[Asterisk-Users] help
whminfo at hotlink.com.br
whminfo at hotlink.com.br
Mon Apr 11 15:22:55 MST 2005
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> Today's Topics:
>
> 1. Re: append # to dial string (Eric Wieling)
> 2. Re: VAD/DTX implementation through zaptel cards (Eric Wieling)
> 3. Re: CDR and TDS (Eric Wieling)
> 4. RE: Zaptel Compile on a virtual dedicated host.
> (vgrskovic at optonline.net)
> 5. Re: TE110P/Hipath3750 - Yellow Alarm (Henry Jensen)
> 6. Re: Re: PTSN POTS Differences SOLVED (Robert Keller)
> 7. Re: Can you comment on this Qos script? How does one shape
> RTP? (Sean Kennedy)
> 8. Interface bonding + asterisk (Jesus Mogollon)
> 9. Re: Can you comment on this Qos script? How doesone shape
> RTP? (Henry)
> 10. Re: Can you comment on this Qos script? How does one shape
> RTP? (trixter http://www.0xdecafbad.com)
> 11. RE: Sangoma A101 + Rhino channelbank (mattf)
> 12. Re: Can you comment on this Qos script? How does one shape
> RTP? (Andrew Kohlsmith)
> 13. Re: TDM400P power supply (Ricardo Peironcely)
> 14. Problem with X101P (Yusuf Iqbal)
> 15. Re: Can you comment on this Qos script? How does one shape
> RTP? (trixter http://www.0xdecafbad.com)
> 16. wcfxo problem (Dave Weis)
> 17. (no subject) (Robert Webb)
> 18. Re: Sipura SPA-841 Phone Review (Doug Millsaps)
> 19. Re: From OH323 to SIP or OH323 without gatekeeper (Bruno Hertz)
> 20. Re: wcfxo problem (Sahil Gupta)
> 21. Re: TDM400P Revision question. (Robert Webb)
> 22. Intercom with Aastra 480e? (Bobby Lacey)
> 23. Manipulate Asterisk Database from manager? (Matt)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 11 Apr 2005 08:39:00 -0500
> From: Eric Wieling <eric at fnords.org>
> Subject: Re: [Asterisk-Users] append # to dial string
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <425A7DF4.3060306 at fnords.org>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> John Breeden wrote:
>
> > Been there, done that - no joy :-)
> >
> > It appears the modifier only excepts a numeric, anyone know if/how you
> > can feed it adecimal/hex for ascii #?
> >
> > Rich Adamson wrote:
> >
> >>> Is there anyway to append the '#' symbol to a dial string? -
> >>> hex/octal whatever? I'm surprised that I can't find anything
> >>> searching the wiki or google.
> >>>
> >>
> >>
> >> Try something like this:
> >>
> >> exten => _9XXXXXXX,1,Dial(Zap/4/${EXTEN}#)
>
> Then you are doing something wrong. The above syntax is correct.
>
> --
> Always do right. This will gratify some people and astonish the rest.
> Mark Twain
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 11 Apr 2005 08:40:53 -0500
> From: Eric Wieling <eric at fnords.org>
> Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel
> cards
> To: parijat at varaha.com, Asterisk Users Mailing List - Non-Commercial
> Discussion <asterisk-users at lists.digium.com>
> Message-ID: <425A7E65.2040400 at fnords.org>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> parijat at varaha.com wrote:
>
> > Hi,
> > How can i implement VAD/DTX using zaptel with asterisk towards PSTN.
>
> TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not
> even a valid idea.
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 11 Apr 2005 08:44:00 -0500
> From: Eric Wieling <eric at fnords.org>
> Subject: Re: [Asterisk-Users] CDR and TDS
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <425A7F20.9070701 at fnords.org>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> David Masure wrote:
>
> >
> > Hi,
> >
> > I want to use the cdr to record the call log to my Microsoft SQL Server
> > using unixodbc and freetds....
> >
> > but when I compile, I've got this message....
> >
> > Does anyone have the same problem and/or know how to solve it ?
>
>
> Update of /usr/cvsroot/asterisk/doc
> In directory mongoose.digium.com:/tmp/cvs-serv24936/doc
>
> Added Files:
> README.tds
> Log Message:
> Add documentation for TDS noting compilation problem on 0.63+
>
>
> --- NEW FILE: README.tds ---
> PLEASE NOTE
>
> The cdr_tds module is NOT compatible with version 0.63 of FreeTDS.
>
> The cdr_tds module is known to work with FreeTDS version 0.62.1;
> it should also work with 0.62.2, 0.62.3 and 0.62.4, which are bug
> fix releases.
>
> The cdr_tds module uses the raw "libtds" API of FreeTDS. It appears
> that from 0.63 onwards, this is not considered a published API
> of FreeTDS and is subject to change without notice.
>
> Between 0.62.x and 0.63 of FreeTDS, many incompatible changes
> have been made to the libtds API.
>
> For newer versions of FreeTDS, it is recommended that you use the
> ODBC driver.
>
>
>
> --
> Always do right. This will gratify some people and astonish the rest.
> Mark Twain
>
>
> ------------------------------
>
> Message: 4
> Date: Mon, 11 Apr 2005 09:45:54 -0400
> From: vgrskovic at optonline.net
> Subject: RE: [Asterisk-Users] Zaptel Compile on a virtual dedicated
> host.
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> <asterisk-users at lists.digium.com>
> Message-ID: <007601c53e9c$c85a2b60$0302a8c0 at zeus>
> Content-Type: text/plain; charset="us-ascii"
>
> It appears to be Virtuozzo..
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Henry
> Sent: Monday, April 11, 2005 9:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated
> host.
>
> Hi,
>
> Do you happen to know what VPS system your host uses (e.g. UML,
> Virtuozzo, VMWare, FreeVPS, etc.)? It could make a lot of difference, as
> some platforms will allow changes that others will not.
>
> -- Henry Owens.
>
>
> On 11/4/05 2:20 pm, "vgrskovic at optonline.net" <vgrskovic at optonline.net>
> wrote:
> Giles thank you for getting back so quickly, "dmesg" doesn't output
> anything, but even if it did, I am not sure that I could recompile the
> kernel.
>
> The server I am using is in a virtual dedicated hosting environment, I
> do not have access to recompile the kernel, nor can I replace it. The
> server prevents me from doing so. I do not have access to the "real"
> /boot and don't have access as far as I can tell to the .config for the
> kernel source. ("make oldconfig" seems to work)
>
> After a few more days of tech support, google searches and etc, I have
> found that my provider is using kernel 2.24.21.4.0.1.elsmp. Of course,
> cat /proc/version doesn't think so!! It thinks I am running Kernel
> 2.4.20-021stab022.11.777-enterprise. I am able to use rpmfind to source
> the corresponding rpm which installs without incident. The interesting
> part is "rpm -qa kernel" doesn't see it :-(. I even tried to "rpm
> -rebuilddb"
>
> Zaptel appears to compile fine, but when I run "modprobe zaptel" I get
> the following:
>
> ---->
> /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o:
> kernel-module version mismatch
> /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o
> was compiled for kernel version 2.4.21-4.0.1.EL
> while this kernel is version 2.4.20-021stab022.11.777-enterp.
> /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod
> /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o failed
> /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod
> zaptel failed
> <---
>
> Is there a way to override zaptel's kernel check or have linux fool it
> into thinking the kernel is 2.4.21-4.0.1.EL?
>
> thanks!
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com]
> <mailto:asterisk-users-bounces at lists.digium.com%5d> On Behalf Of Giles
> Coochey
> Sent: Wednesday, April 06, 2005 9:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject:
>
>
> >Anyone have any ideas on where I can find the right kernel source? I
> have look at
> > rpmfind.net and google'd with no avail!
>
> You could always download the Vanilla kernel source from
> http://www.kernel.org and compile a kernel from source. I tend to always
> use the Vanilla source, it's what everything has been tested against and
> it tastes better.
>
> You should probably print out the "dmesg" output to help you configure
> the kernel options prior to compilation so that your "hardware" is
> correctly detected.
>
> I would also urge you to use a bootloader such as grub or lilo to ensure
> that you can revert to the original kernel should it panic on boot, I
> suspect Redhat already uses one of those anyway.
> _______________________________________________
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>
> _____
>
> _______________________________________________
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> ------------------------------
>
> Message: 5
> Date: Mon, 11 Apr 2005 15:54:44 +0200
> From: Henry Jensen <hjensen at gmx.de>
> Subject: Re: [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <20050411135444.GA1282 at jensen.local>
> Content-Type: text/plain; charset=iso-8859-15
>
>
> On Tue, Apr 05, 2005 at 09:06:33PM +0200, Peter Svensson wrote:
> > A yellow alarm means the remote end is sensing some error condition. Try
> > looking for an error message at the remote end. It may be as easy as a
> > broken cable (where the Hipath does not hear the Asterisk box).
>
> The problem is, that the TMS2-Card in the HiPath is not activated,
> it says, that the line is dead. According to the Siemens-People the
> Card should activate itself as soon as a signal reaches the card.
> But it appears, that Asterisk sends no signal.
>
>
> This is what the layout looks like:
>
> Asterisk|TE110P - TMS2|HiPath|TMS2 - PSTN
>
>
> The cable is functional and the wiring is correct. But I'm not sure how I
> must configure the TMS2 card.
>
> Regards,
> Henry
>
>
>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Mon, 11 Apr 2005 07:01:03 -0700
> From: Robert Keller <rkeller at ferndale.wednet.edu>
> Subject: Re: [Asterisk-Users] Re: PTSN POTS Differences SOLVED
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <BE7FD12F.1A2CD%rkeller at ferndale.wednet.edu>
> Content-Type: text/plain; charset="US-ASCII"
>
> Tony, I don't see "${EXTEN}" anywhere in the [macro-dialout-trunk] context.
> Am I missing something?
>
> Robert Andrew Keller
> Ferndale School District #502
> rkeller at ferndale.wednet.edu
> 360-383-9228 PH.
> 360-383-9218 FAX
> "Paving the way for tomorrows genius."
>
> > From: tony at softins.clara.co.uk (Tony Mountifield)
> > Organization: Software Insight Ltd., Winchester, UK
> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> > <asterisk-users at lists.digium.com>
> > Date: Mon, 11 Apr 2005 07:48:19 +0000 (UTC)
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Re: PTSN POTS Differences SOLVED
> >
> > In article <BE7F361F.1A279%rkeller at ferndale.wednet.edu>,
> > Robert Keller <rkeller at ferndale.wednet.edu> wrote:
> >> Thanks Rich, I wasn't sure where to find that context. I found the
> outbound
> >> context in the extensions_additional.conf and added w's in the following
> >> manner:
> >>
> >> [outrt-001-Out1]
> >> include => outrt-001-Out1-custom
> >> exten => _1NXXNXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN})
> >> exten => _1NXXNXXXXXX,2,Macro(outisbusy) ; No available circuits
> >> exten => _9.,1,Macro(dialout-trunk,1,w${EXTEN:1})
> >> exten => _9.,2,Macro(outisbusy) ; No available circuits
> >> exten => _NXXNXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN})
> >> exten => _NXXNXXXXXX,2,Macro(outisbusy) ; No available circuits
> >> exten => _NXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN})
> >> exten => _NXXXXXX,2,Macro(outisbusy) ; No available circuits
> >
> > Couldn't you have just put the w in once, in the Dial command that
> > is inside [macro-dialout-trunk] ?
> >
> > Cheers
> > Tony
> > --
> > Tony Mountifield
> > Work: tony at softins.co.uk - http://www.softins.co.uk
> > Play: tony at mountifield.org - http://tony.mountifield.org
> > _______________________________________________
> > 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
>
>
>
> ------------------------------
>
> Message: 7
> Date: Mon, 11 Apr 2005 07:08:54 -0700
> From: Sean Kennedy <skennedy at tpno-co.org>
> Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How
> does one shape RTP?
> To: asterisk-users at lists.digium.com
> Message-ID: <425A84F6.1050005 at tpno-co.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Honestly, the best script I've ever found is the wondershaper script (
> google it ). I tried the correct one posted in this thread, tried
> modifying it, but in the end I just used wondershaper.
>
> Does a great job. My only fear is it doesn't specifically target IAX2
> traffic as high priority, but I can modify it later to do so if needed.
>
> On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no
> noticable problems. Along with someone streaming a shoutcast station (
> sigh ). The station broke up, but the calls didn't.
>
> cmisip wrote:
>
> >I got this from the voip wiki but the original script didn't seem to
> >work right so I fiddled with it a little bit. I am no expert so maybe
> >someone can look at it for errors. This is for my cable connection. So
> >far asterisk seems to use 1:10 while all other traffic uses 1:102. How
> >does one packet shape RTP?
> >
> >Thanks for any help.
> >
>
>
> ------------------------------
>
> Message: 8
> Date: Mon, 11 Apr 2005 07:15:30 -0700
> From: Jesus Mogollon <gocho26 at gmail.com>
> Subject: [Asterisk-Users] Interface bonding + asterisk
> To: asterisk-users at lists.digium.com
> Message-ID: <d18206f705041107157958062f at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi all
>
> I installed asterisk on a dual PIII 700 with two NICs. I then proceeded to
> configure both NICs with bonding enable (bonding miimon=100 mode=1). I know
> certain features (like load balancing) under a bonded configuration is not
> understood by some switches, so I configured it using mode=1 (Failover
> only). The problem I'm having is that, sometimes, calls start fine but then
> one of the parties loses audio (it could be the caller of the callee who
> loses audio, there is no pattern). I was wondering if someone has hit the
> same wall as me. There are people using this server right now, so I haven't
> tried the no-bonding option as it means downtime. Any help would be
> appreciated.
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> ------------------------------
>
> Message: 9
> Date: Mon, 11 Apr 2005 15:19:38 +0100
> From: Henry <henry at adiungo.com>
> Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How
> doesone shape RTP?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <BE80460A.DE62%henry at adiungo.com>
> Content-Type: text/plain; charset="US-ASCII"
>
> I agree that Wondershaper is a great script; prior to using it in an office
> where I set up asterisk, there were some major problems with call quality,
> but it seems to have helped hugely (the same DSL line is used for both VoIP
> and everyday 'net usage for seven people - not ideal, but I didn't set the
> budget :-) ).
>
> If you happen to modify it to to prioritize IAX2, drop me a copy!
>
> -- Henry Owens.
>
>
> On 11/4/05 3:08 pm, "Sean Kennedy" <skennedy at tpno-co.org> wrote:
>
> > Honestly, the best script I've ever found is the wondershaper script (
> > google it ). I tried the correct one posted in this thread, tried
> > modifying it, but in the end I just used wondershaper.
> >
> > Does a great job. My only fear is it doesn't specifically target IAX2
> > traffic as high priority, but I can modify it later to do so if needed.
> >
> > On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no
> > noticable problems. Along with someone streaming a shoutcast station (
> > sigh ). The station broke up, but the calls didn't.
> >
> > cmisip wrote:
> >
> >> I got this from the voip wiki but the original script didn't seem to
> >> work right so I fiddled with it a little bit. I am no expert so maybe
> >> someone can look at it for errors. This is for my cable connection. So
> >> far asterisk seems to use 1:10 while all other traffic uses 1:102. How
> >> does one packet shape RTP?
> >>
> >> Thanks for any help.
> >>
> > _______________________________________________
> > 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
>
>
>
> ------------------------------
>
> Message: 10
> Date: Mon, 11 Apr 2005 07:21:43 -0700
> From: "trixter http://www.0xdecafbad.com" <trixter at 0xdecafbad.com>
> Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How
> does one shape RTP?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1113229303.6647.14.camel at rufus.home.tld>
> Content-Type: text/plain; charset="us-ascii"
>
> I used the one posted to this list and for a test did a
> speedtest.dslreports.com bandwidth test duringa call, no loss in
> quality.
>
> I set ports 10000-11024 to RTP in rtp.conf, I dont need 10k ports for
> that as I have few calls being processed. I also added sip to the queue
> although that prolly doesnt matter becuase its such a low bandwidth
> protocol comparitevly speaking.
>
>
> # udp/5060 is SIP
> tc filter add dev $DSLDEV parent 1:0 protocol ip prio 1 u32 match ip
> dport 506
> 0 0xffff match ip protocol 17 0xff flowid 1:0
> tc filter add dev $DSLDEV parent 1:0 protocol ip prio 2 u32 match ip
> sport 506
> 0 0xffff match ip protocol 17 0xff flowid 1:0
>
> # udp/10000-11024 is RTP
> tc filter add dev $DSLDEV parent 1:0 protocol ip prio 1 u32 match ip
> dport 100
> 00 0xf670 match ip protocol 17 0xff flowid 1:0
> tc filter add dev $DSLDEV parent 1:0 protocol ip prio 2 u32 match ip
> sport 100
> 00 0xf670 match ip protocol 17 0xff flowid 1:0
>
>
>
> On Mon, 2005-04-11 at 07:08 -0700, Sean Kennedy wrote:
> > Honestly, the best script I've ever found is the wondershaper script (
> > google it ). I tried the correct one posted in this thread, tried
> > modifying it, but in the end I just used wondershaper.
> >
> > Does a great job. My only fear is it doesn't specifically target IAX2
> > traffic as high priority, but I can modify it later to do so if needed.
> >
> > On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no
> > noticable problems. Along with someone streaming a shoutcast station (
> > sigh ). The station broke up, but the calls didn't.
> >
> > cmisip wrote:
> >
> > >I got this from the voip wiki but the original script didn't seem to
> > >work right so I fiddled with it a little bit. I am no expert so maybe
> > >someone can look at it for errors. This is for my cable connection. So
> > >far asterisk seems to use 1:10 while all other traffic uses 1:102. How
> > >does one packet shape RTP?
> > >
> > >Thanks for any help.
> > >
> > _______________________________________________
> > 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
> --
> Trixter http://www.0xdecafbad.com
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> ------------------------------
>
> Message: 11
> Date: Mon, 11 Apr 2005 10:34:51 -0400
> From: mattf <mattf at vicimarketing.com>
> Subject: RE: [Asterisk-Users] Sangoma A101 + Rhino channelbank
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> <asterisk-users at lists.digium.com>
> Message-ID:
>
<DB43F516702AAF4392AA45573F18181901267879 at vicimail.vicimarketinggroup.co
m>
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Keep on bugging the Sangoma guys, I know they are working on several RBS T1
> issues right now(They called me Friday to go over a few things) They just
> need help from users like you and I to find the bugs in their drivers.
>
> Have you tried any other signalling types other than LOOP?
>
> MATT---
>
>
> -----Original Message-----
> From: Felician CHELU [mailto:tehnic at intertel.ro]
> Sent: Monday, April 11, 2005 9:52 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Sangoma A101 + Rhino channelbank
>
>
> Hello,
>
> I have Asterisk 1.0.6 - I try to setup Sangoma A101 T1 board together with
> the Rhino fxs chanelbank.
> Things done:
> - T1 cross cable = I have carrier, signalling and framnig leds on
> the channelbank green.
> - channelbank configuration:
> t1 - Proto: LOOP Frame: esf Clock: slave Coding:
> b8zs
> channels(analog) : Function:A-fxs Mode:loop
> - zaptel.conf
> span=2,1,0,esf,b8zs
> fxols=32-55
> (i have a span 1 with a digium e1)
> - zapata.conf
> .... signalling=fxo_ls
> - wanpipe1.conf
>
> [devices]
> wanpipe1 = WAN_AFT, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE, Comment
>
> [wanpipe1]
> CARD_TYPE = AFT
> S514CPU = A
> CommPort = PRI
> AUTO_PCISLOT = NO
> PCISLOT = 10
> PCIBUS = 2
> FE_MEDIA = T1
> FE_LCODE = B8ZS
> FE_FRAME = ESF
> FE_LINE = 1
> TE_CLOCK = MASTER
> ACTIVE_CH = ALL
> TE_HIGHIMPEDANCE = NO
> LBO = 0DB
> INTERFACE = V35
> CLOCKING = EXTERNAL
> BaudRate = 0
> MTU = 1500
> UDPPORT = 9000
> TTL = 255
> IGNORE_FRONT_END = NO
>
> [w1g1]
> PROTOCOL = HDLC
> HDLC_STREAMING = YES
> ACTIVE_CH = ALL
> IDLE_FLAG = 0x7E
> MTU = 1500
> MRU = 1500
> TDMV_SPAN = 2
> TDMV_ECHO_OFF = NO
> MULTICAST = NO
> TRUE_ENCODING_TYPE = NO
>
>
> I already called Sangoma and Rhino support, but after hours of long distance
> call conversation the problem is still not solved. Finnaly, a guy from Rhino
> told me that their "asterisk expert" (which was not avaliable) knows about
> this problem and that it is that the sangoma driver is not communicating
> with asterisk.
>
> The wanrouter starts ok, after ztcfg I see the channels configured.
> The problem: i don't have dialtone on phones.
>
> Question: When i enter zttoll, if i go to the sangoma span and I make "loop"
> then it freezes. Is it normal?
>
> If someone has experienced this combination and made it work please give me
> a sign.
>
> Thank you.
>
> PS:
>
> Felician
>
> _______________________________________________
> 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
>
>
> ------------------------------
>
> Message: 12
> Date: Mon, 11 Apr 2005 10:32:26 -0400
> From: Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com>
> Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How
> does one shape RTP?
> To: asterisk-users at lists.digium.com
> Message-ID: <200504111032.27090.akohlsmith-asterisk at benshaw.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On April 11, 2005 10:08 am, Sean Kennedy wrote:
> > Honestly, the best script I've ever found is the wondershaper script (
> > google it ). I tried the correct one posted in this thread, tried
> > modifying it, but in the end I just used wondershaper.
>
> :-) I started out with wshaper and just didn't like it, which is where rc.tc
>
> came from.
>
> -A.
>
>
> ------------------------------
>
> Message: 13
> Date: Mon, 11 Apr 2005 16:48:27 +0200
> From: Ricardo Peironcely <rpr_listas at telefonica.net>
> Subject: Re: [Asterisk-Users] TDM400P power supply
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <425A8E3B.2010104 at telefonica.net>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Thanks,
>
> I will try with external power supply.
>
> Rpr
>
> Rich Adamson escribió:
>
> >>I've a problem with a TDM400P digium card.
> >>
> >>My box has no molex connectors for power supply. Simply has no any power
> >>connector, because is not a normal PC) And I need to know if i can use a
> >>external supply. But I've several questions:
> >>
> >>1.- Are both circuits (PCI-power and Phone-line-power) electrically
> >>separated?
> >>2.- A little voltage difference can create an undesired internal current?
> >>3.- What are the current needs for this supply?
> >>
> >>I need the power supply because I want to use both FXS and FXO ports.
> >>And I can't use a Y cable, because I've no molex connectors.
> >>
> >>
> >
> >Been discussed several times before and you should have found the
> >answer using google.
> >
> >The TDM connector is only used for the fxs modules, and then only the
> >+12 volt lead on that connector (and ground) is actually wired to
> >anything on the TDM board. So, there is no conflict with internal
> >system voltages.
> >
> >Yes you can use an external 12 volt power supply.
> >
> >The 12 volts is only used on the card to generate ringing voltage to
> >the fxs modules. No ringing, no significant current draw. Just about
> >any 12 volt supply should do, however I think I'd be looking for
> >one that is at least somewhat regulated. No other idea on the power
> >supply specs.
> >
> >
> >_______________________________________________
> >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: 14
> Date: Mon, 11 Apr 2005 20:52:03 +0600
> From: "Yusuf Iqbal" <yusii_bd at hotmail.com>
> Subject: [Asterisk-Users] Problem with X101P
> To: asterisk-users at lists.digium.com
> Message-ID: <BAY15-F4152F83923D0C75DB42882E9320 at phx.gbl>
> Content-Type: text/plain; charset="us-ascii"
>
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users/attachments/20050411/4930a523/attachment-0001.htm
>
> ------------------------------
>
> Message: 15
> Date: Mon, 11 Apr 2005 07:52:07 -0700
> From: "trixter http://www.0xdecafbad.com" <trixter at 0xdecafbad.com>
> Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How
> does one shape RTP?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1113231127.6650.16.camel at rufus.home.tld>
> Content-Type: text/plain; charset="us-ascii"
>
> On Mon, 2005-04-11 at 10:32 -0400, Andrew Kohlsmith wrote:
> > On April 11, 2005 10:08 am, Sean Kennedy wrote:
> > > Honestly, the best script I've ever found is the wondershaper script (
> > > google it ). I tried the correct one posted in this thread, tried
> > > modifying it, but in the end I just used wondershaper.
> >
> > :-) I started out with wshaper and just didn't like it, which is where
> rc.tc
> > came from.
> you may want to pull at least the RTP lines I just posted and add them
> to your rc.tc since that is what I got and tweaked since I use RTP :)
>
> --
> Trixter http://www.0xdecafbad.com
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users/attachments/20050411/c5e30ef3/attachment-0001.pgp
>
> ------------------------------
>
> Message: 16
> Date: Mon, 11 Apr 2005 09:49:17 -0500 (CDT)
> From: Dave Weis <djweis at sjdjweis.com>
> Subject: [Asterisk-Users] wcfxo problem
> To: asterisk-users at lists.digium.com
> Message-ID:
> <Pine.LNX.4.62.0504110947390.27314 at charmed.internetsolver.com>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
>
> I've got a X100P in a compaq proliant 3000. My system stops taking calls
> and making calls. I had been getting the FXO PCI Master abort before
> updating, I am now running a cvs head checkout from a week or so ago. Now
> I still have the problem but get more error messages:
>
> Found a Wildcard FXO: Wildcard X101P
> Registered tone zone 0 (United States / North America)
> Registered tone zone 0 (United States / North America)
> FXO PCI Master abort
> wcfxo: Out of space to write register 05 with 02
> wcfxo: Out of space to write register 05 with 03
> wcfxo: Out of space to write register 05 with 0a
> wcfxo: Out of space to write register 05 with 0a
> wcfxo: Out of space to write register 05 with 0a
> wcfxo: Out of space to write register 05 with 0a
>
> Any solution?
>
> --
> Dave Weis "I believe there are more instances of the abridgment
> djweis at sjdjweis.com of the freedom of the people by gradual and silent
> encroachments of those in power than by violent
> and sudden usurpations."- James Madison
>
>
> ------------------------------
>
> Message: 17
> Date: Mon, 11 Apr 2005 10:54:30 -0400
> From: "Robert Webb" <asterisk at ropeguru.com>
> Subject: [Asterisk-Users] (no subject)
> To: Asterisk-Users at lists.digium.com
> Message-ID: <web-1103722 at ropeguru.com>
> Content-Type: text/plain; charset="ISO-8859-1"; format="flowed"
>
>
> Good morning all..
>
> I was following a discussion on this list about the
> TDM400P revisions. It is my understanding that the current
> revision that one should have is the Rev. H and not the
> E/F. I have not yet been able to verify the rev stamped on
> the board, but zaptel is reporting that I have the Rev.
> E/F. I just bought this card in January direct from Digium
> and was wondering if I got the wrong Rev. somehow?? I have
> been having some intermittent problems but only thought it
> was my setup.
>
>
>
> ------------------------------
>
> Message: 18
> Date: Mon, 11 Apr 2005 09:57:16 -0500
> From: Doug Millsaps <asterisk at txpe.net>
> Subject: Re: [Asterisk-Users] Sipura SPA-841 Phone Review
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <6.1.2.0.2.20050411095336.033c5b20 at mail.txpe.net>
> Content-Type: text/plain; charset="us-ascii"; format=flowed
>
> I use a headset w/out any problems, except for if my cell phone is close by
> and rings. Otherwise, volume is ok and no humming. Could it be your
> headset?
>
> At 01:56 PM 4/10/2005, you wrote:
>
> >Just make sure you don't have a cordless or cell phone near by or the
> >headset jack will "receive" a considerable amount of interference into
> >your conversation (when NOT using a headset).
> >
> >Also don't even try using a headset... volume is low and there is a loud
> >humming noise.
>
>
>
> ------------------------------
>
> Message: 19
> Date: Mon, 11 Apr 2005 17:03:32 +0200
> From: "Bruno Hertz" <brrhtz at yahoo.de>
> Subject: Re: [Asterisk-Users] From OH323 to SIP or OH323 without
> gatekeeper
> To: asterisk-users at lists.digium.com
> Message-ID: <m3is2ttkxn.fsf at caruso.quasi.local>
> Content-Type: text/plain; charset=us-ascii
>
> "Joe S" <printingfoot at hotmail.com> writes:
>
> > Hi,
> >
> > I am new with asterisk. I was wondering if there is a way to call a
> > OH323 user or SIP user using Netmeeting/SJPhone with H323 as the
> > default protocol without having a gatekeeper.
> >
> > I can make a call from SIP to OH323 by specifying it in the
> > extensions.conf file, like:
> >
> > exten=>1001, 1, Dial(OH323/10.10.10.1)
> >
> > so I was wondering if there was a way to call from OH323 to SIP or OH323.
>
> Sure. Just specify in oh323.conf the context where incoming calls
> should go. That context then can include dial statements for any
> protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to
> setup dial plans.
>
> Finally, instruct your H323 phone to use asterisk as a gateway
> resp. proxy, not a gatekeeper. Any calls will then go through
> asterisk, and to the context you specified.
>
> I'm doing that with Gnomemeeting all the time, and it works without
> problems.
>
> Regards, Bruno.
>
>
>
> ------------------------------
>
> Message: 20
> Date: Tue, 12 Apr 2005 01:03:51 +1000 (EST)
> From: Sahil Gupta <sgupta at voicevalley.com.au>
> Subject: Re: [Asterisk-Users] wcfxo problem
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Pine.LNX.4.60.0504120103280.6587 at asterisk.in.com.au>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
> I'm having similar issues using an X100P Ambient Chipset Clone Card....
> any ideas?
>
> Regards,
>
>
> Sahil Gupta
> VoiceValley
>
> On Mon, 11 Apr 2005, Dave Weis wrote:
>
> >
> > I've got a X100P in a compaq proliant 3000. My system stops taking calls
> and
> > making calls. I had been getting the FXO PCI Master abort before updating,
> I
> > am now running a cvs head checkout from a week or so ago. Now I still have
>
> > the problem but get more error messages:
> >
> > Found a Wildcard FXO: Wildcard X101P
> > Registered tone zone 0 (United States / North America)
> > Registered tone zone 0 (United States / North America)
> > FXO PCI Master abort
> > wcfxo: Out of space to write register 05 with 02
> > wcfxo: Out of space to write register 05 with 03
> > wcfxo: Out of space to write register 05 with 0a
> > wcfxo: Out of space to write register 05 with 0a
> > wcfxo: Out of space to write register 05 with 0a
> > wcfxo: Out of space to write register 05 with 0a
> >
> > Any solution?
> >
> > --
> > Dave Weis "I believe there are more instances of the
> abridgment
> > djweis at sjdjweis.com of the freedom of the people by gradual and silent
> > encroachments of those in power than by violent
> > and sudden usurpations."- James Madison
> > _______________________________________________
> > 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
> >
>
>
> ------------------------------
>
> Message: 21
> Date: Mon, 11 Apr 2005 11:11:45 -0400
> From: "Robert Webb" <asterisk at ropeguru.com>
> Subject: Re: [Asterisk-Users] TDM400P Revision question.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <web-1103721 at ropeguru.com>
> Content-Type: text/plain; charset="ISO-8859-1"; format="flowed"
>
> Sorry for the initial no subject line. Was in a hurry when
> I typed this and somehow missed putting it in.
>
> Please accept my apologies....
>
> On Mon, 11 Apr 2005 10:54:30 -0400
> "Robert Webb" <asterisk at ropeguru.com> wrote:
> >
> > Good morning all..
> >
> > I was following a discussion on this list about the
> >TDM400P revisions. It is my understanding that the
> >current revision that one should have is the Rev. H and
> >not the E/F. I have not yet been able to verify the rev
> >stamped on the board, but zaptel is reporting that I have
> >the Rev. E/F. I just bought this card in January direct
> >from Digium and was wondering if I got the wrong Rev.
> >somehow?? I have been having some intermittent problems
> >but only thought it was my setup.
> >
> > _______________________________________________
> > 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
>
>
>
> ------------------------------
>
> Message: 22
> Date: Mon, 11 Apr 2005 11:13:45 -0400
> From: "Bobby Lacey" <asterisk at duaneallman.net>
> Subject: [Asterisk-Users] Intercom with Aastra 480e?
> To: <asterisk-users at lists.digium.com>
> Message-ID: <006e01c53ea9$0dfe4550$0e00000a at rosehill>
> Content-Type: text/plain; charset="us-ascii"
>
> Hello list,
>
> I have been successful in setting up my first * box with a pair of
> x100p's, Cisco 7960, and a Digium iAXy.
>
> I would like to incorporate an Aastra 480e using my iAXy and ADSI. I
> want to be able to answer phone calls with my 7960 in the back of the
> house and park the call, then in turn call the intercom on the 480e in
> the front (using two way audio) to announce that there is a call that
> needs to be picked up on 701.
>
> Also, by using the Aastra 480e, can I see my Zap line status to see what
> lines are available and also if extensions are in use?
>
> Thanks in advance.
>
> B. Lacey
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> ------------------------------
>
> Message: 23
> Date: Mon, 11 Apr 2005 11:16:56 -0400
> From: Matt <mhoppes at gmail.com>
> Subject: [Asterisk-Users] Manipulate Asterisk Database from manager?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <c11d025305041108165384be0d at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi,
> Is there anyway to manipulate the asterisk internal database from the
> manager (the one you can telnet to)? And if so.. how does one do it?
> (ie for enabling call forwarding, etc)
>
>
> ------------------------------
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> End of Asterisk-Users Digest, Vol 9, Issue 93
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