[asterisk-users] Asterisk and Voice Recognition Sphinx

[Marco Sambo]

Hi all,
someone has used the voice recognition software named Sphinx??? Can he tell
me how to use and its performance???

Thanks

Marco
___
-- 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

Re: [asterisk-users] Using multiple 'peer' identities on one phone with 1.4

[Olle E. Johansson]

7 apr 2009 kl. 18.26 skrev Florian Hackenberger:

> On Tuesday 07 April 2009, Olle E. Johansson wrote:
>> I don't see any problems there. YOu still have devices with states,
>> as you would have with authentication. Of course, it still depends on
>> your configuration. But authentication should not affect states.
> Ok, thanks for that, I'll have a look at openSER.
>
>> If you use the limitonpeer setting, all states for both the user and
>> the peer part of a friend will only be handled by the peer, which is
>> the device watched for subscriptions.
> That worked like a charm, thanks!
Good to hear.
>
>> There was recently also an
>> overhaul of the states for queues, with a patch to 1.4 that made it
>> possible
>> to build a stronger relationship between a queue member and
>> a state object.
> Could you please point me to a bug report or an SVN revision?

http://lists.digium.com/pipermail/asterisk-commits/2009-March/032220.html

/O


---
* Olle E. Johansson - o...@edvina.net
* Asterisk Training http://edvina.net/training/




___
-- 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

Re: [asterisk-users] i have a probleme and my asterisk and ovh

[Henry]

sip show peer ovh

  * Name   : ovh
  Secret   : 
  MD5Secret: 
  Context  : entrant-ovh
  Subscr.Cont. : 
  Language : fr
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic  : No
  Callerid : "" <>
  MaxCallBR: 384 kbps
  Expire   : -1
  Insecure : port,invite
  Nat  : RFC3581
  ACL  : No
  T38 pt UDPTL : Yes
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode : auto
  Timer T1 : 500
  Timer B  : 32000
  ToHost   : sip.ovh.net
  Addr->IP : 91.121.129.17 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Transport: UDP
  Def. Username: 0033972112355
  SIP Options  : (none)
  Codecs   : 0x100 (g729)
  Codec Order  : (g729:20)
  Auto-Framing :  No
  100 on REG   : No
  Status   : UNREACHABLE
  Useragent:
  Reg. Contact :
  Qualify Freq : 6 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess : 90 secs


---
Retransmitting #1 (no NAT) to 91.121.129.17:5060:
OPTIONS sip:sip.ovh.net SIP/2.0
Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport
Max-Forwards: 70
From: "asterisk" ;tag=as1545fb99
To: 
Contact: 
Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.8
Date: Wed, 08 Apr 2009 05:57:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #6 (no NAT) to 91.121.129.17:5060:
REGISTER sip:91.121.129.17 SIP/2.0
Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK312a379b;rport
Max-Forwards: 70
From: ;tag=as16505dec
To: 
Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51
CSeq: 1465 REGISTER
User-Agent: Asterisk PBX 1.6.0.8
Authorization: Digest username="0033972112355", realm="sip.ovh.net", 
algorithm=MD5, uri="sip:91.121.129.17", 
nonce="0019c92d503f745637b43af4264a11db", 
response="04e848af655d00e03d032d9a1c2fae09", opaque="001934772ef6ed5"
Expires: 120
Contact: 
Event: registration
Content-Length: 0


---
Retransmitting #2 (no NAT) to 91.121.129.17:5060:
OPTIONS sip:sip.ovh.net SIP/2.0
Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport
Max-Forwards: 70
From: "asterisk" ;tag=as1545fb99
To: 
Contact: 
Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.8
Date: Wed, 08 Apr 2009 05:57:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (no NAT) to 91.121.129.17:5060:
OPTIONS sip:sip.ovh.net SIP/2.0
Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport
Max-Forwards: 70
From: "asterisk" ;tag=as1545fb99
To: 
Contact: 
Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.8
Date: Wed, 08 Apr 2009 05:57:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (no NAT) to 91.121.129.17:5060:
OPTIONS sip:sip.ovh.net SIP/2.0
Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport
Max-Forwards: 70
From: "asterisk" ;tag=as1545fb99
To: 
Contact: 
Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.8
Date: Wed, 08 Apr 2009 05:57:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog 
'578ac87b06eaa6526aa313e130be3...@172.20.1.1' Method: OPTIONS
[Apr  8 07:57:48] NOTICE[25949]: chan_sip.c:9490 sip_reg_timeout:-- 
Registration for '0033972112...@91.121.129.17' timed out, trying again 
(Attempt #1262)
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 91.121.129.17:5060:
REGISTER sip:91.121.129.17 SIP/2.0
Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK40d5f950;rport
Max-Forwards: 70
From: ;tag=as02687bc2
To: 
Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51
CSeq: 1466 REGISTER
User-Agent: Asterisk PBX 1.6.0.8
Authorization: Digest username="0033972112355", realm="sip.ovh.net", 
algorithm=MD5, uri="sip:91.121.129.17", 
nonce="0019c92d503f745637b43af4264a11db", 
response="04e848af655d00e03d032d9a1c2fae09", opaque="001934772ef6ed5"
Expires: 120
Contact: 
Event: registration
Content-Length: 0


---
Really destroying SIP dialog 
'165ff552001c7f1e202e67200ae67...@172.25.3.51' Method: REGISTER
Retransmitting #1 (no NAT) to 91.121.129.17:5060:
REGISTER sip:91.121.129.17 SIP/2.0
Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK40d5f950;rport
Max-Forwards: 70
From: ;tag=as02687bc2
To: 
Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51
CSeq: 1466 REGISTER
User-Agent: Asterisk PBX 1.6.0.8
Authoriz

Re: [asterisk-users] app_backticks and 1.6

[Olivier]

I've updated http://www.voip-info.org/wiki/view/Asterisk+func+shell with an
example to test file existence.
___
-- 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

[asterisk-users] Call Pickup Works w/Linksys ATA, not with Cisco 7940G

[George Pajari]

I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G phones. Unit obtains SIP trunking
from an ITSP (server has no PCI boards).

*8 Call Pickup works fine from any of the phones connected using the
Linksys SPA2102.

*8 Call Pickup does not work from the Cisco 7940G phones
(chan_sip.c:13977 handle_request_invite: Nothing to pick up for
000d6556-eeb3001c-76b88543-7f51d...@192.168.0.211)

What could the difference be?

Below you will find:
 (a) the "sip show peer nnn" for an ATA extension and a Cisco extension
 (b) the SIP debug trace for (i) a successful call pickup from the ATA
and (ii) an unsuccessful call pickup from the Cisco

Any light anyone can shed on the perplexing problem would be most
appreciated. I have a forehead-shaped dent in the wall that is growing
larger.


Linksys SPA2102 ATA "sip show peer 101":

* Name   : 101
  Secret   : 
  MD5Secret: 
  Context  : numberplan-custom-1
  Subscr.Cont. : 
  Language : 
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup: 1
  Pickupgroup  : 1
  Mailbox  : 101
  VM Extension : asterisk
  LastMsgsSent : 0/3
  Call limit   : 0
  Dynamic  : Yes
  Callerid : "Dxx Gxxx" <604-123->
  MaxCallBR: 384 kbps
  Expire   : 1861
  Insecure : no
  Nat  : RFC3581
  ACL  : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   : 
  Addr->IP : 192.168.0.205 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 101
  SIP Options  : replaces replace 
  Codecs   : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Auto-Framing:  No 
  Status   : OK (5 ms)
  Useragent: Linksys/SPA2102-5.2.3
  Reg. Contact : sip:1...@192.168.0.205:5060




Cisco 7940G Phone "sip show peer 106":

* Name   : 106
  Secret   : 
  MD5Secret: 
  Context  : numberplan-custom-1
  Subscr.Cont. : 
  Language : 
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup: 1
  Pickupgroup  : 1
  Mailbox  : 106
  VM Extension : asterisk
  LastMsgsSent : 3/1
  Call limit   : 0
  Dynamic  : Yes
  Callerid : "Cxxx Nxxx" <6041234567>
  MaxCallBR: 384 kbps
  Expire   : 247
  Insecure : no
  Nat  : RFC3581
  ACL  : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   : 
  Addr->IP : 192.168.0.211 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 106
  SIP Options  : (none)
  Codecs   : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Auto-Framing:  No 
  Status   : OK (195 ms)
  Useragent: Cisco-CP7940G/8.0
  Reg. Contact : sip:1...@192.168.0.211:5060;transport=udp



Successful *8 Call Pickup (SIP Trace)


<--- SIP read from 192.168.0.205:5060 --->
INVITE sip:*...@192.168.0.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-e7b4459c
From: 101 ;tag=22b459c8b65178bco0
To: 
Remote-Party-ID: 101 ;screen=yes;party=calling
Call-ID: 94dc7b8-591d6...@192.168.0.205
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 101 
Expires: 240
User-Agent: Linksys/SPA2102-5.2.3
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 922981 922981 IN IP4 192.168.0.205
s=-
c=IN IP4 192.168.0.205
t=0 0
m=audio 16412 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<->
--- (15 headers 20 lines) ---
Sending to 192.168.0.205 : 5060 (no NAT)
Using INVITE request as basis request - 94dc7b8-591d6...@192.168.0.205
Found peer '101'
tg2*CLI> 
<--- Reliably Transmitting (no NAT) to 192.168.0.205:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK-e7b4459c;received=192.168.0.205
From: 101 ;tag=22b459c8b65178bco0
To: ;tag=as4bf7113a
Call-ID: 94dc7b8-591d6...@192.168.0.205
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="637bf838"
Content-Length: 0


<>
Scheduling destruction of SIP dialog '94dc7b8-591d6...@192.168.0.205' in 6400 ms (Method: INVITE)
tg2*CLI> 
<--- SIP read from 192.168.0.205:5060 --->
ACK sip:*...@192.168.0.12 S

Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal

[Olivier]

2009/4/7 Mark Michelson 

> Philipp Kempgen wrote:
> > Olivier schrieb:
> >> 2009/4/7 Philipp Kempgen 
> >>> Olivier schrieb:
>  I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext
> in
> >>> an
>  AEL2 file like this :
>  SendText(${BASE64_DECODE(DQ==)});
> 
> 
>  Value sent (8 bytes long) is very strange :
>  Content-Type: text/plain;charset=UTF-8
>  Content-Length: 8
> 
>  �ez?==
> >>> I doubt you will find a way to properly pass CR or LF to an
> >>> application in extensions.(conf|ael) but keep us in the loop.
> >
> >> It's strange how such a silly thing is somehow keeping me from centrally
> >> managing phones forwarding : I can display a phone is forwarded but I
> can't
> >> gracefully return to previous status ...
> >
> > BTW (developer's question) is there a reason why SendText() resp.
> > sendtext_exec() refuses to send zero-length data?
> >
>
> I can't point to any specific reason. I assume that whoever wrote the
> application probably thought that attempting to send zero-length data was
> pointless and that if no data were passed to the application, it likely was
> due
> to an error by the user.


The phone I'm working on (Thomson ST2030) would display in slow blinking,
inversed letters (white on black) any text received in SIP MESSAGE.
Display duration is unlimited.
To erase an old message, you must send a single carriage return (or maybe an
empty string).

I'm wondering how many phones behave like this ?

Maybe, sendtext should then be refactored to accommodate this.

>
>
> Mark Michelson
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] chan_mobile sms compatible phone

[David fire]

hi
i look the list at http://www.voip-info.org/wiki/view/chan_mobile i tryed
whit an N80 and many non listed phones i cant get any one to work to send
sms.
someone know a phone thats really works to send sms? wich one?
thanks!!!

David

-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
___
-- 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

Re: [asterisk-users] Best Practice Advice?

[David Backeberg]

On Tue, Apr 7, 2009 at 11:54 AM, Gabriel - IP Guys
 wrote:
> I have a asterisk setup that is currently running on version 1.4.15 – I wish
> to upgrade or migrate this instance to the current asterisk stable,
>  1.6.0.6. It is my intention to build a FC8 box, then install asterisk from
> source, and begin to migrate over the configuration.

Just rip the band-aid off and go straight for 1.6.0.9. Do not stop at
1.6.0.6 if this is taking public SIP traffic. Also, FC10 is out. You
should probably grab that first.

>The thing is, this sounds so simple in my head, and I’ve had enough issues 
>with asterisk, to
> know that life isn’t simple!

Read the release notes, crank up the verbose on the console, and let
it rip. Some things will break, some things will be fine, and several
things are dramatically better than they used to be in 1.4.

___
-- 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

Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall - canunicall affect zaptel commands?

[John Millican]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, April 07, 2009 6:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall -
canunicall affect zaptel commands?

The message includes a host of irrelevant and relevant information. The
question is not clear. It is a horrible piece of top-posting mess.

Please provide the relevant configuration again and clarify your answer.


What hardware do you have? What connections do you have? Are they
working OK?

Generally chan_zap and chan_unicall should not handle the same spans

-- 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

Forwarded message -- From: Juan Carlos Huerta
 Date: 07-abr-2009 13:41 Subject: Re:
[asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands?
To: asterisk...@lists.digium.com Please wirte to
asterisk-users@lists.digium.com to get help about this problem. Juan
Carlos ~ Lo que no te mata te fortalece ~ On Tue, Apr 7, 2009 at 1:35
PM, Giovanny Magallanes  wrote:
> > Thanks, Juan  Carlos.
> >
> > Yes, I still can t type the commands in the asterisk console . I
dont have
> > dial tone for FXS ports .
> >
> >
> > Regards.
> >
> > Giovanni.
> >
> >
> > 2009/4/7, Juan Carlos Huerta :
>> >>
>> >> And what is the result? still with zap commands problem?
>> >>
>> >> Juan Carlos
>> >> ~ Lo que no te mata te fortalece ~
>> >>
>> >>
>> >>
>> >> On Tue, Apr 7, 2009 at 12:49 PM, Giovanny Magallanes
>> >>  wrote:
>>> >> > I did it:
>>> >> >
>>> >> > elastix*CLI> load chan_zap.so
>>> >> > The 'load' command is deprecated and will be removed in a future
>>> >> > release.
>>> >> > Please use 'module load' instead.
>>> >> >   == Parsing '/etc/asterisk/zapata.conf': Found
>>> >> >   == Parsing '/etc/asterisk/zapata_additional.conf': Found
>>> >> >   == Parsing '/etc/asterisk/zapata-channels.conf': Found
>>> >> > -- Registered channel 63, FXO Kewlstart signalling
>>> >> > -- Registered channel 64, FXO Kewlstart signalling
>>> >> > -- Registered channel 65, FXO Kewlstart signalling
>>> >> > -- Registered channel 66, FXO Kewlstart signalling
>>> >> > -- Registered channel 67, FXO Kewlstart signalling
>>> >> > -- Registered channel 68, FXO Kewlstart signalling
>>> >> > -- Registered channel 68, FXO Kewlstart signalling
>>> >> > -- Registered channel 69, FXO Kewlstart signalling
>>> >> > -- Registered channel 70, FXO Kewlstart signalling
>>> >> > -- Registered channel 71, FXO Kewlstart signalling
>>> >> > -- Registered channel 72, FXO Kewlstart signalling
>>> >> > -- Registered channel 73, FXO Kewlstart signalling
>>> >> > -- Registered channel 74, FXO Kewlstart signalling
>>> >> > -- Registered channel 75, FXO Kewlstart signalling
>>> >> > -- Registered channel 76, FXO Kewlstart signalling
>>> >> > -- Registered channel 77, FXO Kewlstart signalling
>>> >> > -- Registered channel 78, FXO Kewlstart signalling
>>> >> > elastix*CLI>
>>> >> >
>>> >> > Thank you Juan
>>> >> >
>>> >> > Giovanni
>>> >> >
>>> >> >
>>> >> > 2009/4/7, Juan Carlos Huerta :
 >> >>
 >> >> Try to load the chan_zap.so module manually to see if you get
some
 >> >> error.
 >> >>
 >> >> Juan Carlos
 >> >> ~ Lo que no te mata te fortalece ~
 >> >>
 >> >>
 >> >>
 >> >> On Tue, Apr 7, 2009 at 12:44 PM, Giovanny Magallanes
 >> >>  wrote:
> >> >> > I'm using Elastix 1.1-8 with:
> >> >> >
> >> >> > asterisk-1.4.19
> >> >> > spandsp-0.0.4
> >> >> > unicall-0.0.5pre1
> >> >> > zaptel-1.4.9.2
> >> >> > unicall-0.0.5pre1
> >> >> > libmfcr2-0.0.3
> >> >> > libsupertone-0.0.2
> >> >> > libunicall-0.0.3
> >> >> >
> >> >> > Regards,
> >> >> >
> >> >> > Giovanni
> >> >> >
> >> >> > 2009/4/7, Giovanny Magallanes :
>> >> >> >>
>> >> >> >> Ok, Thanks. The chan_zap commands are unavailable, but
unicall
>> >> >> >> commands
>> >> >> >> and my E1 MFC/R2 are OK.
>> >> >> >>
>> >> >> >> Giovanni
>> >> >> >>
>> >> >> >>
>> >> >> >> 2009/4/7, Moises Silva :
>>> >> >> >>>
>>> >> >> >>> I don't understand your problem. And no, unicall has
nothing to do
>>> >> >> >>> with chan_zap.so commands.
>>> >> >> >>>
>>> >> >> >>> Please, in the future, don't hijack threads, open a
new thread for
>>> >> >> >>> your discussion. This time I changed the subject already.
>>> >> >> >>>
>>> >> >> >>> Moy
>>> >> >> >>>
>>> >> >> >>> On Tue, Apr 7, 2009 at 1:08 AM, Giovanny Magallanes
>>> >> >> >>>  wrote:
 >> >> >>> > Hi, Guys.
 >> >> >>> >
 >> >> >>> > I did not type any zaptel commands in asterisk
console. I have
 >> >> >>> > installed
 >> >> >>> > Elast

Re: [asterisk-users] Best Practice Advice?

[Paul Hales]

I would upgrade to the latest 1.4, if stable is what is needed.

PaulH


Gabriel - IP Guys wrote:
>
> Dear All,
>
> I have a asterisk setup that is currently running on version 1.4.15 –
> I wish to upgrade or migrate this instance to the current asterisk
> stable, 1.6.0.6. It is my intention to build a FC8 box, then install
> asterisk from source, and begin to migrate over the configuration. The
> thing is, this sounds so simple in my head, and I’ve had enough issues
> with asterisk, to know that life isn’t simple!
>
> What I plan to do, is to copy the old configuration over to a box
> running FC8 – and then compile and run asterisk 1.4.15 – and gradually
> upgrade it, until I reach 1.6.0.6 – Any input on this matter will be
> appreciated. Thank you
>
> ---
>
> Mr Gabriel
>
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Hacked

[Martin]

I thought so. Unless someone can write a buffer overrun code to email
them the sip.conf or other config files
then you should be fine if you don't provision unsecured contexts to
dial out to PSTN ...

there was a buffer overrun in chan_sip but it was a couple years ago

Martin

On Tue, Apr 7, 2009 at 11:28 AM, Tilghman Lesher
 wrote:
> On Monday 06 April 2009 19:22:30 Martin wrote:
>> Can you give more information about this vulnerability ?
>
> It's unlikely that it's this vulnerability.  Every Asterisk box allows guest
> access to the machine, by default.  The context it goes to is generally
> the "default" context.  This is what allows you to publish an addresses
> like sip:foo.example.com and have it get through to your company.  There's
> no preexisting relationship between caller and callee; it's merely a method
> of contacting the machine.
>
> What is a vulnerability is the way that some people have configured this.
> They put in that context patterns that can dial out.  There's nothing
> specifically wrong with this configuration, from an Asterisk perspective;
> however, in many cases, guest access is not what the administrator intended;
> thus, the machine may be used to make illicit outbound telephone calls by
> anybody who sends a SIP call to that machine.
>
> The recent vulnerability had nothing to do with this, but with the ability of
> an attacker to scan a SIP server for legitimate usernames and passwords.
> This, by the way, merely took advantage of the SIP protocol, as written.
> Normally, SIP allows you to differentiate between invalid usernames (404) and
> invalid passwords (403).  What we closed in the recent vulnerability patch was
> to allow administrators to send back 403, regardless of whether the username
> existed or not.
>
> --
> Tilghman
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall - canunicall affect zaptel commands?

[Cary Fitch]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, April 07, 2009 6:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall -
canunicall affect zaptel commands?

The message includes a host of irrelevant and relevant information. The
question is not clear. It is a horrible piece of top-posting mess.

Please provide the relevant configuration again and clarify your answer.


What hardware do you have? What connections do you have? Are they
working OK?

Generally chan_zap and chan_unicall should not handle the same spans

-- 
   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

___
-- 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


-- Forwarded message --
From: Juan Carlos Huerta 
Date: 07-abr-2009 13:41
Subject: Re: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel
commands?
To: asterisk...@lists.digium.com

Please wirte to asterisk-users@lists.digium.com to get help about this
problem.

Juan Carlos
~ Lo que no te mata te fortalece ~



On Tue, Apr 7, 2009 at 1:35 PM, Giovanny Magallanes
 wrote:
> Thanks, Juan  Carlos.
>
> Yes, I still can t type the commands in the asterisk console . I dont have
> dial tone for FXS ports .
>
>
> Regards.
>
> Giovanni.
>
>
> 2009/4/7, Juan Carlos Huerta :
>>
>> And what is the result? still with zap commands problem?
>>
>> Juan Carlos
>> ~ Lo que no te mata te fortalece ~
>>
>>
>>
>> On Tue, Apr 7, 2009 at 12:49 PM, Giovanny Magallanes
>>  wrote:
>> > I did it:
>> >
>> > elastix*CLI> load chan_zap.so
>> > The 'load' command is deprecated and will be removed in a future
>> > release.
>> > Please use 'module load' instead.
>> >   == Parsing '/etc/asterisk/zapata.conf': Found
>> >   == Parsing '/etc/asterisk/zapata_additional.conf': Found
>> >   == Parsing '/etc/asterisk/zapata-channels.conf': Found
>> > -- Registered channel 63, FXO Kewlstart signalling
>> > -- Registered channel 64, FXO Kewlstart signalling
>> > -- Registered channel 65, FXO Kewlstart signalling
>> > -- Registered channel 66, FXO Kewlstart signalling
>> > -- Registered channel 67, FXO Kewlstart signalling
>> > -- Registered channel 68, FXO Kewlstart signalling
>> > -- Registered channel 68, FXO Kewlstart signalling
>> > -- Registered channel 69, FXO Kewlstart signalling
>> > -- Registered channel 70, FXO Kewlstart signalling
>> > -- Registered channel 71, FXO Kewlstart signalling
>> > -- Registered channel 72, FXO Kewlstart signalling
>> > -- Registered channel 73, FXO Kewlstart signalling
>> > -- Registered channel 74, FXO Kewlstart signalling
>> > -- Registered channel 75, FXO Kewlstart signalling
>> > -- Registered channel 76, FXO Kewlstart signalling
>> > -- Registered channel 77, FXO Kewlstart signalling
>> > -- Registered channel 78, FXO Kewlstart signalling
>> > elastix*CLI>
>> >
>> > Thank you Juan
>> >
>> > Giovanni
>> >
>> >
>> > 2009/4/7, Juan Carlos Huerta :
>> >>
>> >> Try to load the chan_zap.so module manually to see if you get some
>> >> error.
>> >>
>> >> Juan Carlos
>> >> ~ Lo que no te mata te fortalece ~
>> >>
>> >>
>> >>
>> >> On Tue, Apr 7, 2009 at 12:44 PM, Giovanny Magallanes
>> >>  wrote:
>> >> > I'm using Elastix 1.1-8 with:
>> >> >
>> >> > asterisk-1.4.19
>> >> > spandsp-0.0.4
>> >> > unicall-0.0.5pre1
>> >> > zaptel-1.4.9.2
>> >> > unicall-0.0.5pre1
>> >> > libmfcr2-0.0.3
>> >> > libsupertone-0.0.2
>> >> > libunicall-0.0.3
>> >> >
>> >> > Regards,
>> >> >
>> >> > Giovanni
>> >> >
>> >> > 2009/4/7, Giovanny Magallanes :
>> >> >>
>> >> >> Ok, Thanks. The chan_zap commands are unavailable, but unicall
>> >> >> commands
>> >> >> and my E1 MFC/R2 are OK.
>> >> >>
>> >> >> Giovanni
>> >> >>
>> >> >>
>> >> >> 2009/4/7, Moises Silva :
>> >> >>>
>> >> >>> I don't understand your problem. And no, unicall has nothing to do
>> >> >>> with chan_zap.so commands.
>> >> >>>
>> >> >>> Please, in the future, don't hijack threads, open a new thread for
>> >> >>> your discussion. This time I changed the subject already.
>> >> >>>
>> >> >>> Moy
>> >> >>>
>> >> >>> On Tue, Apr 7, 2009 at 1:08 AM, Giovanny Magallanes
>> >> >>>  wrote:
>> >> >>> > Hi, Guys.
>> >> >>> >
>> >> >>> > I did not type any zaptel commands in asterisk console. I have
>> >> >>> > installed
>> >> >>> > Elastix 1.1-8 with unicall and it is working fine with a Digium
>> >> >>> > TE212-P.
>> >> >>> >
>> >> >>> > [r...@elastix asterisk]# find -P / -name chan_zap.so
>> >> >>> > /usr/lib/asterisk/modules/chan_zap.so
>> >> >>> >
>> >> >>> >

Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands?

[Tzafrir Cohen]

The message includes a host of irrelevant and relevant information. The
question is not clear. It is a horrible piece of top-posting mess.

Please provide the relevant configuration again and clarify your answer.


What hardware do you have? What connections do you have? Are they
working OK?

Generally chan_zap and chan_unicall should not handle the same spans

-- 
   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

___
-- 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

[asterisk-users] dahdi_dummy: Unable to register DAHDI rtc driver

[David Backeberg]

Hello there:

I think I have a silly kernel configuration problem. I'm running:
* vanilla 2.6.27.10 kernel built from source
* dahdi-2.1.0.4 built from source

So far so good,
dahdi module loads just fine:
dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.1.0.4

when I try to:
hal04 dahdi # modprobe dahdi_dummy
FATAL: Error inserting dahdi_dummy
(/lib/modules/2.6.27.10/dahdi/dahdi_dummy.ko): Input/output error

kernel messages gives me:
dahdi_dummy: Unable to register DAHDI rtc driver

I'm probably doing something silly here.

Then I was curious, so I backed up to a 2.6.25.9 kernel I already had,
and dahdi_dummy loaded just fine:
dahdi_dummy: RTC rate is 1024

Does anybody know whether:
* something changed in mainline kernel that breaks dahdi
* there was a new kernel parameter that I should have set differently?

No emergency, more like an observation. I'm just backing off kernel
version for now.

___
-- 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

Re: [asterisk-users] i have a probleme and my asterisk and ovh

[Danny Nicholas]

And sip set debug peer ovh?


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Sent: Tuesday, April 07, 2009 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] i have a probleme and my asterisk and ovh

[ovh]
type=peer
secret=
username=0033972xx
fromuser=0033972xx
host=sip.ovh.net
canreinvite=no
disallow=all
allow=g729
tos_sip=1; Sets TOS for SIP packets.
tos_audio=1   ; Sets TOS for RTP audio packets.
tos_video=1
dtmfmode=rfc28335
relaxdtmf=yes
nat=yes
qualify=yes
insecure=port,invite
context=entrant-ovh

thank you.

Danny Nicholas a écrit :
> Show us your sip.conf
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
> Sent: Tuesday, April 07, 2009 2:54 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] i have a probleme and my asterisk and ovh
>
> hello every body
>
> my connexion on ovh to pass in UNREACHABLE and not reidentified were not 
> reboot the server.
>
> [Apr  7 20:17:21] NOTICE[19947]: chan_sip.c:15605 
> handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
> [Apr  7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswer: 
> Peer 'ovh' is now UNREACHABLE!  Last qualify: 2067
>
> but my probleme is the adress ip 172.25.3.51 is not my adress.
>
> Really destroying SIP dialog 
> '13ff06ae3e4bb3bf04987f5f5b497...@172.20.1.1' Method: OPTIONS
> Really destroying SIP dialog 
> '6eac266b68dbc2566209fbb74aec7...@172.25.3.51' Method: REGISTER
>
> I do not know where she comes out, my asterisk ip is 172.20.1.1 and my 
> router is 172.20.1.254.
>
> thank you for help
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] i have a probleme and my asterisk and ovh

[Danny Nicholas]

Output of CLI sip show peer ovh?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Sent: Tuesday, April 07, 2009 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] i have a probleme and my asterisk and ovh

[ovh]
type=peer
secret=
username=0033972xx
fromuser=0033972xx
host=sip.ovh.net
canreinvite=no
disallow=all
allow=g729
tos_sip=1; Sets TOS for SIP packets.
tos_audio=1   ; Sets TOS for RTP audio packets.
tos_video=1
dtmfmode=rfc28335
relaxdtmf=yes
nat=yes
qualify=yes
insecure=port,invite
context=entrant-ovh

thank you.

Danny Nicholas a écrit :
> Show us your sip.conf
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
> Sent: Tuesday, April 07, 2009 2:54 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] i have a probleme and my asterisk and ovh
>
> hello every body
>
> my connexion on ovh to pass in UNREACHABLE and not reidentified were not 
> reboot the server.
>
> [Apr  7 20:17:21] NOTICE[19947]: chan_sip.c:15605 
> handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
> [Apr  7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswer: 
> Peer 'ovh' is now UNREACHABLE!  Last qualify: 2067
>
> but my probleme is the adress ip 172.25.3.51 is not my adress.
>
> Really destroying SIP dialog 
> '13ff06ae3e4bb3bf04987f5f5b497...@172.20.1.1' Method: OPTIONS
> Really destroying SIP dialog 
> '6eac266b68dbc2566209fbb74aec7...@172.25.3.51' Method: REGISTER
>
> I do not know where she comes out, my asterisk ip is 172.20.1.1 and my 
> router is 172.20.1.254.
>
> thank you for help
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] i have a probleme and my asterisk and ovh

[Henry]

[ovh]
type=peer
secret=
username=0033972xx
fromuser=0033972xx
host=sip.ovh.net
canreinvite=no
disallow=all
allow=g729
tos_sip=1; Sets TOS for SIP packets.
tos_audio=1   ; Sets TOS for RTP audio packets.
tos_video=1
dtmfmode=rfc28335
relaxdtmf=yes
nat=yes
qualify=yes
insecure=port,invite
context=entrant-ovh

thank you.

Danny Nicholas a écrit :
> Show us your sip.conf
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
> Sent: Tuesday, April 07, 2009 2:54 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] i have a probleme and my asterisk and ovh
>
> hello every body
>
> my connexion on ovh to pass in UNREACHABLE and not reidentified were not 
> reboot the server.
>
> [Apr  7 20:17:21] NOTICE[19947]: chan_sip.c:15605 
> handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
> [Apr  7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswer: 
> Peer 'ovh' is now UNREACHABLE!  Last qualify: 2067
>
> but my probleme is the adress ip 172.25.3.51 is not my adress.
>
> Really destroying SIP dialog 
> '13ff06ae3e4bb3bf04987f5f5b497...@172.20.1.1' Method: OPTIONS
> Really destroying SIP dialog 
> '6eac266b68dbc2566209fbb74aec7...@172.25.3.51' Method: REGISTER
>
> I do not know where she comes out, my asterisk ip is 172.20.1.1 and my 
> router is 172.20.1.254.
>
> thank you for help
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


___
-- 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

Re: [asterisk-users] Grandstream blind transfer issue

[Klaus Darilion]

Max Alex wrote:
> Hi All,
> I have working asterisk version 1.4.24.
> I have a blind transfer issue with grandstream bt200.

Does it work with other phones? That means is it a Grandstream isue or a 
general issue?

> I have updated the latest firmware to the phone.
> The phone is sending the *refer* to asterisk but asterisk is not able to 
> connect with the another call

Why? some log messages would help us helping you.

> that i have checked in sip debug.
> I am using transfer button of the grandstream phone.
> Can anybody provide help for this issue?

Please ask again on the user mailing lists and provide some log messages

> Thanks in advance!!
>  
> Thanks,
> Max Alex
> Voip Developer
> 
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] i have a probleme and my asterisk and ovh

[Danny Nicholas]

Show us your sip.conf

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Sent: Tuesday, April 07, 2009 2:54 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] i have a probleme and my asterisk and ovh

hello every body

my connexion on ovh to pass in UNREACHABLE and not reidentified were not 
reboot the server.

[Apr  7 20:17:21] NOTICE[19947]: chan_sip.c:15605 
handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
[Apr  7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswer: 
Peer 'ovh' is now UNREACHABLE!  Last qualify: 2067

but my probleme is the adress ip 172.25.3.51 is not my adress.

Really destroying SIP dialog 
'13ff06ae3e4bb3bf04987f5f5b497...@172.20.1.1' Method: OPTIONS
Really destroying SIP dialog 
'6eac266b68dbc2566209fbb74aec7...@172.25.3.51' Method: REGISTER

I do not know where she comes out, my asterisk ip is 172.20.1.1 and my 
router is 172.20.1.254.

thank you for help

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] i have a probleme and my asterisk and ovh

[Henry]

hello every body

my connexion on ovh to pass in UNREACHABLE and not reidentified were not 
reboot the server.

[Apr  7 20:17:21] NOTICE[19947]: chan_sip.c:15605 
handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
[Apr  7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswer: 
Peer 'ovh' is now UNREACHABLE!  Last qualify: 2067

but my probleme is the adress ip 172.25.3.51 is not my adress.

Really destroying SIP dialog 
'13ff06ae3e4bb3bf04987f5f5b497...@172.20.1.1' Method: OPTIONS
Really destroying SIP dialog 
'6eac266b68dbc2566209fbb74aec7...@172.25.3.51' Method: REGISTER

I do not know where she comes out, my asterisk ip is 172.20.1.1 and my 
router is 172.20.1.254.

thank you for help

___
-- 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

[asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands?

[Giovanny Magallanes]

-- Forwarded message --
From: Juan Carlos Huerta 
Date: 07-abr-2009 13:41
Subject: Re: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel
commands?
To: asterisk...@lists.digium.com

Please wirte to asterisk-users@lists.digium.com to get help about this
problem.

Juan Carlos
~ Lo que no te mata te fortalece ~



On Tue, Apr 7, 2009 at 1:35 PM, Giovanny Magallanes
 wrote:
> Thanks, Juan  Carlos.
>
> Yes, I still can t type the commands in the asterisk console . I dont have
> dial tone for FXS ports .
>
>
> Regards.
>
> Giovanni.
>
>
> 2009/4/7, Juan Carlos Huerta :
>>
>> And what is the result? still with zap commands problem?
>>
>> Juan Carlos
>> ~ Lo que no te mata te fortalece ~
>>
>>
>>
>> On Tue, Apr 7, 2009 at 12:49 PM, Giovanny Magallanes
>>  wrote:
>> > I did it:
>> >
>> > elastix*CLI> load chan_zap.so
>> > The 'load' command is deprecated and will be removed in a future
>> > release.
>> > Please use 'module load' instead.
>> >   == Parsing '/etc/asterisk/zapata.conf': Found
>> >   == Parsing '/etc/asterisk/zapata_additional.conf': Found
>> >   == Parsing '/etc/asterisk/zapata-channels.conf': Found
>> > -- Registered channel 63, FXO Kewlstart signalling
>> > -- Registered channel 64, FXO Kewlstart signalling
>> > -- Registered channel 65, FXO Kewlstart signalling
>> > -- Registered channel 66, FXO Kewlstart signalling
>> > -- Registered channel 67, FXO Kewlstart signalling
>> > -- Registered channel 68, FXO Kewlstart signalling
>> > -- Registered channel 68, FXO Kewlstart signalling
>> > -- Registered channel 69, FXO Kewlstart signalling
>> > -- Registered channel 70, FXO Kewlstart signalling
>> > -- Registered channel 71, FXO Kewlstart signalling
>> > -- Registered channel 72, FXO Kewlstart signalling
>> > -- Registered channel 73, FXO Kewlstart signalling
>> > -- Registered channel 74, FXO Kewlstart signalling
>> > -- Registered channel 75, FXO Kewlstart signalling
>> > -- Registered channel 76, FXO Kewlstart signalling
>> > -- Registered channel 77, FXO Kewlstart signalling
>> > -- Registered channel 78, FXO Kewlstart signalling
>> > elastix*CLI>
>> >
>> > Thank you Juan
>> >
>> > Giovanni
>> >
>> >
>> > 2009/4/7, Juan Carlos Huerta :
>> >>
>> >> Try to load the chan_zap.so module manually to see if you get some
>> >> error.
>> >>
>> >> Juan Carlos
>> >> ~ Lo que no te mata te fortalece ~
>> >>
>> >>
>> >>
>> >> On Tue, Apr 7, 2009 at 12:44 PM, Giovanny Magallanes
>> >>  wrote:
>> >> > I'm using Elastix 1.1-8 with:
>> >> >
>> >> > asterisk-1.4.19
>> >> > spandsp-0.0.4
>> >> > unicall-0.0.5pre1
>> >> > zaptel-1.4.9.2
>> >> > unicall-0.0.5pre1
>> >> > libmfcr2-0.0.3
>> >> > libsupertone-0.0.2
>> >> > libunicall-0.0.3
>> >> >
>> >> > Regards,
>> >> >
>> >> > Giovanni
>> >> >
>> >> > 2009/4/7, Giovanny Magallanes :
>> >> >>
>> >> >> Ok, Thanks. The chan_zap commands are unavailable, but unicall
>> >> >> commands
>> >> >> and my E1 MFC/R2 are OK.
>> >> >>
>> >> >> Giovanni
>> >> >>
>> >> >>
>> >> >> 2009/4/7, Moises Silva :
>> >> >>>
>> >> >>> I don't understand your problem. And no, unicall has nothing to do
>> >> >>> with chan_zap.so commands.
>> >> >>>
>> >> >>> Please, in the future, don't hijack threads, open a new thread for
>> >> >>> your discussion. This time I changed the subject already.
>> >> >>>
>> >> >>> Moy
>> >> >>>
>> >> >>> On Tue, Apr 7, 2009 at 1:08 AM, Giovanny Magallanes
>> >> >>>  wrote:
>> >> >>> > Hi, Guys.
>> >> >>> >
>> >> >>> > I did not type any zaptel commands in asterisk console. I have
>> >> >>> > installed
>> >> >>> > Elastix 1.1-8 with unicall and it is working fine with a Digium
>> >> >>> > TE212-P.
>> >> >>> >
>> >> >>> > [r...@elastix asterisk]# find -P / -name chan_zap.so
>> >> >>> > /usr/lib/asterisk/modules/chan_zap.so
>> >> >>> >
>> >> >>> >
/usr/src/astunicall-1.4.19-0.1/asterisk-1.4.19/channels/chan_zap.so
>> >> >>> > [r...@elastix asterisk]#
>> >> >>> > elastix*CLI> load chan_zap.so
>> >> >>> >   == Parsing '/etc/asterisk/zapata.conf': Found
>> >> >>> >   == Parsing '/etc/asterisk/zapata_additional.conf': Found
>> >> >>> >   == Parsing '/etc/asterisk/zapata-channels.conf': Found
>> >> >>> > -- Registered channel 63, FXO Kewlstart signalling
>> >> >>> > -- Registered channel 64, FXO Kewlstart signalling
>> >> >>> > -- Registered channel 65, FXO Kewlstart signalling
>> >> >>> > -- Registered channel 66, FXO Kewlstart signalling
>> >> >>> >
>> >> >>> >
>> >> >>> > Can unicall affect zaptel commands in asterisk??
>> >> >>> >
>> >> >>> > Thanks
>> >> >>> >
>> >> >>> > Giovanni Magallanes
>> >> >>> >
>> >> >>> >
>> >> >>> >
>> >> >>> > 2009/4/6, Fernando Berretta :
>> >> >>> >>
>> >> >>> >> Hi,
>> >> >>> >>
>> >> >>> >> I've installed Asterisk 1.4 and Openr2 1.1 all is working well.
>> >> >>> >> I've
>> >> >>> >> connected a crossover cable to TE122 board and board lite has
>> >> >>> >> went
>> >> >>> >> green. But when I've  tried to make a call, I've got t

Re: [asterisk-users] is shared_lastcall available in 1.4

[Gabriel Ortiz Lour]

I've just backported it to asterisk 1.4.19,
the patch is atached


2008/8/27 Bob Pierce 

>
> On Wed, 2008-08-27 at 11:21 +0300, Atis Lezdins wrote:
> > If you doubt about some part, you're welcome to ask, i'll try to help
> > you, but i don't want to provide complete backport to you, as i won't
> > be able to test it :)
>
> Thanks Atis,
>
> I'll probably try this in a few weeks when I start rebuilding the
> permanent system that will replace our current temporary system.
> That should give us the opportunity to test it on the bench instead of
> playing around with the production box.
>
> I'll probably be back to ask for help.
>
> Have a great day,
> Bob
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--- app_queue-orig.c	2009-04-07 15:10:12.256746320 -0300
+++ app_queue.c	2009-04-07 15:14:54.602271840 -0300
@@ -272,6 +272,9 @@
 /*! \brief queues.conf [general] option */
 static int montype_default = 0;
 
+/*! \brief queues.conf [general] option */
+static int shared_lastcall = 0;
+
 enum queue_result {
 	QUEUE_UNKNOWN = 0,
 	QUEUE_TIMEOUT = 1,
@@ -315,6 +318,7 @@
 	int metric;
 	int oldstatus;
 	time_t lastcall;
+	struct call_queue *lastqueue;
 	struct member *member;
 };
 
@@ -352,6 +356,7 @@
 	int status; /*!< Status of queue member */
 	int paused; /*!< Are we paused (not accepting calls)? */
 	time_t lastcall;/*!< When last successful call was hungup */
+	struct call_queue *lastqueue;   /*!< Last queue we received a call */
 	unsigned int dead:1;/*!< Used to detect members deleted in realtime */
 	unsigned int delme:1;   /*!< Flag to delete entry on reload */
 };
@@ -795,7 +800,7 @@
 static int member_cmp_fn(void *obj1, void *obj2, int flags)
 {
 	struct member *mem1 = obj1, *mem2 = obj2;
-	return strcmp(mem1->interface, mem2->interface) ? 0 : CMP_MATCH;
+	return strcasecmp(mem1->interface, mem2->interface) ? 0 : CMP_MATCH;
 }
 
 static void init_queue(struct call_queue *q)
@@ -1783,9 +1788,10 @@
 	char *location;
 
 	/* on entry here, we know that tmp->chan == NULL */
-	if (qe->parent->wrapuptime && (time(NULL) - tmp->lastcall < qe->parent->wrapuptime)) {
+	if ((tmp->lastqueue && tmp->lastqueue->wrapuptime && (time(NULL) - tmp->lastcall < tmp->lastqueue->wrapuptime)) ||
+		(!tmp->lastqueue && qe->parent->wrapuptime && (time(NULL) - tmp->lastcall < qe->parent->wrapuptime))) {
 		if (option_debug)
-			ast_log(LOG_DEBUG, "Wrapuptime not yet expired for %s\n", tmp->interface);
+			ast_log(LOG_DEBUG, "Wrapuptime not yet expired on queue %s for %s\n", (tmp->lastqueue? tmp->lastqueue->name : qe->parent->name), tmp->interface);
 		if (qe->chan->cdr)
 			ast_cdr_busy(qe->chan->cdr);
 		tmp->stillgoing = 0;
@@ -2461,12 +2467,33 @@
 
 static int update_queue(struct call_queue *q, struct member *member, int callcompletedinsl)
 {
+	struct member *mem;
+	struct call_queue *qtmp;
+	
+	if (shared_lastcall) {
+		AST_LIST_LOCK(&queues);
+		AST_LIST_TRAVERSE(&queues, qtmp, list) {
+			ast_mutex_lock(&qtmp->lock);
+			if ((mem = ao2_find(qtmp->members, member, OBJ_POINTER))) {
+time(&mem->lastcall);
+mem->calls++;
+mem->lastqueue = q;
+ao2_ref(mem, -1);
+			}
+			ast_mutex_unlock(&qtmp->lock);
+		}
+		AST_LIST_UNLOCK(&queues);
+	}
+
 	ast_mutex_lock(&q->lock);
-	time(&member->lastcall);
-	member->calls++;
 	q->callscompleted++;
 	if (callcompletedinsl)
 		q->callscompletedinsl++;
+	if (!shared_lastcall) {
+		time(&member->lastcall);
+		member->calls++;
+		member->lastqueue = q;
+	}
 	ast_mutex_unlock(&q->lock);
 	return 0;
 }
@@ -2726,6 +2753,7 @@
 		tmp->member = cur;
 		tmp->oldstatus = cur->status;
 		tmp->lastcall = cur->lastcall;
+		tmp->lastqueue = cur->lastqueue;
 		ast_copy_string(tmp->interface, cur->interface, sizeof(tmp->interface));
 		/* Special case: If we ring everyone, go ahead and ring them, otherwise
 		   just calculate their metric for the appropriate strategy */
@@ -2870,7 +2898,7 @@
 pbx_builtin_setvar_helper(qe->chan, "MEMBERINTERFACE", member->interface);
 
 		/* Begin Monitoring */
-		if (qe->parent->monfmt && *qe->parent->monfmt) {
+		if (qe->parent->monfmt && strlen(qe->parent->monfmt) > 2) {
 			if (!qe->parent->montype) {
 if (option_debug)
 	ast_log(LOG_DEBUG, "Starting Monitor as requested.\n");
@@ -4146,6 +4174,9 @@
 			if ((general_val = ast_variable_retrieve(cfg, "general", "monitor-type")))
 if (!strcasecmp(general_val, "mixmonitor"))
 	montype_default = 1;
+			shared_lastcall = 0;
+			if ((general_val = ast_variable_retrieve(cfg, "general", "shared_lastcall")))
+shared_lastcall = ast_true(general_val);
 		} else {	/* Define queue */
 			/* Look f

Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal

[Mark Michelson]

Philipp Kempgen wrote:
> Olivier schrieb:
>> 2009/4/7 Philipp Kempgen 
>>> Olivier schrieb:
 I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
>>> an
 AEL2 file like this :
 SendText(${BASE64_DECODE(DQ==)});


 Value sent (8 bytes long) is very strange :
 Content-Type: text/plain;charset=UTF-8
 Content-Length: 8

 �ez?==
>>> I doubt you will find a way to properly pass CR or LF to an
>>> application in extensions.(conf|ael) but keep us in the loop.
> 
>> It's strange how such a silly thing is somehow keeping me from centrally
>> managing phones forwarding : I can display a phone is forwarded but I can't
>> gracefully return to previous status ...
> 
> BTW (developer's question) is there a reason why SendText() resp.
> sendtext_exec() refuses to send zero-length data?
> 

I can't point to any specific reason. I assume that whoever wrote the 
application probably thought that attempting to send zero-length data was 
pointless and that if no data were passed to the application, it likely was due 
to an error by the user.

Mark Michelson

___
-- 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

Re: [asterisk-users] Grandstream blind transfer issue

[Gordon Henderson]

On Tue, 7 Apr 2009, Max Alex wrote:

> Hi All,
> I have working asterisk version 1.4.24.
> I have a blind transfer issue with grandstream bt200.
> I have updated the latest firmware to the phone.
> The phone is sending the *refer* to asterisk but asterisk is not able to
> connect with the another call
> that i have checked in sip debug.
> I am using transfer button of the grandstream phone.
> Can anybody provide help for this issue?
> Thanks in advance!!

How are you doing the entire transfer operation?

For blind transfers, I do:

  Push Transfer
 (caller is now on hold, you get a new dial-tone)
   dial extension and push SEND

At this point, called phone rings and caller is immediately taken off hold 
and transfered to the new ringing phone... you can hang up at that point.

Don't use the 'flash' key.

I have many BT200's and GXP280's out there - this seems to work for them 
without any issues. Asterisk 1.2 though.


Gordon



___
-- 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

Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal

[Philipp Kempgen]

Olivier schrieb:
> 2009/4/7 Philipp Kempgen 
>> Olivier schrieb:
>> > I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
>> an
>> > AEL2 file like this :
>> > SendText(${BASE64_DECODE(DQ==)});
>> >
>> >
>> > Value sent (8 bytes long) is very strange :
>> > Content-Type: text/plain;charset=UTF-8
>> > Content-Length: 8
>> >
>> > �ez?==
>>
>> I doubt you will find a way to properly pass CR or LF to an
>> application in extensions.(conf|ael) but keep us in the loop.

> It's strange how such a silly thing is somehow keeping me from centrally
> managing phones forwarding : I can display a phone is forwarded but I can't
> gracefully return to previous status ...

BTW (developer's question) is there a reason why SendText() resp.
sendtext_exec() refuses to send zero-length data?


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

Re: [asterisk-users] async agi question

[Moises Silva]

It's a bug in the Async AGI feature. I have created a new patch
http://www.moythreads.com/asterisk-1.4.18-async-agi.patch

Please test it and let me know if it works for you,

Moy

On Tue, Apr 7, 2009 at 11:50 AM, Moises Silva  wrote:
> "Released" means no patching needed, it means it was tested and put
> into Asterisk tree. So, I published a patch for 1.4 so it could be
> used in 1.4 however the feature per se was just released for Asterisk
> 1.6.
>
> Moy
>
> On Tue, Apr 7, 2009 at 10:01 AM,   wrote:
>> Moy,
>> I apologize if you felt under some pressure. It wasn't my mind. I only 
>> wanted to know if either there was a mistake in my configuration, or I was 
>> failing in the procedure, or it was a bug, as you said, in order to move 
>> forward.
>>
>> By the way, there's a thing I don't understand:
>>
>> In your blog at 
>> http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ , 
>> talking about AsyncAGI, there's an entry where you say:
>>
>> "If you want the patch back-ported to 1.4, I have one here: 
>> http://www.moythreads.com/asterisk-1.4.15-async-agi.patch";
>>
>> However, in an earlier answer to this thread, you said:
>>
>> "Async AGI was never released for Asterisk 1.4.X, so probably the patch
>> you used has a bug or something ..."
>>
>> Does it mean you are talking about different things? Can you clarify, please?
>>
>> Many thanks
>> Jose Arias
>>
>>
>>
>> --
>> This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com
>> http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11864934.html
>>
>> ___
>> -- 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
>>
>
>
>
> --
> "I do not agree with what you have to say, but I’ll defend to the
> death your right to say it." Voltaire
>



-- 
"I do not agree with what you have to say, but I’ll defend to the
death your right to say it." Voltaire

___
-- 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

Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal

[Olivier]

2009/4/7 Philipp Kempgen 

> Olivier schrieb:
> > I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
> an
> > AEL2 file like this :
> > SendText(${BASE64_DECODE(DQ==)});
> >
> >
> > Value sent (8 bytes long) is very strange :
> > Content-Type: text/plain;charset=UTF-8
> > Content-Length: 8
> >
> > �ez?==
>
> I doubt you will find a way to properly pass CR or LF to an
> application in extensions.(conf|ael) but keep us in the loop.


I broke my teeth on that all day long !
I was about to try AGI ...

It's strange how such a silly thing is somehow keeping me from centrally
managing phones forwarding : I can display a phone is forwarded but I can't
gracefully return to previous status ...


>
>
>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
___
-- 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

Re: [asterisk-users] AEL2, BASE64_DECODE and hexadecimal

[Philipp Kempgen]

Olivier schrieb:
> I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in an
> AEL2 file like this :
> SendText(${BASE64_DECODE(DQ==)});
> 
> 
> Value sent (8 bytes long) is very strange :
> Content-Type: text/plain;charset=UTF-8
> Content-Length: 8
> 
> �ez?==

I doubt you will find a way to properly pass CR or LF to an
application in extensions.(conf|ael) but keep us in the loop.


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

Re: [asterisk-users] app_backticks and 1.6

[Philipp Kempgen]

Olivier schrieb:

> Is there any app_backticks
> equivalent or
> workaround for 1.6 ?

SHELL()
http://www.das-asterisk-buch.de/2.1/functions-shell.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
-- 

___
-- 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

Re: [asterisk-users] Hacked

[Tilghman Lesher]

On Monday 06 April 2009 19:22:30 Martin wrote:
> Can you give more information about this vulnerability ?

It's unlikely that it's this vulnerability.  Every Asterisk box allows guest
access to the machine, by default.  The context it goes to is generally
the "default" context.  This is what allows you to publish an addresses
like sip:foo.example.com and have it get through to your company.  There's
no preexisting relationship between caller and callee; it's merely a method
of contacting the machine.

What is a vulnerability is the way that some people have configured this.
They put in that context patterns that can dial out.  There's nothing
specifically wrong with this configuration, from an Asterisk perspective;
however, in many cases, guest access is not what the administrator intended;
thus, the machine may be used to make illicit outbound telephone calls by
anybody who sends a SIP call to that machine.

The recent vulnerability had nothing to do with this, but with the ability of
an attacker to scan a SIP server for legitimate usernames and passwords.
This, by the way, merely took advantage of the SIP protocol, as written.
Normally, SIP allows you to differentiate between invalid usernames (404) and
invalid passwords (403).  What we closed in the recent vulnerability patch was
to allow administrators to send back 403, regardless of whether the username
existed or not.

-- 
Tilghman

___
-- 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

Re: [asterisk-users] Using multiple 'peer' identities on one phone with 1.4

[Florian Hackenberger]

On Tuesday 07 April 2009, Olle E. Johansson wrote:
> I don't see any problems there. YOu still have devices with states,
> as you would have with authentication. Of course, it still depends on
> your configuration. But authentication should not affect states.
Ok, thanks for that, I'll have a look at openSER.

> If you use the limitonpeer setting, all states for both the user and
> the peer part of a friend will only be handled by the peer, which is
> the device watched for subscriptions. 
That worked like a charm, thanks!

> There was recently also an 
> overhaul of the states for queues, with a patch to 1.4 that made it
> possible
> to build a stronger relationship between a queue member and
> a state object.
Could you please point me to a bug report or an SVN revision?

Thanks a lot for you help!
Florian

-- 
DI Florian Hackenberger
flor...@hackenberger.at
www.hackenberger.at

___
-- 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

Re: [asterisk-users] Best Practice Advice?

[Darrick Hartman]

Gabriel - IP Guys wrote:
> Dear All,
> 
>  
> 
> I have a asterisk setup that is currently running on version 1.4.15 – I 
> wish to upgrade or migrate this instance to the current asterisk stable, 
>  1.6.0.6. It is my intention to build a FC8 box, then install asterisk 
> from source, and begin to migrate over the configuration. The thing is, 
> this sounds so simple in my head, and I’ve had enough issues with 
> asterisk, to know that life isn’t simple!
> 
>  
> 
> What I plan to do, is to copy the old configuration over to a box 
> running FC8 – and then compile and run asterisk 1.4.15 – and gradually 
> upgrade it, until I reach 1.6.0.6 – Any input on this matter will be 
> appreciated. Thank you

You might reconsider Fedora8.  It's support life is gotta be nearing the 
end.

http://fedoraproject.org/wiki/LifeCycle

CentOS is generally a better choice for something you want to have in 
service for any length of time.

Darrick

___
-- 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

[asterisk-users] Best Practice Advice?

[Gabriel - IP Guys]

Dear All,

 

I have a asterisk setup that is currently running on version 1.4.15 - I
wish to upgrade or migrate this instance to the current asterisk stable,
1.6.0.6. It is my intention to build a FC8 box, then install asterisk
from source, and begin to migrate over the configuration. The thing is,
this sounds so simple in my head, and I've had enough issues with
asterisk, to know that life isn't simple!

 

What I plan to do, is to copy the old configuration over to a box
running FC8 - and then compile and run asterisk 1.4.15 - and gradually
upgrade it, until I reach 1.6.0.6 - Any input on this matter will be
appreciated. Thank you

 

---

Mr Gabriel

 

___
-- 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

Re: [asterisk-users] async agi question

[Moises Silva]

"Released" means no patching needed, it means it was tested and put
into Asterisk tree. So, I published a patch for 1.4 so it could be
used in 1.4 however the feature per se was just released for Asterisk
1.6.

Moy

On Tue, Apr 7, 2009 at 10:01 AM,   wrote:
> Moy,
> I apologize if you felt under some pressure. It wasn't my mind. I only wanted 
> to know if either there was a mistake in my configuration, or I was failing 
> in the procedure, or it was a bug, as you said, in order to move forward.
>
> By the way, there's a thing I don't understand:
>
> In your blog at 
> http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ , 
> talking about AsyncAGI, there's an entry where you say:
>
> "If you want the patch back-ported to 1.4, I have one here: 
> http://www.moythreads.com/asterisk-1.4.15-async-agi.patch";
>
> However, in an earlier answer to this thread, you said:
>
> "Async AGI was never released for Asterisk 1.4.X, so probably the patch
> you used has a bug or something ..."
>
> Does it mean you are talking about different things? Can you clarify, please?
>
> Many thanks
> Jose Arias
>
>
>
> --
> This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com
> http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11864934.html
>
> ___
> -- 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
>



-- 
"I do not agree with what you have to say, but I’ll defend to the
death your right to say it." Voltaire

___
-- 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

Re: [asterisk-users] immediate=yes is populated to all channels

[Tzafrir Cohen]

On Tue, Apr 07, 2009 at 04:28:50PM +0200, Loic Didelot wrote:
> Hello,
> I connected a twinbus system to a xorcom fxs port. I have to set
> immediate=yes to make thins work as expected. And it works.
> 
> The problem is that the parameter immediate=yes seems to applied to
> every port, also my PRI port. This means that extensions are no longer
> working for incoming calls.
> 
> Is this a known issue?

Yes.

Fix?

with users.conf you can use separate sections for channels and not have
to worry too much about the order.

Chan_dahdi supports that syntax as well as of 1.6.1 .

-- 
   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

___
-- 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

[asterisk-users] AEL2, BASE64_DECODE and hexadecimal

[Olivier]

Hi,

I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in an
AEL2 file like this :
SendText(${BASE64_DECODE(DQ==)});


Value sent (8 bytes long) is very strange :
Content-Type: text/plain;charset=UTF-8
Content-Length: 8

�ez?==


Any workaround ?

Regards
___
-- 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

Re: [asterisk-users] immediate=yes is populated to all channels

[Loic Didelot]

Hi,
I just figued it out with some help in the irc channel. I just need to
set it back to no for the other channels.

Thanks,
Loic.

On Tue, 2009-04-07 at 16:28 +0200, Loic Didelot wrote:
> Hello,
> I connected a twinbus system to a xorcom fxs port. I have to set
> immediate=yes to make thins work as expected. And it works.
> 
> The problem is that the parameter immediate=yes seems to applied to
> every port, also my PRI port. This means that extensions are no longer
> working for incoming calls.
> 
> Is this a known issue?
> 
> I am using asterisk 1.4.22  and here is my zapata.conf
> 
> [channels]
> language=en
> switchtype=euroisdn
> 
> signalling = fxo_ks
> context=OUTGOING-pbx
> faxdetect=no
> accountcode=98
> immediate=yes
> dtmfmode=auto
> musiconhold=default
> subscribecontext=BLF-shrmpbx1
> call-limit=3
> mailbox...@shrmpbx1
> vmexten=898910
> channel => 60
> 
> signalling = pri_cpe
> context=ZAP-pri
> group=1
> faxdetect=no
> accountcode=carrier-1-pri
> channel => 1
> channel => 2
> channel => 3
> channel => 4
> channel => 5
> channel => 6
> channel => 7
> channel => 8
> channel => 9
> channel => 10
> channel => 11
> channel => 12
> channel => 13
> channel => 14
> channel => 15
> 
> 
> 
> Best regards,
> Loïc Didelot.
> 
> 
> 
> 
> ___
> -- 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
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk and patton

[Olivier]

2009/4/7 mahboob zaman 

>
> Hellow
>
> Can any body helps how can interfacing between asterisk and patton media
> getway.
>
Which smartware version ?

>
> Thanks
> mahboob
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] immediate=yes is populated to all channels

[Loic Didelot]

Hello,
I connected a twinbus system to a xorcom fxs port. I have to set
immediate=yes to make thins work as expected. And it works.

The problem is that the parameter immediate=yes seems to applied to
every port, also my PRI port. This means that extensions are no longer
working for incoming calls.

Is this a known issue?

I am using asterisk 1.4.22  and here is my zapata.conf

[channels]
language=en
switchtype=euroisdn

signalling = fxo_ks
context=OUTGOING-pbx
faxdetect=no
accountcode=98
immediate=yes
dtmfmode=auto
musiconhold=default
subscribecontext=BLF-shrmpbx1
call-limit=3
mailbox...@shrmpbx1
vmexten=898910
channel => 60

signalling = pri_cpe
context=ZAP-pri
group=1
faxdetect=no
accountcode=carrier-1-pri
channel => 1
channel => 2
channel => 3
channel => 4
channel => 5
channel => 6
channel => 7
channel => 8
channel => 9
channel => 10
channel => 11
channel => 12
channel => 13
channel => 14
channel => 15



Best regards,
Loïc Didelot.




___
-- 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

[asterisk-users] Grandstream blind transfer issue

[Max Alex]

Hi All,
I have working asterisk version 1.4.24.
I have a blind transfer issue with grandstream bt200.
I have updated the latest firmware to the phone.
The phone is sending the *refer* to asterisk but asterisk is not able to
connect with the another call
that i have checked in sip debug.
I am using transfer button of the grandstream phone.
Can anybody provide help for this issue?
Thanks in advance!!

Thanks,
Max Alex
Voip Developer
___
-- 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

Re: [asterisk-users] async agi question

[cyr2242]

Moy,
I apologize if you felt under some pressure. It wasn't my mind. I only wanted 
to know if either there was a mistake in my configuration, or I was failing in 
the procedure, or it was a bug, as you said, in order to move forward.

By the way, there's a thing I don't understand:

In your blog at 
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ , 
talking about AsyncAGI, there's an entry where you say:

"If you want the patch back-ported to 1.4, I have one here: 
http://www.moythreads.com/asterisk-1.4.15-async-agi.patch";

However, in an earlier answer to this thread, you said:

"Async AGI was never released for Asterisk 1.4.X, so probably the patch
you used has a bug or something ..."

Does it mean you are talking about different things? Can you clarify, please?

Many thanks
Jose Arias 



--
This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com
http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11864934.html

___
-- 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

[asterisk-users] Zaptel connectivity issues

[Danny Nicholas]

Greetings listers,

 I've posted this at least once previously, but
thought I'd try again.  I've got a TDM410P card on Asterisk 1.4.21.2 and
experience these two problems.

 

1.  When placing an outgoing call,  I get no audio until Asterisk
bridges the connection (2-15 second delay).  I can Answer before Dialing,
but this give me a incorrect CDR and no way of knowing that the other end
did not answer.
2.  When calling an automated number such as AT&T Teleconference, I
never get an indication of connection from the receiver, though I know they
answered because I hear them.

 

Any thoughts or suggestions.

Danny Nicholas

___
-- 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

Re: [asterisk-users] Logging Asterisk console

[Marco Sambo]

Hi Enrico,
I do that by modifying logger.conf

[logfiles]
logpro => notice,warning,error,debug,verbose

and modifying asterisk.conf

[directories]
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

[options]
verbose = 3

and so I find into /var/log/asterisk the logpro file with the output of CLI
(verbose) and notice, warning, error, debug message of Asterisk.


Ciao
Marco


2009/4/7 Enrico Pasqualotto 

> Hi all, in witch way can I put in a log file the asterisk console?
> I have tried with some settings in file logger.conf but the log not
> contain the same debug that I can see with "asterisk -rvvv".
> I need it in debugging purpose for tracking some bug.
>
> Thanks Enrico.
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way AUDIO

[Danny Nicholas]

Here's my .02 - local lan is probably behind a firewall meaning that the
5060 gets out ok to send your audio, but the 1-2 range that the
other side comes in on is blocked.  You don't have the problem with static
WAN because it is not behind the firewall or has more ports open.  Do a
netstat -an during each call and see what is different.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Monday, April 06, 2009 6:04 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] One way AUDIO

 

Few Running figures !!

On Tue, Apr 7, 2009 at 3:41 AM, David @ULC  wrote:

 

I have a server with 2 Lan Cards. 

Now, when I am trying to make calls using Local Lan, its One way Audio which
means customer cant hear me but if I use Static IP with Wan Connection, it
works perfectly. 

I changed the network from loc1 to loc2 but its same. 

I tried changing Ethernet Card but no use. 

What could be the Issue ?

 

___
-- 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

Re: [asterisk-users] Logging Asterisk console

[Danny Nicholas]

Here is a suggestion from the Digium Bug site
asterisk -cvvvgn | tee /tmp/my_log_file.txt

http://bugs.digium.com/view.php?id=14255

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Enrico
Pasqualotto
Sent: Tuesday, April 07, 2009 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Logging Asterisk console

Hi all, in witch way can I put in a log file the asterisk console?
I have tried with some settings in file logger.conf but the log not
contain the same debug that I can see with "asterisk -rvvv".
I need it in debugging purpose for tracking some bug.

Thanks Enrico.


___
-- 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

Re: [asterisk-users] Logging Asterisk console

[Giancarlo Rubio]

2009/4/7 Enrico Pasqualotto :
> Hi all, in witch way can I put in a log file the asterisk console?
> I have tried with some settings in file logger.conf but the log not
> contain the same debug that I can see with "asterisk -rvvv".
> I need it in debugging purpose for tracking some bug.

asterisk -rvc |tee /tmp/my_log.txt

>
> Thanks Enrico.
>
> ___
> -- 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
>



-- 
Giancarlo Rubio

___
-- 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

[asterisk-users] Logging Asterisk console

[Enrico Pasqualotto]

Hi all, in witch way can I put in a log file the asterisk console?
I have tried with some settings in file logger.conf but the log not
contain the same debug that I can see with "asterisk -rvvv".
I need it in debugging purpose for tracking some bug.

Thanks Enrico.


smime.p7s
Description: S/MIME cryptographic signature
___
-- 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

Re: [asterisk-users] [OT] Re: async agi question

[Philipp Kempgen]

cyr2...@gmail.com schrieb:
> I'm sorry but I can't find where to change this one in the opensubscriber 
> service. I'm sending a request to them for it. As soon as I get the answer 
> I''ll do it.

Never mind. I just couldn't resisit. :-)

> -- Philipp Kempgen wrote : 
> cyr2...@gmai... schrieb:
>> This message was sent on behalf of cyr2...@gmai... at openSubscriber.com
> 
> Use the appropriate header field for that information.
> It's called "From" (in contrast to "Sender").


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

Re: [asterisk-users] Using multiple 'peer' identities on one phone with 1.4

[Olle E. Johansson]

7 apr 2009 kl. 11.49 skrev Florian Hackenberger:

> On Tuesday 07 April 2009, Olle E. Johansson wrote:
>> Well, you can have OpenSER doing the authentication and turn it
>> off in Asterisk, but still match a device.
> Ok, but what about sip device state? Will that work? Will asterisk
> report the device as busy when the sip device is engaged in a call?
I don't see any problems there. YOu still have devices with states,
as you would have with authentication. Of course, it still depends on
your configuration. But authentication should not affect states.

>
>> Why not use type=user if you have the need for multiple accounts
>> on the same IP? Using type=peer for incoming call matching is
>> when you can't match on from: but have to match on IP.
> Because call limits for queues don't work when using type=friend. If  
> the
> user gets a call through a queue, another queue won't send a call to
> the user, but if the user places a call, asterisk does not consider  
> the
> user busy and tries to forward queue calls to that user. Are there
> workarounds for that?

Depends on the version you are using. In 1.0 it was a nightmare,
in 1.2 it was a tiny bit better and in 1.4 we had a lot of improvements.

If you use the limitonpeer setting, all states for both the user and the
peer part of a friend will only be handled by the peer, which is the
device watched for subscriptions. There was recently also an
overhaul of the states for queues, with a patch to 1.4 that made it  
possible
to build a stronger relationship between a queue member and
a state object.

/O

___
-- 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

Re: [asterisk-users] SIP Registration and INVITE question

[Olle E. Johansson]

7 apr 2009 kl. 12.08 skrev Steve Davies:

> 2009/4/7 Olle E. Johansson :
>>
> [snip]
>>
>> The REGISTER request in the RFC was really written for a device.
>> The way providers use it for trunks with multiple DIDs is outside  
>> of the
>> RFC and is discussed in relation to the SIPconnect specification in
>> the SIP forum.
>>
>> Some providers solve this by not using the Contact: in the register
>> request at all for the calls, instead guessing a URI with the DID
>> in the user name part, something that breaks communication
>> even more as the Contact might include other hints on call routing
>> internally, like line button in a SNOM phone.
>>
>> I would say that the only way right now is to parse the To: header.
>> I started working on some code a while ago that would handle
>> this better, but never completed it. We simply registered a random
>> string and then replaced it with whatever was sent in the To: header
>> (which should be the original destination) before hitting the  
>> dialplan.
>> That code still exists in a branch somewhere and in Pineapple.
>>
>> This code would also solve the issue with registering multiple
>> accounts with one provider.
>> /O
>>
>
> Thanks Olle, as always, a useful response :)
>
> In the meantime, I suspect  that the following is the current dialplan
> based workaround for calls that come in to 's' because of a default
> Registration Contact?

Yes, if you don't add an extension at the end of the register=
configuration, Asterisk defaults to "s" which really is used
all around Asterisk when we don't have a given extension.


/O

___
-- 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

Re: [asterisk-users] ISDN Timer T309

[Afonso Zimmermann]

Martin escreveu:

  Hi,

You're right. I wasn't aware of this patch getting into the code.
In the version you're running the code is already present.

The only problem I see is that some other timer kicks in here and the
T309 cannot be scheduled.

q931.c has this ...
/* For a call in Active state, activate T309 only if there is no timer
already running. */

You'd have to probably dig deeper in it to find out more. But this is
the "latest" explanation I see.
That would explain why the call is disconnected/hanged up right when
the alarm happens.

One way to fix it for you would be to remove the already running timer
so the T309 could be scheduled since
anyways all other timers do not matter since without T309 the call is
hanged up anyways.

Martin

On Mon, Apr 6, 2009 at 12:23 PM, Afonso Zimmermann  wrote:
  
  
Martin escreveu:

Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:

"The implementation of timer T309 in the user side is optional"

Martin

On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann 
wrote:


Martin escreveu:

What is the specification for T309 ? I'm too lazy to look it up.

The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.

The timer is simply deactivated since all the calls are supposed to
drop. I believe that agrees with Q921/Q931 specs.

Martin

On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann 
wrote:


Hi everione,

I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
timer fail with a telco link in this scenario:

Telco Phone <--> Telco <---> Asterisk <> Sip
Phone

When i make a call from Telco Phone to Sip Phone, the call complete, but
when i disconnect the link and reconnect in few seconds, the Asterisk clear
call:

[Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 1: Red Alarm
[Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 1
  == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
'DAHDI/1-1'
[Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
D-channels available!  Using Primary channel 16 as D-channel anyway!
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 2: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 2: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 3: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 3: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 4: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 4: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 5: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 5: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 6: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 6: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 7: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 7: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 8: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 8: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 9: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 9: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 10: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 10: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 11: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 11: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 12: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable

Re: [asterisk-users] Please Advice SIP 183 progessl

[Khaled W. Chehab]

Kindly can you send me the code ,or how to 


Regards


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Monday, April 06, 2009 7:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl

Hi,

The easiest is to turn off MOH on the Dial. Otherwise the patch is
easy but not trivial.
Once the B-leg receives the ringing message and passes it in Dial app
then the code has to turn off the MOH
and tell the A-leg to send the ringing message. At the same time the
code that skips passing the ringing to A-leg
has to be disabled.

Martin

On Mon, Apr 6, 2009 at 2:38 AM, Khaled W. Chehab 
wrote:
> Dear Martin
>
> Can you inform me how to make the patch or from where I can get it
otherwise
> if there is an application can generate it?
> Or if its relate to chan_sip.c ,please can you tell me which function to
> edit or lines to be added
>
> Regards
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
> Sent: Sunday, April 05, 2009 5:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
>
> Hi Khaled,
>
> app Dial clearly is coded to ignore the 180 Ringing being passed if
> you have 'm' option to Dial and you do.
> Try to take the 'm' out and see if 180 Ringing is passed to the A-leg.
>
> So if you want MOH and then when 180 Ringing comes to turn it off =>
> you need a patch.
>
> Martin
>
> 2009/4/4 Khaled W. Chehab :
>> 10x Martin ,
>>
>> But B-Leg is sending 180 ringing
>>
>> Regards
>
> ___
> -- 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
>
>
>
> *
> No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
>
> This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
>
> If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
>
> Xplorium does not guarantee the integrity of this electronic message and
any of its attachments, or that they are free from computer viruses or other
defects.
> *
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



*
No employee or agent is authorized to conclude any binding agreement on behalf 
of Xplorium with another party by e-mail without express written confirmation 
by an officer of Xplorium. Any views expressed by an individual in this 
electronic message do not necessarily reflect views of Xplorium or its 
subsidiaries and associates.

This electronic message and its attachments are solely addressed to the 
addressee(s), and contain confidential information protected from disclosure 
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its 
attachments, kindly delete it immediately from your system and notify the 
sender by electronic mail. You must not copy this message or attachment or 
disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any of 
its attachments, or that they are free from computer viruses or other defects.
*



___
-- 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

Re: [asterisk-users] [OT] Re: async agi question

[cyr2242]

I'm sorry but I can't find where to change this one in the opensubscriber 
service. I'm sending a request to them for it. As soon as I get the answer 
I''ll do it.
Regards
Jose Arias

-- Philipp Kempgen wrote : 
cyr2...@gmai... schrieb:
> This message was sent on behalf of cyr2...@gmai... at openSubscriber.com

Use the appropriate header field for that information.
It's called "From" (in contrast to "Sender").


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


--
This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com
http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11865296.html

___
-- 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

Re: [asterisk-users] SIP Registration and INVITE question

[Steve Davies]

2009/4/7 Olle E. Johansson :
>
[snip]
>
> The REGISTER request in the RFC was really written for a device.
> The way providers use it for trunks with multiple DIDs is outside of the
> RFC and is discussed in relation to the SIPconnect specification in
> the SIP forum.
>
> Some providers solve this by not using the Contact: in the register
> request at all for the calls, instead guessing a URI with the DID
> in the user name part, something that breaks communication
> even more as the Contact might include other hints on call routing
> internally, like line button in a SNOM phone.
>
> I would say that the only way right now is to parse the To: header.
> I started working on some code a while ago that would handle
> this better, but never completed it. We simply registered a random
> string and then replaced it with whatever was sent in the To: header
> (which should be the original destination) before hitting the dialplan.
> That code still exists in a branch somewhere and in Pineapple.
>
> This code would also solve the issue with registering multiple
> accounts with one provider.
> /O
>

Thanks Olle, as always, a useful response :)

In the meantime, I suspect  that the following is the current dialplan
based workaround for calls that come in to 's' because of a default
Registration Contact?

[default]
exten => s,1,Set(DN=${SIP_HEADER(TO):5})
exten => s,n,Set(DN=${CUT(DN,@,1)})
exten => s,n,GotoIf($["${DN}" = "s"]?:default,${DN},1)
exten => s,n,Hangup()

Comments or improvements anyone?

Thanks again.
Steve

___
-- 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

Re: [asterisk-users] Using multiple 'peer' identities on one phone with 1.4

[Florian Hackenberger]

On Tuesday 07 April 2009, Olle E. Johansson wrote:
> Well, you can have OpenSER doing the authentication and turn it
> off in Asterisk, but still match a device.
Ok, but what about sip device state? Will that work? Will asterisk 
report the device as busy when the sip device is engaged in a call?

> Why not use type=user if you have the need for multiple accounts
> on the same IP? Using type=peer for incoming call matching is
> when you can't match on from: but have to match on IP.
Because call limits for queues don't work when using type=friend. If the 
user gets a call through a queue, another queue won't send a call to 
the user, but if the user places a call, asterisk does not consider the 
user busy and tries to forward queue calls to that user. Are there 
workarounds for that?

Cheers,
Florian

-- 
DI Florian Hackenberger
flor...@hackenberger.at
www.hackenberger.at

___
-- 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

Re: [asterisk-users] app_backticks and 1.6

[Olivier]

2009/4/7 Tzafrir Cohen 

> On Tue, Apr 07, 2009 at 10:16:41AM +0200, Olivier wrote:
> > Hello,
> >
> > Is there any app_backticks (see
> > http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks) equivalent or
> > workaround for 1.6 ?
>
> That page, while still messy, now references the new
> http://www.voip-info.org/wiki/view/Asterisk+func+shell


Thanks !!
I didn't know this one ...
I'll update voip-info.org accordingly ...


> 
>
> --
>   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
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] OT - SIP MESSAGE, newline chars and formatting

[Olivier]

Hi,

I'm using a SIP phone (Thomson ST2030) which is able to display text
received though Asterisk's SendText() application.

I'm using this to display from Asterisk "Forwarded to 0123456789" whenever a
user forwards his calls to another number or extension.
Test is displayed with white letters on black background.

What I can't do at the moment is erasing this "Forwarded to 0123456789" text
when user cancels previous forwarding.
If I'm sending a string full of space chars, I've got an ugly string of
black rectangles on LCD screen.

Phone vendor says it can be done sending a "single carriage return" string
to the phone (using usual SendText, I suppose) but either :
A- I can't build correctly such "single carriage return" string,
B- I can't send it (I shouldn't use SendText()),
C- or I misunderstood vendor's advice.

When setting an AEL2 variable with "Hello\rWorld"-like string value, I can
see this string passed in SIP MESSAGE like this :

Content-Type: text/plain
Content-Length: 12

Hello\rWorld


Unfortunately, this string (and other variations) are literally displayed (I
hoped to use \r to erase Hello word and see only the remaning World on my
phone screen).

Is there a way to format text embedded in SIP MESSAGES or work around this ?
Any hint or advice ?

Regards
___
-- 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

Re: [asterisk-users] app_backticks and 1.6

[Tzafrir Cohen]

On Tue, Apr 07, 2009 at 10:16:41AM +0200, Olivier wrote:
> Hello,
> 
> Is there any app_backticks (see
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks) equivalent or
> workaround for 1.6 ?

That page, while still messy, now references the new
http://www.voip-info.org/wiki/view/Asterisk+func+shell

-- 
   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

___
-- 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

Re: [asterisk-users] Sangoma and BT single lines

[bails]

Ed W wrote:
> Hi, got a Sangoma A200 with a bunch of extension cards and having real 
> problems getting it to deal with a normal single BT line
> 
> Symptoms are that incoming calls are fine.  Outgoing calls ring the far 
> end, BUT asterisk never sees that the call is answered (ie no message in 
> the logs files saying so), as a result the remove end can hear the PBX 
> side talking, but there is no audio back from the remote side to us.  
> When we hangup the log files show messages thave suggest it thinks the 
> line is still ringing
> 
> Comparing with another line which works fine (this is a BT multi-line 
> system with what they call "PBX signalling" on it) I see that as soon as 
> the remote end answers then asterisk gets a log message stating the same 
> and audio is fine on this line
> 
> 
> Have now spent nearly 4 months trying to get the signalling sorted on 
> this line.  Most recently we requested "dual signalling" on the line - 
> the end result is now that outbound calls work and asterisk reports that 
> the phone answers, however, when you hangup the call then asterisk 
> obviously gets a bunch of extra line reversals and things there is an 
> immediate incoming call on the back of that outgoing call...
> 
> Please - any suggestions on how to configure a Sangoma card for use with 
> a normal BT single line?
> 
> Thanks
> 
> Ed W

Hi Ed

This is from a system with a Sangoma A200 on a Virgin/Telewest analogue 
line, haven't had a problem in 2 years

/etc/wanpipe/wanpipe1.conf

[devices]
wanpipe1 = WAN_AFT_ANALOG, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE   = AFT
S514CPU = A
CommPort= PRI
AUTO_PCISLOT= NO
PCISLOT = 11
PCIBUS  = 0
FE_MEDIA= FXO/FXS
TDMV_LAW= MULAW
TDMV_OPERMODE   = FCC
RM_NETWORK_SYNC = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN   = 1

[w1g1]
ACTIVE_CH   = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC   = NO

/etc/zaptel.conf

loadzone=us
defaultzone=us

#Sangoma A200 [slot:11 bus:0 span:1]
fxsks=3
fxsks=4

/etc/asterisk/zapata.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=4.0
txgain=2.0
group=1
callgroup=1
pickupgroup=1

immediate=no

;Sangoma A200 [slot:11 bus:0 span:1]
context=from-zaptel
group=0
signalling = fxs_ks
channel => 3

#context=from-zaptel
#group=0
#signalling = fxs_ks
#channel => 4

As you can see only channel 3 is in use and yes I know zaptel.conf says 
loadzone=us defaultzone=us but it appeasr to make no difference with 
this card.

Hope this helps

Bails

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


___
-- 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

[asterisk-users] app_backticks and 1.6

[Olivier]

Hello,

Is there any app_backticks (see
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks) equivalent or
workaround for 1.6 ?
In the past, I had trouble trying to use ENV() function.

Cheers
___
-- 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

Re: [asterisk-users] Sangoma and BT single lines

[Gordon Henderson]

On Mon, 6 Apr 2009, Ed W wrote:

> Hi, got a Sangoma A200 with a bunch of extension cards and having real
> problems getting it to deal with a normal single BT line
>
> Symptoms are that incoming calls are fine.  Outgoing calls ring the far
> end, BUT asterisk never sees that the call is answered (ie no message in
> the logs files saying so), as a result the remove end can hear the PBX
> side talking, but there is no audio back from the remote side to us.
> When we hangup the log files show messages thave suggest it thinks the
> line is still ringing

I've not used Sangoma analogue cards - just digium and OpenVox.

Never really had an issue though. For analogue lines, Asterisk assumes the 
line is "open" or answered as soon as the last DTMF digit is sent down the 
line - it's got no real way of knowing when the far-end answers. (At least 
with normal BT lines I've used)

So I'd be wondering if there isn't something else going on? One way audio 
is often NAT issues, but it's working with incoming calls for you, but not 
outgoing - which seems odd..

> Comparing with another line which works fine (this is a BT multi-line
> system with what they call "PBX signalling" on it) I see that as soon as
> the remote end answers then asterisk gets a log message stating the same
> and audio is fine on this line

I've never asked BT for anything special on lines - I wasn't aware they 
could do much on analogue lines... (I think you can tell them the lines 
are for PBX use, but I've never had any issues)

> Have now spent nearly 4 months trying to get the signalling sorted on
> this line.  Most recently we requested "dual signalling" on the line -
> the end result is now that outbound calls work and asterisk reports that
> the phone answers, however, when you hangup the call then asterisk
> obviously gets a bunch of extra line reversals and things there is an
> immediate incoming call on the back of that outgoing call...
>
> Please - any suggestions on how to configure a Sangoma card for use with
> a normal BT single line?

Bit of a long-shot, but try an OpenVox card as a cheap test of alternative 
technology?

Gordon

___
-- 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

Re: [asterisk-users] SIP Registration and INVITE question

[Olle E. Johansson]

6 apr 2009 kl. 18.46 skrev Steve Davies:

> Thanks for the reply - Perhaps I was not clear.
>
> On the register=> line, if I set /extension to be /12345, then this
> just replaces 's' with 12345, and ALL calls, regardless of their
> destination number will be routed on the INVITE line to 12...@x.x.x.x,
> and the actual destination is specified in the To: header.
>
> Not particularly useful, and I'd prefer not to have to go fumbling
> through the SIP headers to find what was really dialled :)
>
> Looking at the SIP RFC, the idea is that you specify a set of "What I
> will accept" details with each registration in the Contact: headers,
> which is intended to include _multiple_ possible destination
> addresses. The Registrar will then only ever send calls addressed to
> that list of destinations. Sadly, the RFC authors did not think to
> consider private point-to-point links where you can usefully say "send
> me anything, you know best". Asterisk "fills" by defaulting to a
> single s...@x.x.x.x, where the 's' can be replaced by any single number.
>

The REGISTER request in the RFC was really written for a device.
The way providers use it for trunks with multiple DIDs is outside of the
RFC and is discussed in relation to the SIPconnect specification in
the SIP forum.

Some providers solve this by not using the Contact: in the register
request at all for the calls, instead guessing a URI with the DID
in the user name part, something that breaks communication
even more as the Contact might include other hints on call routing
internally, like line button in a SNOM phone.

I would say that the only way right now is to parse the To: header.
I started working on some code a while ago that would handle
this better, but never completed it. We simply registered a random
string and then replaced it with whatever was sent in the To: header
(which should be the original destination) before hitting the dialplan.
That code still exists in a branch somewhere and in Pineapple.

This code would also solve the issue with registering multiple
accounts with one provider.
/O


---
* Olle E. Johansson - o...@edvina.net
* Asterisk Training http://edvina.net/training/




___
-- 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