[Asterisk-Users] hide caller id
Pedro Vela
pvela at bomontec.com
Thu Jun 10 23:55:34 MST 2004
Hi,
We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using
restrictcid=yes and doesn´t work.
What can I do, thaks
Pedro
-----Mensaje original-----
De: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]En nombre de
asterisk-users-request at lists.digium.com
Enviado el: miércoles, 31 de marzo de 2004 12:00
Para: asterisk-users at lists.digium.com
Asunto: Asterisk-Users digest, Vol 1 #3275 - 3 msgs
Send Asterisk-Users mailing list submissions to
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You can reach the person managing the list at
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of Asterisk-Users digest..."
Today's Topics:
1. Re: AW: [Asterisk-Users] CAPI problems when loading chan_capi.so
(Martin Mielke)
2. RE: RxFax/spandsp: not disconnecting (Reynaldo Simbulan)
3. Re: Asterisk and picoCell GSM Base Stations (Simon Anderson)
--__--__--
Message: 1
Date: Wed, 31 Mar 2004 10:42:23 +0200
From: Martin Mielke <martin.mielke at thales-is.com>
To: asterisk-users at lists.digium.com
Subject: Re: AW: [Asterisk-Users] CAPI problems when loading chan_capi.so
Reply-To: asterisk-users at lists.digium.com
Hallo Sacha, :-P
Sascha Knific wrote:
>Hi
>
>
>
>>capiinfo gives:
>>---
>>capi not installed - No such device or address (6)
>>---
>>
>>
>
>It´s not just about installing the apropriate package but you have to load
>the capi kernel module for your isdn card.
>
>The module to load on boot time is set in /etc/isdn/capi.conf (on Debian).
>You have to check how it´s done on your distro (I presume RedHat or SuSE).
>
>
I use SuSE 9.0 Pro.
I don't see any capi.conf - the only similar thing is
/etc/capisuite/capisuite.conf but I don't know if we're talking about
the same file...
The module is loaded at system boot:
---
pbx:~ # dmesg | grep -i capi
capifs: Rev 1.1.4.1
CAPI-driver Rev 1.1.4.1: loaded
capi20: started up with major 68
kcapi: capi20 attached
capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs)
---
I hope it's the right one...
>You can load the module manually. For a AVM Fritz!Card PCI you would do:
>"modprobe fcpci"
>
>
The system has an Eicon Diva Server BRI 2M... and by now I can't find an
specific module...
Martin
--__--__--
Message: 2
From: "Reynaldo Simbulan" <rsimbulan at hotmail.com>
To: <asterisk-users at lists.digium.com>
Date: Wed, 31 Mar 2004 18:39:30 +1000
Subject: [Asterisk-Users] RE: RxFax/spandsp: not disconnecting
Reply-To: asterisk-users at lists.digium.com
Hi Steve,
I am having this problem in which RxFax is still holding the file after
receiving a complete fax. Somehow the zap channel is still active but on the
fax client it was sent successfully.
If you call the line it is still busy.
Changed from phase 3 to 4
>>> MCF: 8c
HDLC underflow in state 8
Changed from phase 4 to 3
Slow carrier up
<<< DCN: fb
DCN with final frame tag
In state 8
Disconnecting
Changed from phase 3 to 7
*CLI> show channels
Channel (Context Extension Pri ) State Appl. Data
Zap/1-1 (faxserver s 3 ) Up RxFAX
/var/lib/asterisk/fax/new/20040329-234801-0755965128.tif
1 active channel(s)
----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, March 31, 2004 6:27 PM
Subject: Asterisk-Users digest, Vol 1 #3273 - 10 msgs
> Send Asterisk-Users mailing list submissions to
> asterisk-users at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-admin at lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Asterisk-Users digest..."
>
>
> Today's Topics:
>
> 1. Re: Asterisk Security Audit? (Steven Critchfield)
> 2. DTMF Detection Problem (Ron McMillin)
> 3. Re: Caller entered digits ignored during wait.... (Tilghman Lesher)
> 4. Re: Sipcall.co.uk & [*] (Dave Cotton)
> 5. Re: IAX2 trunk mode over satellite (clive18 at webmail.co.za)
> 6. Register vith SIP provider from behind NAT (Simon Brown)
> 7. Can't talk on Cisco VIP 30 using Chan Skinny (Dean)
> 8. Re: Caller entered digits ignored during
> wait.... (Stig Andersson)
> 9. RE: Exception flag set - snom200 (jc)
>
> -- __--__--
>
> Message: 1
> Subject: Re: [Asterisk-Users] Asterisk Security Audit?
> From: Steven Critchfield <critch at basesys.com>
> To: asterisk-users at lists.digium.com
> Date: Tue, 30 Mar 2004 23:03:55 -0600
> Reply-To: asterisk-users at lists.digium.com
>
> On Tue, 2004-03-30 at 16:53, Jim Rosenberg wrote:
> > Has Asterisk ever been audited for common security holes, such as buffer
> > overruns?
> >
> > A quick grep through the source for routines that should never be used,
> > like strcpy, strcat, etc., reveals a lot of it. I fear I fear.
>
> These functions aren't as bad as you make out. They are only dangerous
> when used with unchecked buffers that where accepted from outside
> sources. There are quite a few instances of strcpy and strcat that are
> using string constants and therefore are safe.
>
> Don't take that as an argument against checking other possible security
> concerns. Just as a reminder that the mere existence of certain
> functions doesn't mean it is unsafe.
>
> Also this discussion is probably better dealt with on the -dev list
> where the noise level is better suited for the developers you need to
> target to actually see this message.
> --
> Steven Critchfield <critch at basesys.com>
>
>
> -- __--__--
>
> Message: 2
> Date: Tue, 30 Mar 2004 21:45:19 -0800 (PST)
> From: Ron McMillin <sipnow at sbcglobal.net>
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] DTMF Detection Problem
> Reply-To: asterisk-users at lists.digium.com
>
> --0-1376241818-1080711919=:58147
> Content-Type: text/plain; charset=us-ascii
>
> Hi,
> My set up is like this
> Asterisk--->SipuraATA----->AnalogPhone
> When I'm calling into asterisk from a cell phone, there's no dtmf
detection problem as asterisk can detect correct extensions that I press.
But when the phone is further connected to the AnalogPhone thru the ATA, the
dtmf signal is really short/weak. I've tried to adjust dtmf lengths, gain,
etc. on the ATA and it helps a little bit, but not much. And this seems to
be a problem only if I call in from a cell phone. If I were to use a SIP
phone to call in, it works much better.
>
> Is there a way to make Asterisk to regenerate the DTMF tones to improve
the DTMF tones? Such as making it interpret the DTMF tones and regenerate it
w/ a certain length regardless of original signal length. The reason I want
to DTMF comes to AnalogPhone clearly is because I want to ultimately connect
it to a FXSFXO converter and go back out to PSTN line.
>
> Thank you
> Ron
>
> --0-1376241818-1080711919=:58147
> Content-Type: text/html; charset=us-ascii
>
> <DIV>Hi,</DIV>
> <DIV> My set up is like this</DIV>
> <DIV>Asterisk--->SipuraATA----->AnalogPhone</DIV>
> <DIV>When I'm calling into asterisk from a cell phone, there's no dtmf
detection problem as asterisk can detect correct extensions that I press.
But when the phone is further connected to the AnalogPhone thru the ATA, the
dtmf signal is really short/weak. I've tried to adjust dtmf lengths, gain,
etc. on the ATA and it helps a little bit, but not much. And this seems to
be a problem only if I call in from a cell phone. If I were to use a SIP
phone to call in, it works much better.</DIV>
> <DIV> </DIV>
> <DIV>Is there a way to make Asterisk to regenerate the DTMF tones to
improve the DTMF tones? Such as making it interpret the DTMF tones and
regenerate it w/ a certain length regardless of original signal length. The
reason I want to DTMF comes to AnalogPhone clearly is because I want to
ultimately connect it to a FXSFXO converter and go back out to PSTN
line.</DIV>
> <DIV> </DIV>
> <DIV>Thank you</DIV>
> <DIV>Ron</DIV>
> --0-1376241818-1080711919=:58147--
>
> -- __--__--
>
> Message: 3
> From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
> Subject: Re: [Asterisk-Users] Caller entered digits ignored during
wait....
> Date: Tue, 30 Mar 2004 23:46:13 -0600
> To: asterisk-users at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
>
>
> On 2004 Mar 30, at 20:56, Gene Kochanowsky wrote:
>
> > How would you use the t extension to accomplish this?
>
> exten => s,1,Wait(1)
> exten => s,2,Answer
> exten => s,3,SetVar(loopCnt=0)
> exten => s,4,Background(welcome)
> exten => s,5,Background(parties)
>
> exten => t,1,SetVar(loopCnt=$[${loopCnt} + 1])
> exten => t,2,GotoIf($[${loopCnt} < 3]?s|4)
> exten => t,3,Background(vm-goodbye)
> exten => t,4,Hangup
>
> -Tilghman
>
>
> -- __--__--
>
> Message: 4
> Subject: Re: [Asterisk-Users] Sipcall.co.uk & [*]
> From: Dave Cotton <dcotton at linuxautrement.com>
> To: Asterisk List <asterisk-users at lists.digium.com>
> Date: Wed, 31 Mar 2004 08:19:26 +0200
> Reply-To: asterisk-users at lists.digium.com
>
> On Wed, 2004-03-31 at 01:33, Matt wrote:
> > Hello all.
> >
> > Has anyone managed to get SIPCALL.co.uk's service working with the [*]
box?
> >
> > I've managed to register with other SIP providers but not SIPcall.
> >
> I spent a lot of time trying to get * to connect to SIPcall, I even got
> directly in contact with the support depart of the supplier of the
> hardware, who informed me that it is because * does not handle SIP
> correctly, as I had no trouble connecting to SIPPhone, Nikotel, VoIPTalk
> etc I decided to drop it.
> YMMV
>
> --
> Dave Cotton <dcotton at linuxautrement.com>
>
>
> -- __--__--
>
> Message: 5
> From: <clive18 at webmail.co.za>
> Subject: Re: [Asterisk-Users] IAX2 trunk mode over satellite
> To: asterisk-users at lists.digium.com
> Date: Wed, 31 Mar 2004 08:27:59 +0200
> Reply-To: asterisk-users at lists.digium.com
>
> Hi
>
> I have even used H323 over satelite, and beside the lagg,
> no trouble. My only issue is the jitter buffer on IAX2
> seems to be broken. On a very jittery connection, I can
> hardly make a decent call on IAX2.
>
> Good luck!
> regards
> Clive
>
>
> On Tue, 30 Mar 2004 10:15:52 -0800
> John Todd <jtodd at loligo.com> wrote:
> >
> > Today has been the day for satellite questions,
> > apparently, so I'll proxy one out to the rest of the
> > community... I asked this tangentially a month or two
> > ago, but I'll put it in a more blunt way:
> >
> > If you have IAX2 trunking mode experience over satellite,
> > please let us know your experiences with that
> > protocol/transport combination.
> >
> > I've got several people asking about IAX2 and trunk mode
> > over satellite. I have not experimented with IAX2 over
> > satellite (though I have used IAX1 over satellite) and
> > I'm wondering if anyone has direct experiences with
> > IAX2's jitter buffer control over such long-latency
> > connections.
> >
> > I've had SIP working very well over satellite (despite
> > what some people have found to the contrary on this list)
> > and other than the lag there have been no issues that
> > have come up on a reasonably-managed satellite segment.
> > However, the IP overhead really starts to cost
> > significant amounts of pennies when you add it up on
> > multiple SIP RTP sessions over the same link. Plus,
> > packet contention and buffering may (_may_) be an issue
> > when pushing multiple simultaneous streams out the same
> > transponder.
> >
> > It would seem to me that IAX2 in trunk mode would be
> > optimal for people on very expensive satellite bandwidth,
> > as a G.729 9.6kbps channel starts to actually look like
> > 9.6kbps instead of 24kbps. However, I have had mixed
> > success with IAX2 in certain circumstances. Before I
> > start to ask for favors and get satellite time for
> > testing, I'd like to see if anyone else has performed
> > this experiment. If you'd wish to remain anonymous,
> > please mail me directly and I'll appropriately trim
> > identity information and re-distribute, or re-write as
> > appropriate.
> >
> > Other hints I have heard/used on VoIP over satellite:
> > - use small transmit cell (packet) sizes on your
> > satellite gear
> > - turn off error correction (why use it for VoIP?)
> > - turn off compression (G.729 is already compressed;
> > you ARE using
> > G.729, right?)
> > - ensure minimal latency on the terrestrial portions
> > of the call
> > - tell your users to suck it up and deal with the
> > half-second lag
> >
> > JT
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> __________________________________________________________________________
> http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price
>
> -- __--__--
>
> Message: 6
> Date: Wed, 31 Mar 2004 16:37:06 +1000
> From: "Simon Brown" <Simon.Brown at otterson.com.au>
> To: <asterisk-users at lists.digium.com>
> Subject: [Asterisk-Users] Register vith SIP provider from behind NAT
> Reply-To: asterisk-users at lists.digium.com
>
> I cannot successfully register with, or even make calls to, a SIP =
> provider
> (such as FWD) with my * server sitting behind a NAT. The firewall is a =
> Cisco
> 827 router running 12.3 IOS.
>
> Has anyone successfully got their server behind NAT to register or make =
> a
> call to a SIP provider?
>
> TIA=20
>
> Simon
>
> -----
> This mail was content checked for malicious code and viruses
> by GFI MailSecurity.
>
>
> -- __--__--
>
> Message: 7
> From: Dean <aster at zanadoo.net>
> To: asterisk-users at lists.digium.com
> Organization:
> Date: 31 Mar 2004 00:28:06 -0800
> Subject: [Asterisk-Users] Can't talk on Cisco VIP 30 using Chan Skinny
> Reply-To: asterisk-users at lists.digium.com
>
> I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like
> to use with asterisk, I have set them up using chan_skinny. The phones
> work well, except the only problem is that it is like the cisco phones
> are muted. When I talk on the cisco phones I can hear my self through
> the ear peice, but the person who I am calling can not hear me at all. I
> have tried various cisco phones from various sources on 2 different
> linux computers (one running redhat 7.3 and one running redhat 9) I have
> tried using the 0.7.2 code and the latest development code from the CVS
> and I still get the same results. All help will be greatly appreciated.
> Below is the error log and my skinny.conf
>
> Thanks,
>
> Dean
>
> Error Log:
>
> Mar 31 00:09:29 WARNING[1024]: Ignoring port for now
> Mar 31 00:09:29 WARNING[10251]: Read error on sound device: Resource
> temporarily unavailable
> Mar 31 00:09:29 WARNING[1024]: Ignoring rxwink
> Mar 31 00:16:48 WARNING[1024]: Ignoring port for now
> Mar 31 00:16:48 WARNING[10251]: Read error on sound device: Resource
> temporarily unavailable
> Mar 31 00:16:48 WARNING[1024]: Ignoring rxwink
> Mar 31 00:22:30 WARNING[16401]: No audio available on
> Skinny/133 at flex-6??
>
>
> Console:
>
> skinny_answer(Skinny/133 at flex-6) on 133 at flex-6
> Recieved Open Recieve Channel Ack
> us port: 17874
> sin port: 53316
> -- Playing 'voicemail/default/1234/unavail' (language 'en')
> -- Playing 'vm-intro' (language 'en')
> -- Playing 'beep' (language 'en')
> -- x=0, open writing:
> /var/spool/asterisk/voicemail/default/1234/INBOX/msg0004 format: wav49,
> 0x80dc958
> -- x=1, open writing:
> /var/spool/asterisk/voicemail/default/1234/INBOX/msg0004 format: gsm,
> 0x811a3f8
> -- x=2, open writing:
> /var/spool/asterisk/voicemail/default/1234/INBOX/msg0004 format: wav,
> 0x811a570
> Mar 31 00:22:30 WARNING[16401]: app_voicemail.c:1222 play_and_record: No
> audio available on Skinny/133 at flex-6??
> -- User hung up
> == Spawn extension (demo, 1235, 1) exited non-zero on
> 'Skinny/133 at flex-6'
> skinny_hangup(Skinny/133 at flex-6) on 133 at flex
>
> Error Log
>
> Mar 31 00:09:29 WARNING[1024]: Ignoring port for now
> Mar 31 00:09:29 WARNING[10251]: Read error on sound device: Resource
> temporarily unavailable
> Mar 31 00:09:29 WARNING[1024]: Ignoring rxwink
> Mar 31 00:16:48 WARNING[1024]: Ignoring port for now
> Mar 31 00:16:48 WARNING[10251]: Read error on sound device: Resource
> temporarily unavailable
> Mar 31 00:16:48 WARNING[1024]: Ignoring rxwink
> Mar 31 00:22:30 WARNING[16401]: No audio available on
> Skinny/133 at flex-6??
>
> ;
> ; Skinny Configuration for Asterisk
> ;
> [general]
> port = 2000 ; Port to bind to, default tcp/2000
> bindaddr = 0.0.0.0 ; Address to bind to
> dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)
> keepAlive = 120
>
> ;allow = all
> ;disallow =
>
>
> ; Typical config for 12SP+
> [florian]
> device=SEPXXXXXXXXXXXX
> version=P002G204 ; Thanks critch
> context=demo
> line => 120 ; Dial(Skinny/120 at florian)
>
> ; Typical config for 12SP+
> [florianx]
> device=SEPXXXXXXXXXXXX
> version=P0020301003 ; Thanks critch
> context=default
> line => 122 ; Dial(Skinny/120 at florian)
>
> ; Typical config for 12SP+
> [flex]
> device=SEPXXXXXXXXXXXX
> version=P002F202
> context=demo
> line => 133
>
>
>
> -- __--__--
>
> Message: 8
> Date: Wed, 31 Mar 2004 09:08:53 +0200
> To: asterisk-users at lists.digium.com
> From: Stig Andersson <stig at ymex.se>
> Subject: Re: [Asterisk-Users] Caller entered digits ignored during
> wait....
> Reply-To: asterisk-users at lists.digium.com
>
> Asterisk doesn't accept keys during wait, use Background
> and play 1 sec silence instead.
>
> /Stig
>
> At 23:46 2004-03-30 -0600, you wrote:
> >
> >On 2004 Mar 30, at 20:56, Gene Kochanowsky wrote:
> >
> >> How would you use the t extension to accomplish this?
> >
> >exten => s,1,Wait(1)
> >exten => s,2,Answer
> >exten => s,3,SetVar(loopCnt=0)
> >exten => s,4,Background(welcome)
> >exten => s,5,Background(parties)
> >
> >exten => t,1,SetVar(loopCnt=$[${loopCnt} + 1])
> >exten => t,2,GotoIf($[${loopCnt} < 3]?s|4)
> >exten => t,3,Background(vm-goodbye)
> >exten => t,4,Hangup
> >
> >-Tilghman
> >
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> --------------------------------------------------------------------------
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>
> -- __--__--
>
> Message: 9
> From: "jc" <asterisk-user at christoffersonrobb.com>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] Exception flag set - snom200
> Date: Wed, 31 Mar 2004 08:16:07 +0100
> Reply-To: asterisk-users at lists.digium.com
>
> This is a multi-part message in MIME format.
>
> ------=_NextPart_000_007D_01C416F8.6B262920
> Content-Type: text/plain;
> charset="us-ascii"
> Content-Transfer-Encoding: 7bit
>
> Asterisk CVS-03/11/04 18:18:12
>
> snom200-SIP 2.03o
>
>
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Ernest W.
> Lessenger
> Sent: Wednesday, March 31, 2004 1:01 AM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Exception flag set - snom200
>
>
>
> What version of asterisk are you using, and what version of the SNOM
> firmware?
>
>
>
> --Ernest
>
>
>
>
> _____
>
>
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of jc
> Sent: Tuesday, March 30, 2004 10:20 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Exception flag set - snom200
>
> Sorry I forgot the subject in the last post.
>
>
>
> When my snom200 receives an inbound SIP external sip call, it somehow
> rejects the call and with a busy tone. The debug shows the following
> error:
>
>
>
> channel.c:1142 ast_read: Exception flag set on 'SIP/sipphone-7796', but
> no exception handler
>
>
>
>
>
> what does this mean and how can I debug it further??
>
>
>
> Thanks
>
> JC
>
>
>
>
> ------=_NextPart_000_007D_01C416F8.6B262920
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> {font-family:Arial;
> color:navy;}
> @page Section1
> {size:8.5in 11.0in;
> margin:99.35pt 1.25in 83.5pt 1.25in;}
> div.Section1
> {page:Section1;}
> -->
> </style>
>
> </head>
>
> <body lang=3DEN-US link=3Dblue vlink=3Dpurple>
>
> <div class=3DSection1>
>
> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
> lang=3DSV
> style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Asterisk =
> CVS-03/11/04
> 18:18:12 </span></font></p>
>
> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
> lang=3DSV
> style=3D'font-size:10.0pt;font-family:Arial;color:navy'>snom200-SIP =
> 2.03o</span></font></p>
>
> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
> lang=3DSV
> style=3D'font-size:10.0pt;font-family:Arial;color:navy'> </span></fo=
> nt></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
> face=3DTahoma><span
> style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original =
> Message-----<br>
> <b><span style=3D'font-weight:bold'>From:</span></b> =
> asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] <b><span =
> style=3D'font-weight:
> bold'>On Behalf Of </span></b>Ernest W. Lessenger<br>
> <b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
> 31, 2004
> 1:01 AM<br>
> <b><span style=3D'font-weight:bold'>To:</span></b>
> asterisk-users at lists.digium.com<br>
> <b><span style=3D'font-weight:bold'>Subject:</span></b> RE: =
> [Asterisk-Users]
> Exception flag set - snom200</span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
> face=3D"Times New Roman"><span
> style=3D'font-size:12.0pt'> </span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
> color=3Dblue face=3DArial><span
> style=3D'font-size:10.0pt;font-family:Arial;color:blue'>What version of =
> asterisk
> are you using, and what version of the SNOM firmware?</span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
> face=3D"Times New Roman"><span
> style=3D'font-size:12.0pt'> </span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
> color=3Dblue face=3DArial><span
> style=3D'font-size:10.0pt;font-family:Arial;color:blue'>--Ernest</span></=
> font></p>
>
> <blockquote style=3D'border:none;border-left:solid blue =
> 1.5pt;padding:0in 0in 0in 4.0pt;
> margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'=
> >
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
> face=3D"Times New Roman"><span
> style=3D'font-size:12.0pt'> </span></font></p>
>
> <div class=3DMsoNormal align=3Dcenter =
> style=3D'margin-left:.5in;text-align:center'><font
> size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>
>
> <hr size=3D2 width=3D"100%" align=3Dcenter>
>
> </span></font></div>
>
> <p class=3DMsoNormal =
> style=3D'margin-right:0in;margin-bottom:12.0pt;margin-left:
> .5in'><b><font size=3D2 face=3DTahoma><span =
> style=3D'font-size:10.0pt;font-family:
> Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2 =
> face=3DTahoma><span
> style=3D'font-size:10.0pt;font-family:Tahoma'>
> asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] <b><span =
> style=3D'font-weight:
> bold'>On Behalf Of </span></b>jc<br>
> <b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, March 30, =
> 2004
> 10:20 AM<br>
> <b><span style=3D'font-weight:bold'>To:</span></b> =
> asterisk-users at lists.digium.com<br>
> <b><span style=3D'font-weight:bold'>Subject:</span></b> [Asterisk-Users]
> Exception flag set - snom200</span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
> color=3Dnavy face=3DArial><span
> style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Sorry I forgot =
> the
> subject in the last post.</span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
> face=3D"Times New Roman"><span
> style=3D'font-size:12.0pt'> </span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
> color=3Dnavy face=3DArial><span
> style=3D'font-size:10.0pt;font-family:Arial;color:navy'>When my snom200 =
> receives
> an inbound SIP external sip call, it somehow rejects the call and with a =
> busy
> tone. The debug shows the following error:</span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
> face=3D"Times New Roman"><span
> style=3D'font-size:12.0pt'> </span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
> color=3Dnavy face=3DArial><span
> style=3D'font-size:10.0pt;font-family:Arial;color:navy'>channel.c:1142 =
> ast_read:
> Exception flag set on 'SIP/sipphone-7796', but no exception =
> handler</span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
> face=3D"Times New Roman"><span
> style=3D'font-size:12.0pt'> </span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
> face=3D"Times New Roman"><span
> style=3D'font-size:12.0pt'> </span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
> color=3Dnavy face=3DArial><span
> style=3D'font-size:10.0pt;font-family:Arial;color:navy'>what does this =
> mean and
> how can I debug it further??</span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
> face=3D"Times New Roman"><span
> style=3D'font-size:12.0pt'> </span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
> color=3Dnavy face=3DArial><span
> style=3D'font-size:10.0pt;font-family:Arial;color:navy'>Thanks =
> </span></font></p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
> color=3Dnavy face=3DArial><span
> style=3D'font-size:10.0pt;font-family:Arial;color:navy'>JC</span></font><=
> /p>
>
> <p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
> face=3D"Times New Roman"><span
> style=3D'font-size:12.0pt'> </span></font></p>
>
> </blockquote>
>
> </div>
>
> </body>
>
> </html>
>
> ------=_NextPart_000_007D_01C416F8.6B262920--
>
>
>
> -- __--__--
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> End of Asterisk-Users Digest
>
--__--__--
Message: 3
Subject: Re: [Asterisk-Users] Asterisk and picoCell GSM Base Stations
From: Simon Anderson <lists at anderson.gen.nz>
To: asterisk-users at lists.digium.com
Date: Wed, 31 Mar 2004 20:50:21 +1200
Reply-To: asterisk-users at lists.digium.com
On Wed, 2004-03-31 at 20:30, John Todd wrote:
> At 7:18 PM +1200 3/31/04, Simon Anderson wrote:
> >
> >picoCell/microCell/nanoCell Base stations are low powered GSM nodes
> >designed for indoor or small area GSM coverage. Their transmission power
> >is low enough to meet the legal restrictions of many countries.
> >
> >Here are two examples;
> >
> >http://www.rivanetworks.com/nano/nano.htm
> >http://www.ipaccess.com/ipaccess_pages2/bts2.html
> >
> >Both of these Base Stations have an RJ-45 in order to do Ethernet on the
> >back end; GSM in, IP out.
> >
> >I wonder if anyone has experience interfacing one of these or a similar
> >product with Asterisk?
> >
> >I also wonder if Digium has any plans to supply PCI cards which provide
> >similar functionality?
> >
> >The prospect of recycling old cell phones for use as Asterisk extensions
> >is extremely attractive.
> >
> >Cellular phone ---GSM---> picoCell ---IP---> Asterisk
> >
> >Regards,
> >Simon.
>
> This looks interesting, but can you explain this a bit more? The
> term "IP out" is incredibly vague, and on neither of the websites
> (after a 30 second review) did they say exactly what they mean with
> "IP out". Does this turn GSM calls into SIP transactions? Does it
> need another similar device somewhere else on the IP network that
> then turns the calls back into TDM via an FXS interface? What
> _does_ IP mean in this context?
For the Riva model, these are the only specs I could find;
http://www.rivanetworks.com/nano/nanospec.htm
I agree that this looks interesting. I'm hoping that the creation of
this thread on this list fills in some of the blanks.
Regards,
Simon.
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