of the .ebuild files, or your 
/etc/portage/packages.unmask depending on your asterisk build.

Regards,

Chris

Yes, that is what I did, I used overlay but I had a hard time to unmak it.

The 1.4 is not even in the portage unstable and it was masked in:
/usr/portage/profile/package.mask

It looks like somebody is trying very hard for Gentoo user not to use 
1.4-version.

-- 
#Joseph

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[asterisk-users] Zaptel/DAHDI ztdummy only

[Roderick A. Anderson]

Let me know if I should post this on the asterisk-dev list instead.

I am building a Linux-Vserver (http://www.linux-vserver.org) host system 
that will have several guests running Asterisk.  Since the guests can't 
load kernel modules or do other dangerous stuff, but can access them I 
built zaptel 1.4 and it is now loaded by the host.

The issue I see is there will be no Zaptel hardware and these guests 
will only do SIP, IAX, etc. but do appear to need ztdummy for timing 
with other services.  So I'm looking for a way to not load (or even 
build) all the other modules that come as part of Zaptel.

Possible?

Will the complete change to DAHDI make this easier/harder?


TIA,
Rod
-- 



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Re: [asterisk-users] No route to destination error

[Andres]

 -- Executing [EMAIL PROTECTED]:1] Set(SIP/21-081ceea8, 
 CALLERID(all)= 88821268) in new stack
 -- Executing [EMAIL PROTECTED]:2] Dial(SIP/21-081ceea8, 
 IAX2/88821268/40618405|30|r) in new stack
 [Sep 11 12:05:58] WARNING[7098]: app_dial.c:1202 dial_exec_full: 
 Unable to create channel of type 'IAX2' (cause 3 - No route to 
 destination)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing [EMAIL PROTECTED]:3] Congestion(SIP/21-081ceea8, 
 ) in new stack
   == Spawn extension (default, 40618405, 3) exited non-zero on 
 'SIP/21-081ceea8'


 I can't see any traffic on the wire using ngrep, and the registry 
 looks good:

 filserver*CLI iax2 show registry
 Host  dnsmgr  UsernamePerceived 
 Refresh  State
 85.nnn.nnn.83:4569N   8882126885.nnn.nn.197:1
 60  Registered
 85.nnn.nnn.82:4569N   8882126885.nnn.nn.197:10002
 60  Registered


 I can see traffic with ngrep while registering, and every 60 seconds 
 after that.

 That no route to destination error is causing my hair to thin, and 
 my trunk provider tells me that it's usually something else, and 
 that the errormessage is not that descriptive.

 What can I do to get more/better debugging info? I can't figure out 
 what's wrong.

After looking at your iax.conf and extensions.conf I believe you are 
under the misconception that if you 'register' to a provider, then you 
can send and receive calls.   The fact is that you 'register' to receive 
calls, but you must define a trunk in order to Dial Out.  Your iax.conf 
[88821268] entry is not a trunk as you have not defined a host.  That is 
why you get cause 3 - No route to destination.  Asterisk does not have 
any host defined in order to route that call.  You need to talk to your 
provider for instructions on how to setup the trunk.

Andres
http://www.neuroredes.com


 Thanks!

 - Martin

 ( my iax.conf and extensions.conf on http://pastebin.com/mb0020bd )



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Re: [asterisk-users] IAX Hangup floods link with repeated VNAK and HANGUP

[Nathan Dennis]

Thanks for pointing that out Tony, Should have included that in my first
post.
Below is the version and the IAX config for each end

Server 1
Version : 1.4.18

IAX2.conf peer details

[brisbane]
type=friend
host=XXX.XXX.XXX.XXX
trunk=yes
context=internal
context=parkinglot
qualify=1
username=XXX
secret=
disallow=all
allow=g729

Server 2 
Version : 1.4.21.2

IAX2.conf peer details

[cairns]
type=friend
host=XXX.XXX.XXX.XXX
trunk=yes
context=internal
context=callagents
context=parkinglot
qualify=1
username=XXX
secret=
disallow=all
allow=g729



Nathan Dennis 
__ 
Integrated Solutions (QLD)P/LPhone: +61 (7) 4044 0300 
Direct: +61 (7) 4044
0302
124 Spence Street   Fax:+61 (7) 4041 6600
CAIRNS QLD 4868Mobile: 0418 608609

Australia 

E-mail: [EMAIL PROTECTED]
Web Site: www.i-solutions.net.au

Offices and agents in Cairns - Brisbane - Melbourne -- Adelaide --
Sydney
__ 
The information transmitted is intended only for the person or entity to
which 
it is addressed and may contain confidential and/or privileged material.

Any review, retransmission, dissemination or other use of, or taking of
any 
action in reliance upon, this information by persons or entities other
than the 
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contact 
the sender and delete the material from any computer.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Wednesday, 24 September 2008 8:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAX Hangup floods link with repeated
VNAKand HANGUP

In article
[EMAIL PROTECTED]
.au,
Nathan Dennis [EMAIL PROTECTED] wrote:
 We have been using asterisk for a while now but have recently needed 
 to install a second server in a remote office and set up a iax trunk 
 between the 2 servers. The dial plan seems to work well when I tested 
 it on the same LAN. However this afternoon I connected the system at 
 the remote office and made some calls. All the calls connect and work 
 fine, voice quality is great no really couldn't have hoped for better.

 Hang up the call and tried to make another call and nothing, the link 
 was not responding, after much trouble shooting I have found that 
 after the call is hung up the 2 asterisk servers seem to go into some 
 kind of loop sending each other message. I have pasted a debug for 
 both servers below that include everything from the start of the call 
 to after hangup. I have cut them short at the VNAK and Hangup cycle 
 just continues for 30seconds or so flooding the link completely.
  
 Any help you may be able to provide would be greatly appreciated

I can't help with your problem, sorry, but anyone who can help will need
to know exactly what version of Asterisk you have at each end.

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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[asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6

[Joseph]

On 09/24/08 13:50, Artem Makhutov wrote:
Hi,

On Tue, Sep 23, 2008 at 01:05:17PM -0600, Joseph wrote:
 I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo 
 portage but I think this version has a problem with RFC2833 DTMF signaling 
 and I don't think there 
 will be any newer version available anytime soon on portage.
 
 I need stable version, I'm using Asterisk mostly with ATA adapter (Linksys 
 and Sipura);  should I go to 1.6 or 1.4?

You can use the gentoo voip overlay. Asterisk 1.4.21.2 is included in
the overlay.

# emerge layman
# layman -a voip

I got this part: asterisk-1.4.21.2 but I need as add NVFaxDetect 
NVBackgroundDetect.
There is an instruction on wiki to compile it from source but I need some 
instructions on how to compile it on Gentoo.
When I try to compile current version from portage it keeps complaining it can 
not find /usr/bin/asterisk-config

Any idea how to go about it.

-- 
#Joseph

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Re: [asterisk-users] Asterisk 1.4 or 1.6

[Steve Totaro]

1.6 = Windows Vista :-P

On Tue, Sep 23, 2008 at 3:05 PM, Joseph [EMAIL PROTECTED] wrote:

 I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo
 portage but I think this version has a problem with RFC2833 DTMF signaling
 and I don't think there
 will be any newer version available anytime soon on portage.

 I need stable version, I'm using Asterisk mostly with ATA adapter (Linksys
 and Sipura);  should I go to 1.6 or 1.4?

 --
 #Joseph

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Re: [asterisk-users] Asterisk 1.4 or 1.6

[Paul Hales]

I would think

1.6 = Windows Vista

:)

PaulH


Steve Totaro wrote:
 1.6 = Windows Vista :-P

 On Tue, Sep 23, 2008 at 3:05 PM, Joseph [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 I need to upgrade my Asterisk, currently I'm using 1.2.27 from
 Gentoo portage but I think this version has a problem with RFC2833
 DTMF signaling and I don't think there
 will be any newer version available anytime soon on portage.

 I need stable version, I'm using Asterisk mostly with ATA adapter
 (Linksys and Sipura);  should I go to 1.6 or 1.4?

 --
 #Joseph

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[asterisk-users] Asterisk on VMware Workstation 6

[Michael J. Liberatore]

Hi, i am running a small personal asterisk server for my business, and
instead of getting a dedicated machine to run linux which would waste
power and money i decided to run it on my windows xp sp2 machine.  The
machine is barely used but it does have some crucial programs i need to
run in windows so reformating or dual booting is not an option.
 
Its basically a iax2 connection to my voip provider and a sip connection
to my phone.  It does work well, but the calls especially the voicemail
are all garbarled alot.  Its definetly not the provider or internet
connection because i use this provider for many clients asterisk setups
and i also even setup a temp. asterisk setup on this very pc to test to
make sure it was infact vmware causing the problem.  
 
I upgraded from vmware player to the latest vmware workstation hoping
that would fix the problem since its a better system but it hasnt.  I
also installed and compiled the vmware tools when  i installed
workstation version.  
 
Is this a known issue with vmware?  Is there a way to correct the issue
either on the windows/vmware side or on the asterisk/linux side?  Any
other ways to do this project?  i looked into astwind or something but
either couldnt get it to work or it was unreliable.
 
thanks
 
mike
 


This E-mail, including any attachments, may be intended solely for 
the personal and confidential use of the sender and recipient(s) named 
above. This message may include advisory, consultative and/or 
deliberative material and, as such, would be privileged and confidential 
and not a public document. Pursuant to 42 CFR, any information in this 
e-mail identifying a former, present, or potential client of Straight  Narrow 
is confidential. If you have received this e-mail in error, you must not 
review, transmit, convert to hard copy, copy, use or disseminate this e-mail or 
any attachments to it and you must delete this message. You are requested to 
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Re: [asterisk-users] Asterisk on VMware Workstation 6

[Matt Gibson]

Do you have ztdummy loaded in the VM? 

 

Thanks,

Matt G

 

: http://www.voipphreak.ca

: http://www.ratemydialplan.com

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
Liberatore
Sent: Wednesday, September 24, 2008 8:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk on VMware Workstation 6

 

Hi, i am running a small personal asterisk server for my business, and
instead of getting a dedicated machine to run linux which would waste power
and money i decided to run it on my windows xp sp2 machine.  The machine is
barely used but it does have some crucial programs i need to run in windows
so reformating or dual booting is not an option.

 

Its basically a iax2 connection to my voip provider and a sip connection to
my phone.  It does work well, but the calls especially the voicemail are all
garbarled alot.  Its definetly not the provider or internet connection
because i use this provider for many clients asterisk setups and i also even
setup a temp. asterisk setup on this very pc to test to make sure it was
infact vmware causing the problem.  

 

I upgraded from vmware player to the latest vmware workstation hoping that
would fix the problem since its a better system but it hasnt.  I also
installed and compiled the vmware tools when  i installed workstation
version.  

 

Is this a known issue with vmware?  Is there a way to correct the issue
either on the windows/vmware side or on the asterisk/linux side?  Any other
ways to do this project?  i looked into astwind or something but either
couldnt get it to work or it was unreliable.

 

thanks

 

mike

 

This E-mail, including any attachments, may be intended solely for the
personal and confidential use of the sender and recipient(s) named above.
This message may include advisory, consultative and/or deliberative material
and, as such, would be privileged and confidential and not a public
document. Pursuant to 42 CFR, any information in this e-mail identifying a
former, present, or potential client of Straight  Narrow is confidential.
If you have received this e-mail in error, you must not review, transmit,
convert to hard copy, copy, use or disseminate this e-mail or any
attachments to it and you must delete this message. You are requested to
notify the sender by return e-mail.

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[asterisk-users] Asterisk 1.4 is asking me for Mailbox #

[Joseph]

I just installed *-1.4 and when I enter mail extension it keep asking me for 
Mailbox #

I have in sip.conf under my extension mailbox=11 type=friend
*-1.2 was jumping straight to messages.

What did change?

-- 
#Joseph

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[asterisk-users] g729 capacity

[Robert McNaught]

Hi,

Does anyone know what happens if you exceed your G729 license
capacity?  Lets say you have 10 of 10 licenses being used by a PBX,
then an 11th call comes in set up to use G729.

Does asterisk has the ability to stop offering that codec in the SDP
once the capacity is reached.

Robert

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Re: [asterisk-users] Asterisk on VMware Workstation 6

[Dean Collins]

Mike, 

 

Buy an asterisk appliance like http://www.taa.com/products-vdex-40.html
problem solved.

 

If you are worried about good call quality it's either a dedicated pc or
a dedicated appliance, one or the other.

 

 

 


Cheers,

Dean



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
Liberatore
Sent: Wednesday, 24 September 2008 8:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk on VMware Workstation 6

 

Hi, i am running a small personal asterisk server for my business, and
instead of getting a dedicated machine to run linux which would waste
power and money i decided to run it on my windows xp sp2 machine.  The
machine is barely used but it does have some crucial programs i need to
run in windows so reformating or dual booting is not an option.

 

Its basically a iax2 connection to my voip provider and a sip connection
to my phone.  It does work well, but the calls especially the voicemail
are all garbarled alot.  Its definetly not the provider or internet
connection because i use this provider for many clients asterisk setups
and i also even setup a temp. asterisk setup on this very pc to test to
make sure it was infact vmware causing the problem.  

 

I upgraded from vmware player to the latest vmware workstation hoping
that would fix the problem since its a better system but it hasnt.  I
also installed and compiled the vmware tools when  i installed
workstation version.  

 

Is this a known issue with vmware?  Is there a way to correct the issue
either on the windows/vmware side or on the asterisk/linux side?  Any
other ways to do this project?  i looked into astwind or something but
either couldnt get it to work or it was unreliable.

 

thanks

 

mike

 

This E-mail, including any attachments, may be intended solely for the
personal and confidential use of the sender and recipient(s) named
above. This message may include advisory, consultative and/or
deliberative material and, as such, would be privileged and confidential
and not a public document. Pursuant to 42 CFR, any information in this
e-mail identifying a former, present, or potential client of Straight 
Narrow is confidential. If you have received this e-mail in error, you
must not review, transmit, convert to hard copy, copy, use or
disseminate this e-mail or any attachments to it and you must delete
this message. You are requested to notify the sender by return e-mail.

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Re: [asterisk-users] g729 capacity

[Igor H]

Hey Robert,

In my experience you get dead silence and the call goes through. We
run 1.4, it might be different for different setups.

On Wed, Sep 24, 2008 at 9:21 PM, Robert McNaught [EMAIL PROTECTED] wrote:
 Hi,

 Does anyone know what happens if you exceed your G729 license
 capacity?  Lets say you have 10 of 10 licenses being used by a PBX,
 then an 11th call comes in set up to use G729.

 Does asterisk has the ability to stop offering that codec in the SDP
 once the capacity is reached.

 Robert

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Re: [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6

[Jonn R Taylor]

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Wednesday, September 24, 2008 6:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 
1.4 or 1.6

On 09/24/08 13:50, Artem Makhutov wrote:
Hi,

On Tue, Sep 23, 2008 at 01:05:17PM -0600, Joseph wrote:
 I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo 
 portage but I think this version has a problem with RFC2833 DTMF signaling 
 and I don't think there 
 will be any newer version available anytime soon on portage.
 
 I need stable version, I'm using Asterisk mostly with ATA adapter (Linksys 
 and Sipura);  should I go to 1.6 or 1.4?

You can use the gentoo voip overlay. Asterisk 1.4.21.2 is included in
the overlay.

# emerge layman
# layman -a voip

I got this part: asterisk-1.4.21.2 but I need as add NVFaxDetect 
NVBackgroundDetect.
There is an instruction on wiki to compile it from source but I need some 
instructions on how to compile it on Gentoo.
When I try to compile current version from portage it keeps complaining it can 
not find /usr/bin/asterisk-config

Any idea how to go about it.

-- 
#Joseph

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http://sourceforge.net/projects/agx-ast-addons/

This is the best way to install them.

Jonn





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[asterisk-users] What happened to the register= setting in sip.conf?

[David Kerr]

I've setup a new asterisk box using asterisk 1.4.21.2 and used the
asterisk-gui 2.0 to configure trunks, etc. Everything is working fine except
that I am unable to register one of my trunks... stanaphone never responds
to a REGISTER request, and so they keep timing out.  Other trunks are
registering OK.  In looking at the sip debug, I see that asterisk is
providing a contact of sip:[EMAIL PROTECTED]  I seam to recall from past
experience that I need to register as sip:[EMAIL PROTECTED]
which was accomplished at the end of the register line in sip.conf
So I went to look for this line, but cannot find it in sip.conf nor any
other asterisk conf file.  I see a registersip=yes in users.conf, but that
is all.

Has something changed? Does asterisk no longer use a register= line in
sip.conf, and if so how are the parameters for the registration handled?

Thanks
David
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Re: [asterisk-users] g729 capacity

[Steve Totaro]

You urge and help Bret with his terribly intelligent G729 license sharing,
clearing house plan.  I think he should register a domain name and have a
PayPal Donation link.  I would certainly donate for the development and even
share a few licenses.

Not sure of the legal ramifications but the idea could cause a revolution in
licensing in general, not just G729.

Thanks,
Steve Totaro

On Wed, Sep 24, 2008 at 9:40 PM, Igor H [EMAIL PROTECTED] wrote:

 Hey Robert,

 In my experience you get dead silence and the call goes through. We
 run 1.4, it might be different for different setups.

 On Wed, Sep 24, 2008 at 9:21 PM, Robert McNaught [EMAIL PROTECTED]
 wrote:
  Hi,
 
  Does anyone know what happens if you exceed your G729 license
  capacity?  Lets say you have 10 of 10 licenses being used by a PBX,
  then an 11th call comes in set up to use G729.
 
  Does asterisk has the ability to stop offering that codec in the SDP
  once the capacity is reached.
 
  Robert
 
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Re: [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6

[Tzafrir Cohen]

On Wed, Sep 24, 2008 at 05:02:53PM -0600, Joseph wrote:

 I got this part: asterisk-1.4.21.2 but I need as add NVFaxDetect 
 NVBackgroundDetect.
 There is an instruction on wiki to compile it from source but I need some 
 instructions on how to compile it on Gentoo.
 When I try to compile current version from portage it keeps complaining it 
 can not find /usr/bin/asterisk-config
 
 Any idea how to go about it.

Look in the gentoo patches for a script called asterisk-config . It is
nor part of Asterisk. I also can't recall any pending patch to make it
so.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Zaptel/DAHDI ztdummy only

[Tzafrir Cohen]

On Wed, Sep 24, 2008 at 03:23:52PM -0700, Roderick A. Anderson wrote:
 Let me know if I should post this on the asterisk-dev list instead.
 
 I am building a Linux-Vserver (http://www.linux-vserver.org) host system 
 that will have several guests running Asterisk.  Since the guests can't 
 load kernel modules or do other dangerous stuff, but can access them I 
 built zaptel 1.4 and it is now loaded by the host.

modprobe ztdummy (alone) on the host. You'll have to create the basic
device files on the gusts from the host. You'll have to use static
device files.

 
 The issue I see is there will be no Zaptel hardware and these guests 
 will only do SIP, IAX, etc. but do appear to need ztdummy for timing 
 with other services.  So I'm looking for a way to not load (or even 
 build) all the other modules that come as part of Zaptel.
 
 Possible?

Yes.

 
 Will the complete change to DAHDI make this easier/harder?

No.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
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Re: [asterisk-users] g729 capacity

[Igor Hernandez]

Steve Totaro wrote:
 You urge and help Bret with his terribly intelligent G729 license
 sharing, clearing house plan.  I think he should register a domain name
 and have a PayPal Donation link.  I would certainly donate for the
 development and even share a few licenses.
 
 Not sure of the legal ramifications but the idea could cause a
 revolution in licensing in general, not just G729.
 
 Thanks,
 Steve Totaro
 
 On Wed, Sep 24, 2008 at 9:40 PM, Igor H [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
 
 Hey Robert,
 
 In my experience you get dead silence and the call goes through. We
 run 1.4, it might be different for different setups.
 
 On Wed, Sep 24, 2008 at 9:21 PM, Robert McNaught
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
  Hi,
 
  Does anyone know what happens if you exceed your G729 license
  capacity?  Lets say you have 10 of 10 licenses being used by a PBX,
  then an 11th call comes in set up to use G729.
 
  Does asterisk has the ability to stop offering that codec in the SDP
  once the capacity is reached.
 
  Robert
 
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Hey Steve,

Can you elaborate on that?

Thanks,

-- 
Igor Hernandez
Escape Communications
http://www.escapetel.com

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Re: [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6

[Joseph]

On 09/25/08 06:37, Tzafrir Cohen wrote:
On Wed, Sep 24, 2008 at 05:02:53PM -0600, Joseph wrote:

 I got this part: asterisk-1.4.21.2 but I need as add NVFaxDetect 
 NVBackgroundDetect.
 There is an instruction on wiki to compile it from source but I need some 
 instructions on how to compile it on Gentoo.
 When I try to compile current version from portage it keeps complaining it 
 can not find /usr/bin/asterisk-config
 
 Any idea how to go about it.

Look in the gentoo patches for a script called asterisk-config . It is
nor part of Asterisk. I also can't recall any pending patch to make it
so.


Yes, I have that that file on my other box in: /usr/bin/asterisk-config (still 
running 1.2.27) I could try to to change it.
I don't think I could change much except the AST_VERSION=
see below:


---copy
#!/bin/sh
#
# asterisk-config
#
# Copyright (C) 2004 Stefan Knoblich [EMAIL PROTECTED]
#

# /*
# Changes:
#
# 0.0.2 (stkn: 20041121)
#   Clean-ups, renamed some options (more configure alike)
#
# 0.0.1 (stkn: 20041114)
#   Yeah it's ugly as hell, but it does it's job
# */

##
# These get replaced by sed...
#

SOLINK='-shared -Xlinker -x'
CFLAGS='-O2 -march=athlon-xp -fomit-frame-pointer  -pipe  -Wall 
-Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations  -Iinclude 
-I../include -D_REENTRANT 
-D_GNU_SOURCE  -O2 -march=athlon-xp -fomit-frame-pointer 
-fomit-frame-pointer '
LIBS='-ldl -lpthread -lncurses -lm -lresolv   -lssl'

AST_PREFIX=
AST_LIBDIR=/usr/lib/asterisk
AST_ETCDIR=/etc/asterisk
AST_MODDIR=/usr/lib/asterisk/modules
AST_AGIDIR=/var/lib/asterisk/agi-bin
AST_INCDIR=/usr/include/asterisk
AST_MANDIR=/usr/share/man
AST_LOGDIR=/var/log/asterisk
AST_VARLIBDIR=/var/lib/asterisk
AST_VARRUNDIR=/var/run/asterisk
AST_SPOOLDIR=/var/spool/asterisk
[EMAIL PROTECTED]@
AST_VERSION=1.2.27

##
# Don't even think about touching anything below...
#
...
so I won't even print it.


My problme is that few lines in a source code needs to be modified before 
compiling it.  Changing the source code is a simple thing but now the ebuild 
needs to be   
modified as well to point to the source code; too many problems.

I think maybe it is time to dump Asterisk-ebuild version and get it from 
source. It is less problems and simpler solution.

-- 
#Joseph

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Re: [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6

[Joseph]

On 09/24/08 21:19, Jonn R Taylor wrote:
http://sourceforge.net/projects/agx-ast-addons/

This is the best way to install them.

Jonn

No, NVFaxDetect is not part of Extra AddOns.

The correct instruction are here:
http://www.switzersolutions.com/technology-articles/6-asterisk-pbx/9-asterisk-14-and-nvfaxdetect.html

-- 
#Joseph

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Re: [asterisk-users] dundi and regcontext

[technocrat voip]

According to Your description this is a phone problem.

Asterisk behaves as its expected.

post your dundi.conf to dig more in to this.

regards
rama

On Wed, Sep 24, 2008 at 9:52 PM, ronald ramos [EMAIL PROTECTED]wrote:

 hi,

 when a user register on my asterisk i can see it adding Noop for that
 extension, but after awhile i won't see it anymore:

 what are the reasons for it being removed on the dynamic context?
 one thing i found when i unregister it's removed.

 dialplan show myregcontext
 [ Context 'myregcontext' created by 'SIP' ]
   '100500' =   1. Noop(100500)   [SIP]
   '112802' =   1. Noop(112802)   [SIP]

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

 [ Context 'pfingobizsip' created by 'SIP' ]

 -= 0 extensions (0 priorities) in 1 context. =-

 my prob is when it's removed dundi cant find it anymore so a user calling
 from server 1 cannot call user that is in server 2.

 i've set re-registration to very low (1 minute) to monitor if my phone
 re-register and to see if it will be added again on the regcontext.
 but i don't even see it unregistering after 1 minute i only unregistering
 when i am using x-lite and closing x-lite, i dont see x-lite re-registering
 if i just leave the softphone open. any idea?

 regards,
 ron


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Re: [asterisk-users] g729 capacity

[Robert McNaught]

anyone know if there is an SNMP probe which can monitor the usage of
G729 licenses - I have had a browse through the MIB file and a google
and did not see one? - otherwise you would have no way of knowing if
it happening (other than people screaming at you!)

Robert

On Wed, Sep 24, 2008 at 8:08 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
 You urge and help Bret with his terribly intelligent G729 license sharing,
 clearing house plan.  I think he should register a domain name and have a
 PayPal Donation link.  I would certainly donate for the development and even
 share a few licenses.

 Not sure of the legal ramifications but the idea could cause a revolution in
 licensing in general, not just G729.

 Thanks,
 Steve Totaro

 On Wed, Sep 24, 2008 at 9:40 PM, Igor H [EMAIL PROTECTED] wrote:

 Hey Robert,

 In my experience you get dead silence and the call goes through. We
 run 1.4, it might be different for different setups.

 On Wed, Sep 24, 2008 at 9:21 PM, Robert McNaught [EMAIL PROTECTED]
 wrote:
  Hi,
 
  Does anyone know what happens if you exceed your G729 license
  capacity?  Lets say you have 10 of 10 licenses being used by a PBX,
  then an 11th call comes in set up to use G729.
 
  Does asterisk has the ability to stop offering that codec in the SDP
  once the capacity is reached.
 
  Robert
 
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Re: [asterisk-users] NVFaxDetect (1.0.6), NVBackgroundDetect was: Asterisk 1.4 or 1.6

[Tzafrir Cohen]

On Wed, Sep 24, 2008 at 10:25:45PM -0600, Joseph wrote:

 My problme is that few lines in a source code needs to be modified 
 before compiling it.  Changing the source code is a simple thing but 
 now the ebuild needs to be modified as well to point to the source code; 
 too many problems.

Asterisk 1.2 - 1.4 is a change in the build system. Most of it (except
menuselect) is for the better). Adjusting your build scripts for that
(and a packaging system is essentially a glorified build script) only
takes some work.

I would appreciate it if you hadn't kept your patches for yourselves.
This would have also saved you some time on the next release (there are
already RCs of 1.6.0 for yor test-building pleassure).

BTW: maybe you need a newer version of nvfaxdetect? There has been one
released, IIRC. If not, there should be such a version on agx's modules
addons collection. Again, keeping your changes to yourself is bad.

Also recall that for 1.4 and above you must define AST_MODULE. If you
don't do so, you get very strange errors.

 
 I think maybe it is time to dump Asterisk-ebuild version and get it 
 from source. It is less problems and simpler solution.

Until you need to figure out whatever is installed on your system, and
where this module comes from.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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