[asterisk-users] Cisco 79xx logon/logoff

[Adrian Marsh]

Hi All,

I'd like to know if anyone has figured out a way to be able to have
users logon/logoff manually from Cisco 79xx phones (with SIP firmware
loaded)?
Scenario is, user walks into office, sits at a random desk, and logs
onto the phone. The system would need to "log them off" of the last
hardphone they were on, and then configure the new phone for their
extension.

We're creating hotdesks and it would be good if users could logon/logoff
the desk phones.

At present they all use softphones on the PC too, and I could engineer a
way of maybe doing things via cgi scripting, replacing the tftp config
files for that phone, and then remotely resetting the phone, however
that would be quite clumber sum.
And before I go that route, I wondered if any of the commercial A*k
systems already offer this?

If the Ciscos can't do this.. then can any other hardphones?


Adrian Marsh

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Re: [asterisk-users] Video Conference

[Dovid B]

Dean,
As you know Asterisk is primarily for telephony (yes we have fun with it 
controlling our light's, rebooting servers etc). Video conferencing is a 
completely different game. IMHO it does not make sense to build video 
conferencing for asterisk since lots of people that need video conferencing 
do not need the telephone side of it. It makes more sense to have a video 
solution that plays nice with asterisk.

- Original Message - 
From: "Dean Collins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, October 24, 2007 12:55 AM
Subject: Re: [asterisk-users] Video Conference


> Good to hear someone is using WiredRed.
>
> I suggested that as an alternative several times on this list but to be
> honest I'm still astounded that there isn't an asterisk alternative.
>
>
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +61-2-9016-5642 (Sydney in-dial).
>
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Patrick Davis
>> Sent: Tuesday, 23 October 2007 3:49 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Video Conference
>>
>> we use WiredRed with some success.
>>
>> I've tried dimdim and it was ok, but not as good as WiredRed. with
>> WiredRed its still going to cost $3k plus per year.  it'll do the
>> video, voice and desktop sharing. decent video and audio.
>>
>> Patrick
>>
>> Patrick Davis
>> Study Abroad Canada
>> P.O. Box 3231
>> 51 Univeristy Ave.
>> Charlottetown, PE Canada
>> C1A 7N9
>> Tel: 902-628-2379
>> Fax: 902-892-1198
>> www.studyincanada.ca
>> [EMAIL PROTECTED]
>>
>>
>>
>> On 23-Oct-07, at 6:18 AM, Dovid B wrote:
>>
>> > 
>> >> Thanks for the responce.  Have you had any luck at all even with
>> >> what one
>> >> might not consider straight forward?  I am trying to avoid paying
> the
>> >> $1000+
>> >> per location needed to purchase something from say Polycom or
>> >> Tandberg.  I
>> >> would even be willing to do something along the lines of a web app
>> >> for
>> >> video
>> >> and some how tie that together with the voice through Asterisk.
> Just
>> >> don't
>> >> want to look like one of the old dubbed over Japanese movies from
>> >> when I
>> >> was
>> >> a kid (lips move and then a couple seconds later you hear voice).
>> >> JohnM
>> >
>> > John,
>> > Try contacting [EMAIL PROTECTED] They have some
>> > solution there
>> > that works with Asterisk.
>> >
>> >
>> >
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[asterisk-users] PRI span configuration - span remains down

[David Kennedy]

Hi,

I'm trying to connect to Telewest/Virgin Media with a TE110P using
asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always
appears as

PRI span 1/0: Provisioned, Down, Active

My zapata.conf is currently
---
[channels]
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
switchtype=euroisdn
contect=from-pri
signalling=pri_cpe
group=1
channel => 1-15
channel => 17-31
---

zaptel.conf is
---
span=1,1,0,ccs,hdb3,crc4
dchan=16
bchan=1-15,17-31
loadzone=uk
defaultzone=uk
---

I'm in London and the server is in Manchester, so I can't look at the
server directly, but when we first started setting it up, apparently a
pair of cables were the wrong way round, so the card was in a RED
alarm state. We've switched the cables and now the card is OK. We did
have a lot of IRQ misses, so we've upgraded the kernel and now the
accuracy reported by zttest is about 99.98%. Telewest have checked the
line for faults and have reported that it's fine, but I just can't get
it working.

Does anyone have any ideas/suggestions?

Thanks,

Dave

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Re: [asterisk-users] Remote provisioning for ATA's

[Rizwan Hisham]

:) well i was expecting this type of reply and have been preparing my mind
to do what you said. But need a little help. i have a lttle idea about how
remote provisioning works, we have to send the data using xml. and we can
make an application which can do it in any language. But i dont know how
should i format the data in xml for different ATAs. and there is lots of
other things which im concerned about. So can you help me and give me
starter kit type of thing.

And to other guys: Still waiting to hear from you. If you know any good
provisioning system then plz tell me about it.

On 10/24/07, Matt <[EMAIL PROTECTED]> wrote:
>
> Your best bet may be to write your own.  That's what we ended up doing and
> it isn't that hard.
>
> On 10/24/07, Rizwan Hisham < [EMAIL PROTECTED]> wrote:
>
> > Hi all,
> > I need a fully developed web based remote provisioning system. I cant
> > find anything reliable on the internet. Have already checked
> > ataconfig.com  and voxilla-ays.com. have tried to contact them but got
> > no response. So if anybody knows a good provisioning system then plz tell me
> > about it.
> >
> > --
> > Best Regards
> > Rizwan Hisham
> > Software Engineer
> > Axvoice Inc.
> > www.axvoice.com
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> >
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
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-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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Re: [asterisk-users] AstManProxy Host Prefix?

[Steve Davies]

I can look at adding a server-filter parameter to astmanproxy.users
(no promises on timescale though!) as I wrote the per-user filtering
in the first place.

My problem with astmanproxy at the moment is that I don't get any
responses from the maintainer (Dave at popvox?). I have a couple of
patches that I have emailed with no response, IIRC one of which is a
potential crash bug. The other fixes the use of ActionID: with login
attempts, which is not uncommon.

Cheers,
Steve

On 10/24/07, asterisk <[EMAIL PROTECTED]> wrote:
> What would be nice if it you could specify the host per user in
> astmanproy.users
> Anyone interested in making the change? $$$
>
> Doug
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Richard
> Lyman
> Sent: Wednesday, October 24, 2007 1:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] AstManProxy Host Prefix?
>
> Douglas Garstang wrote:
> > Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that
> output applies to, to the start of each line? If you are proxying
> multiple systems, how can it uniquely identify the output from each
> system?
> >
> > Thanks,
> > Doug.
> >
> >
> each Event block should have a
>
> Server: .
>
> appended to it.
>

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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

[Steve Davies]

Most of thread snipped.

On 10/24/07, marcotasto <[EMAIL PROTECTED]> wrote:
>
> Some days ago I've sent to David Rowe a little patch that preserves the echo 
> cancel
> status between calls.

Surely this is only appropriate where you have a local analogue device
that is unchanging - If you retained the EC status between calls on a
trunk line (where the far end will be different for each call), the
chances are that it would be inappropriate.

I assume you enable/disable this feature per-channel? Or perhaps I am
misunderstanding the feature you describe?

Thanks,
Steve

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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

[Alan Lord]

Steve Davies wrote:
> Most of thread snipped.
> 
> On 10/24/07, marcotasto <[EMAIL PROTECTED]> wrote:
>> Some days ago I've sent to David Rowe a little patch that preserves the echo 
>> cancel
>> status between calls.
> 
> Surely this is only appropriate where you have a local analogue device
> that is unchanging - If you retained the EC status between calls on a
> trunk line (where the far end will be different for each call), the
> chances are that it would be inappropriate.
> 
> I assume you enable/disable this feature per-channel? Or perhaps I am
> misunderstanding the feature you describe?
> 
> Thanks,
> Steve
> 

I had the same thoughts as you on this but assumed I'd just 
misinterpreted what Marco said...

I can't really see why you'd want it to NOT re-converge on each call. 
Analogue is, after all, not a precise and unchanging phenomenon.



Alan


-- 
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Re: [asterisk-users] Compatibility Issues with dell poweredge 195 and TE110P card

[Brian Hutchinson]

You guys are scaring me! I'm building a 2950 with SAS RAID 5 on the new PERC
and it will have 2 TE420P's.  I hope it works or my bacon will fry.

On 10/25/07, Joseph Begumisa <[EMAIL PROTECTED]> wrote:
>
> >
> > Has anyone had any compatibility issues with a TE110P card installed
> > on a Dell Poweredge 1950?I noted the following error on the LCD
> > display of the Dell Poweredge 1950:
> >
> >
> >
> > E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
>
> >Yes, I have had this problem with a dell PE1650, 1850, SC1400, and PE650.
> I
> have a TE410P that does it. It may not be wise, but I just ignore the
> orange
> blinking LCD display (or light, >depending on the model). I did try
> reseating the card, and it "works" for a few weeks, and then goes back to
> the same old thing.
>
> Yes, that happened too.  Digium has graciously offered to send me a TE120P
> with the Digium VoiceBus technology which I will test out on the Dell
> 1950.
> Will post my findings thereafter.
>
> Joseph.
>
>
>
>
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Re: [asterisk-users] AstManProxy Host Prefix?

[Steve Davies]

Utterly untested, but here goes with the server-filtering parameter...
The attached patch should apply to version the 1.21 tarball cleanly,
and includes all my other changes which haven't made it into the main
astmanproxy code.

Please do feed-back on whether this works (it compiles :-) ).

Thanks,
Steve


astmanproxy-1.21-1.21c.patch.gz
Description: GNU Zip compressed data
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[asterisk-users] sip reload causes unreachable

[Admin DeryTelecom]

Hi

I have a asterisk with many phones (type=friend)

When I issue the command "sip reload" some of the phones become unreachable 
and they come back just after.

I guess that the sip.conf file is too big and asterisk takes too much time 
reloading the entire file.

Is there a way to avoid this probleme or another way to add/remove sip 
phones dynamically ?

Patrick


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[asterisk-users] Large voicemail

[Pepo]

I am trying to use Asterisk as the voicemail system of the TELCO where I work. 
I wanna test with 2 mail boxes ( and later with a better machine/server I 
hope try with 7 ).

How do I include in voicemail.conf the file with the mail boxes?, In a big 
system like this,is better use text files or any database? 

Thanks

-- 

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Re: [asterisk-users] PRI span configuration - span remains down

[Rony Ron]

Hello,
Quoting Digium Support:
"The TE110P has been discontinued and replaced in our product lineup with
the TE120P, which features many overall improvements and does not suffer
from the HDLC Abort/Bad FCS problems that the TE110P did."

Better switch to TE120P,

On 10/25/07, David Kennedy <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I'm trying to connect to Telewest/Virgin Media with a TE110P using
> asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always
> appears as
>
> PRI span 1/0: Provisioned, Down, Active
>
> My zapata.conf is currently
> ---
> [channels]
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=yes
> switchtype=euroisdn
> contect=from-pri
> signalling=pri_cpe
> group=1
> channel => 1-15
> channel => 17-31
> ---
>
> zaptel.conf is
> ---
> span=1,1,0,ccs,hdb3,crc4
> dchan=16
> bchan=1-15,17-31
> loadzone=uk
> defaultzone=uk
> ---
>
> I'm in London and the server is in Manchester, so I can't look at the
> server directly, but when we first started setting it up, apparently a
> pair of cables were the wrong way round, so the card was in a RED
> alarm state. We've switched the cables and now the card is OK. We did
> have a lot of IRQ misses, so we've upgraded the kernel and now the
> accuracy reported by zttest is about 99.98%. Telewest have checked the
> line for faults and have reported that it's fine, but I just can't get
> it working.
>
> Does anyone have any ideas/suggestions?
>
> Thanks,
>
> Dave
>
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Re: [asterisk-users] Dedicated Codec Conversion Server

[Steve Totaro]

Um, yeah, the part you suggest is obvious.

What about having two different SIP accounts for each phone, I guess I 
need to do one for inbound and one for outbound?  Different extensions 
or whatever.

Thanks,
Steve

Klaverstyn, David C wrote:
> The way I would accomplish this is to have 2 Asterisk boxes.  Your
> conversion server would just have a dial plan to forward all calls to
> the Asterisk box that has the PSTN interface.  Once the PSTN Asterisk
> Server receives the calls it just routes the call based on dial plan
> rules.
>
> {Internet -> VPN} -> Conversion Server -> Asterisk PSTN.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steve
> Totaro
> Sent: Thursday, 25 October 2007 1:54 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Dedicated Codec Conversion Server
>
> I have a need for a large number of remote phones.  I want to use GSM 
> between the phones and the conversion server which will transcode to 
> ulaw eventually send or recieve calls via the PSTN (ulaw).
>
> I am curious is anyone has any ideas on the easiest way to create a 
> dedicated codec conversion box.  It will be running openvpn and so will 
> the remote PCs with softphones (x-lite).
>
> So I want the remote softphones to connect to the codec conversion 
> asterisk box and then send the call to the main Asterisk server as ulaw 
> and pass call in and out the pstn as ulaw.
>
> Any ideas for a simple implementation without creating all kinds of 
> funky conf files.  Seems simple but the solution eludes me (maybe 
> because I have been working over 18 hours.
>
> Thanks,
> Steve Totaro
>
>   


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Re: [asterisk-users] Asterisk under VMWare

[George Pajari]

Chris Bagnall wrote:
>> Our testing has yielded pretty good results. We had 10 simultaneous
>> calls with ulaw and quality was very good. We are pure VOIP also.
>> 
> How many VMs were you running at the time, and what load were they under?
>
> We've setups running between 3 and 5 VMs on a single box (multi-core, lots of 
> RAM, etc.) and we haven't had any problems with them. Would be interesting to 
> know how well it'll scale with more VMs on each box
>   

Also, the number of successful Asterisk VMs will depend on the 
virtualisation technology used. The literature strongly suggests that 
Xen will work better than VMware, OpenVZ better than Xen, and Virtuozzo 
better than OpenVZ. YMMV.

We offer (as a business) Virtual Private Asterisk Servers (see 
www.vpas.ca) and have no problem running many more than 3 to 5 virtual 
Asterisk instances on a single server in commercially demanding 
environments.

-- 
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
   www.netvoice.ca  www.ip-centrex.ca  www.ip-pbx.ca  www.vpas.ca
 www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102) 


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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

[Brandon Black]

On 10/24/07, Alan Lord <[EMAIL PROTECTED]> wrote:
> And anyone who has echo problems with x100p or other analogue cards
> should really give this a try. Most of the experiences I have read about
> have been very positive. Mine also :-)
>

Any in case anyone's wondering if it's too CPU intensive for little
embedded Asterisk devices, here's the speedtest results on a Soekris
net5501 (500Mhz Geode LX CPU):

Testing OSLEC with 128 taps (16 ms tail)
CPU executes 500.20 MIPS
-

Method 1: gettimeofday() at start and end
  671 ms for 10s of speech
  33.56 MIPS
  14.90 instances possible at 100% CPU load
Method 2: samples clock cycles at start and end
  33.56 MIPS
  14.90 instances possible at 100% CPU load
Method 3: samples clock cycles for each call, IIR average
  cycles_worst 64273 cycles_last 3849 cycles_av: 4018
  32.14 MIPS
  15.56 instances possible at 100% CPU load

-- Brandon

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Re: [asterisk-users] PRI span configuration - span remains down

[David Kennedy]

Hi, thanks for the quick reply

I've literally just got off the phone with a Telewest engineer - after
being told the line was ok yesterday, I've been told it wasn't
actually turned on. D'oh!

If need be I do have a spare TE120P in a box in my desk drawer, so I
can send that to Manchester to be installed.

Thanks

Dave.



On 10/25/07, Rony Ron <[EMAIL PROTECTED]> wrote:
> Hello,
> Quoting Digium Support:
> "The TE110P has been discontinued and replaced in our product lineup with
> the TE120P, which features many overall improvements and does not suffer
> from the HDLC Abort/Bad FCS problems that the TE110P did."
>
> Better switch to TE120P,
>
>
> On 10/25/07, David Kennedy <[EMAIL PROTECTED]> wrote:
> >
> > Hi,
> >
> > I'm trying to connect to Telewest/Virgin Media with a TE110P using
> > asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always
> > appears as
> >
> > PRI span 1/0: Provisioned, Down, Active
> >
> > My zapata.conf is currently
> > ---
> > [channels]
> > echocancel=yes
> > echocancelwhenbridged=no
> > echotraining=yes
> > switchtype=euroisdn
> > contect=from-pri
> > signalling=pri_cpe
> > group=1
> > channel => 1-15
> > channel => 17-31
> > ---
> >
> > zaptel.conf is
> > ---
> > span=1,1,0,ccs,hdb3,crc4
> > dchan=16
> > bchan=1-15,17-31
> > loadzone=uk
> > defaultzone=uk
> > ---
> >
> > I'm in London and the server is in Manchester, so I can't look at the
> > server directly, but when we first started setting it up, apparently a
> > pair of cables were the wrong way round, so the card was in a RED
> > alarm state. We've switched the cables and now the card is OK. We did
> > have a lot of IRQ misses, so we've upgraded the kernel and now the
> > accuracy reported by zttest is about 99.98%. Telewest have checked the
> > line for faults and have reported that it's fine, but I just can't get
> > it working.
> >
> > Does anyone have any ideas/suggestions?
> >
> > Thanks,
> >
> > Dave
> >
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> >
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> >
>
>
>
> --
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Re: [asterisk-users] sip reload causes unreachable

[Atis Lezdins]

On Thursday 25 October 2007 15:36:44 Admin DeryTelecom wrote:
> Hi
>
> I have a asterisk with many phones (type=friend)
>
> When I issue the command "sip reload" some of the phones become unreachable
> and they come back just after.
>
> I guess that the sip.conf file is too big and asterisk takes too much time
> reloading the entire file.
>
> Is there a way to avoid this probleme or another way to add/remove sip
> phones dynamically ?

Realtime?

http://www.voip-info.org/wiki-Asterisk+RealTime+Sip

Regards,
Atis

-- 
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835

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Re: [asterisk-users] Dedicated Codec Conversion Server

[Steven]

Will "All" inbound calls go to SIP phones on the conversion server?

If so, you just need a .X extension that forwards all inbound calls to the 
second server.

But, thinking about this, it would appear that the second "conversion" server 
will need most of your dialplan, and as such may still 
be affected by codec translation load.

Is that what you are trying to avoid?


I am looking at a similar design, but my goal is not to offload the codec 
translation, but to offload the TDM functions.
This seems like it will be quite easy. as the server with TDM function will 
have a very simple dialplan.
But, my "extension" server will be just as complex as it is today. (but IP only)

I am not sure you can offload codec translation without a very complex design.



-- 
-- 
Steven

http://www.glimasoutheast.org



"Steve Totaro" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> Um, yeah, the part you suggest is obvious.
>
> What about having two different SIP accounts for each phone, I guess I
> need to do one for inbound and one for outbound?  Different extensions
> or whatever.
>
> Thanks,
> Steve
>
> Klaverstyn, David C wrote:
>> The way I would accomplish this is to have 2 Asterisk boxes.  Your
>> conversion server would just have a dial plan to forward all calls to
>> the Asterisk box that has the PSTN interface.  Once the PSTN Asterisk
>> Server receives the calls it just routes the call based on dial plan
>> rules.
>>
>> {Internet -> VPN} -> Conversion Server -> Asterisk PSTN.
>>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Steve
>> Totaro
>> Sent: Thursday, 25 October 2007 1:54 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] Dedicated Codec Conversion Server
>>
>> I have a need for a large number of remote phones.  I want to use GSM
>> between the phones and the conversion server which will transcode to
>> ulaw eventually send or recieve calls via the PSTN (ulaw).
>>
>> I am curious is anyone has any ideas on the easiest way to create a
>> dedicated codec conversion box.  It will be running openvpn and so will
>> the remote PCs with softphones (x-lite).
>>
>> So I want the remote softphones to connect to the codec conversion
>> asterisk box and then send the call to the main Asterisk server as ulaw
>> and pass call in and out the pstn as ulaw.
>>
>> Any ideas for a simple implementation without creating all kinds of
>> funky conf files.  Seems simple but the solution eludes me (maybe
>> because I have been working over 18 hours.
>>
>> Thanks,
>> Steve Totaro
>>
>>
>
>
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[asterisk-users] CISCO 7921G with asterisk

[Jordi Guiu]

Any one have experience with this CISCO Wireless IP phone running with 
Asterisk??

It doesn't support SIP protocol I believe, so I need to know if the 
skinny channel can work with the 7921.

Thanks for help.

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Re: [asterisk-users] Remote provisioning for ATA's

[[EMAIL PROTECTED]]

I'm pretty sure, just like Voicepulse service, it won't work, but its
worth a shot, no? Its free...

http://sourceforge.net/projects/vgps/

On 10/24/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote:
> Hi all,
> I need a fully developed web based remote provisioning system. I cant find
> anything reliable on the internet. Have already checked ataconfig.com  and
> voxilla-ays.com. have tried to contact them but got no response. So if
> anybody knows a good provisioning system then plz tell me about it.
>
> --
> Best Regards
> Rizwan Hisham
> Software Engineer
> Axvoice Inc.
> www.axvoice.com
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[asterisk-users] Advanced Dial Plan

[Frederico Madeira]

Hi Guys,

I Have this peers on my sip.conf

[provider-302333-3000]
type=friend
context=provider
secret=xpto
username=302000
host=sip.provider.com
fromuser=302000
insecure=very
canreinvite=no


[provider-30-3001]
type=friend
context=provider
secret=xpto
username=303001
host=sip.provider.com
fromuser=303001
insecure=very
canreinvite=no


I Have in my sip.conf two extension 3000 and 3001.

I have this rule in my extensions.conf

exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten=> _X.,2,Hangup

exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten=> _X.,2,Hangup


And every calls made by my both extension was using the first rule, so
calls from  extension 3000 match with peer and work, but calls from
3001 didn't match with peer and I got error.

How can I use a conditional sentence like:

if {${CALLERID(num)}=3000)
{
exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten=> _X.,2,Hangup
}
else
if {${CALLERID(num)}=3001)
{
exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
exten=> _X.,2,Hangup
}

Thanks.


-- 
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br

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[asterisk-users] Transfer and 407's

[Paul Campbell]

I have a SIP voice server which I want to place an Asterisk server in
front of to handle call routing.

 

At the moment I can call the apps on the server fine, but it cannot
transfer to another extension via asterisk.

 

Even attempting a call from the voice server to an asterisk extension
results in a "407 Proxy Authorisation Required" error on Asterisk and a
bridge_reject "noautho"

 

The server is registering fine with Asterisk.

 

Some debug stuff:

 

=

sip.conf

[general]

context=internal

srvlookup=yes

 

[paul]

type=friend

secret=removed

qualify=no

nat=no

host=dynamic

canreinvite=yes

context=internal

 

[test]

type=friend

secret=removed

qualify=no

nat=no

host=dynamic

canreinvite=yes

context=internal

 

 [vgp]

type=friend

secret=removed

qualify=no

nat=no

host=10.0.2.136

canreinvite=yes

context=internal

insecure=very

permit=10.0.2.136

 

extensions.conf

[internal]

 exten => 101,1,Dial(SIP/paul)

exten => 103,1,Dial(SIP/test)

exten => 202,1,Dial(SIP/[EMAIL PROTECTED]
 )

 

SIP/[EMAIL PROTECTED]
  is a voice app
which simply attempts to transfer the call to: "[EMAIL PROTECTED]".

 

SIP Error:

 

<--- Reliably Transmitting (no NAT) to 10.0.2.136:5060 --->

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP
10.0.2.136:5060;branch=z9hG4bKb675eb1820a7c8;received=10.0.2.136

From: sip:[EMAIL PROTECTED];tag=C1431100-8D08-7D26-6246-3AB1F318B851

To: ;tag=as4c65adb2

Call-ID: [EMAIL PROTECTED]

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="6e23a069"

Content-Length: 0

 

 

I'd be very grateful for any suggestions.

 

Thanks,

 

Paul

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 




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Re: [asterisk-users] Cisco 79xx logon/logoff

[James FitzGibbon]

On 10/25/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:

> I'd like to know if anyone has figured out a way to be able to have
> users logon/logoff manually from Cisco 79xx phones (with SIP firmware
> loaded)?
> Scenario is, user walks into office, sits at a random desk, and logs
> onto the phone. The system would need to "log them off" of the last
> hardphone they were on, and then configure the new phone for their
> extension.
>
> We're creating hotdesks and it would be good if users could logon/logoff
> the desk phones.
>
> At present they all use softphones on the PC too, and I could engineer a
> way of maybe doing things via cgi scripting, replacing the tftp config
> files for that phone, and then remotely resetting the phone, however
> that would be quite clumber sum.
> And before I go that route, I wondered if any of the commercial A*k
> systems already offer this?


I haven't done this with Cisco specifically, but I have done it with other
hardphones.  I didn't go the route of updating the phone's config while the
person was at that desk because I didn't have a reliable way to remotely
restart the phone.

Instead, I gave each phone an extension and dynamically created links in
ASTdb between the person's extension to the phone's extension for the
duration of the login session.  All the hot desk phones are in a context
that when nobody is linked to that desk only allows you to logon and do
basic things like call the operator and emergency services.  If a user is
linked to the desk, then I do a
Goto(proper-context-for-that-user,${EXTEN},1), which gives me dynamic
contexts for outbound calling without having to have my sip users in
realtime.

The downside to this approach is MWI, but all of my users get voicemail via
email with delete-on-send enabled, so I just kind of sidestepped the issue.

I'm not sure if you're specifically asking about using the softkeys to do
this without going through some kind of IVR application; worst case you have
speed-dials to your logon/logoff extensions.  You could even save a key by
having a single logon/logoff extension that changed its behaviour based on
whether a link exists for that desk.  Then again, if someone forgets to log
off it might be easier for users to have a dedicated 'logoff' button rather
than having to press the dual-purposed key twice.

Hope that gives you some ideas.

-- 
j.
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Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

[marcotasto]

Hi Steve.
What I did was to allocate one EC instance the first time a channel asks for it 
and reuse the same memory area for the same channel every time a new call is 
coming.
The memory is then freed when the channel is unloaded.

I've done this with my TDM400P in mind and I don't know what could be wrong in 
other setups. What I'm thinking is that, probably, the old EC should converge 
faster than a new one because the external conditions (even if different) are 
slightly similar.

I'm open to every possible suggestion and solution (compatible with my setup 
and my spare time).

Thank you and bye,
Marco.


> On 10/24/07, marcotasto <[EMAIL PROTECTED]> wrote:
> >
> > Some days ago I've sent to David Rowe a little patch that preserves the
> echo cancel
> > status between calls.
> 
> Surely this is only appropriate where you have a local analogue device
> that is unchanging - If you retained the EC status between calls on a
> trunk line (where the far end will be different for each call), the
> chances are that it would be inappropriate.
> 
> I assume you enable/disable this feature per-channel? Or perhaps I am
> misunderstanding the feature you describe?
> 
> Thanks,
> Steve
> 


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Re: [asterisk-users] Advanced Dial Plan

[Philipp Kempgen]

Frederico Madeira wrote:

> I Have in my sip.conf two extension 3000 and 3001.
> 
> I have this rule in my extensions.conf
> 
> exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
> exten=> _X.,2,Hangup
> 
> exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
> exten=> _X.,2,Hangup
> 
> 
> And every calls made by my both extension was using the first rule, so
> calls from  extension 3000 match with peer and work, but calls from
> 3001 didn't match with peer and I got error.


exten=> 3000,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
exten=> 3000,n,Hangup()

exten=> 3001,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
exten=> 3001,n,Hangup()

That dialplan is about as easy as it can get. :)

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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[asterisk-users] Mantis 10659 - Make it configurable?

[James Texter]

Hello listers,
I went to pull some CDR's from my PBX, and noticed they were a bit
light.  I also noticed output on the console about CDR's not being
posted.  I am currently running 1.4.13, and in looking at the change
log, this was a change in behavior as part of mantis 10659.  Personally,
I think the old behavior was more correct, but obviously at least one
person disagrees.  I think it's worth making it configurable so that the
old behavior can be restored for those of us who would like to have it.
If others agree, I'll work on getting a patch put together to make this
happen.  Thoughts?

Thanks,

James



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Re: [asterisk-users] Advanced Dial Plan

[Frederico Madeira]

Philipp

This didn't wotk.

Let's suppose that my sip extension 3000 want to call to (302).123.3211
I need a rule in extensions.conf to match with this number, right ?
So, I can't use rules that you advice.

My problem is only for outbound calls.

-- 
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br



2007/10/25, Philipp Kempgen <[EMAIL PROTECTED]>:
> Frederico Madeira wrote:
>
> > I Have in my sip.conf two extension 3000 and 3001.
> >
> > I have this rule in my extensions.conf
> >
> > exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
> > exten=> _X.,2,Hangup
> >
> > exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
> > exten=> _X.,2,Hangup
> >
> >
> > And every calls made by my both extension was using the first rule, so
> > calls from  extension 3000 match with peer and work, but calls from
> > 3001 didn't match with peer and I got error.
>
>
> exten=> 3000,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
> exten=> 3000,n,Hangup()
>
> exten=> 3001,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
> exten=> 3001,n,Hangup()
>
> That dialplan is about as easy as it can get. :)
>
> Regards,
>   Philipp Kempgen
>
> --
> amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
> Let's use IT to solve problems and not to create new ones.
>   Asterisk? -> http://www.das-asterisk-buch.de
>
> Geschäftsführer: Stefan Wintermeyer
> Handelsregister: Neuwied B 14998
>
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Re: [asterisk-users] Advanced Dial Plan

[Brett Crapser]

On Thu, 25 Oct 2007, Frederico Madeira wrote:
> Philipp
>
> This didn't wotk.
>
> Let's suppose that my sip extension 3000 want to call to (302).123.3211
> I need a rule in extensions.conf to match with this number, right ?
> So, I can't use rules that you advice.
>
> My problem is only for outbound calls.
>
> 2007/10/25, Philipp Kempgen <[EMAIL PROTECTED]>:
>> Frederico Madeira wrote:
>>
>>> I Have in my sip.conf two extension 3000 and 3001.
>>>
>>> I have this rule in my extensions.conf
>>>
>>> exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
>>> exten=> _X.,2,Hangup
>>>
>>> exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
>>> exten=> _X.,2,Hangup
>>>
>>>
>>> And every calls made by my both extension was using the first rule, so
>>> calls from  extension 3000 match with peer and work, but calls from
>>> 3001 didn't match with peer and I got error.
>>
>>
>> exten=> 3000,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
>> exten=> 3000,n,Hangup()
>>
>> exten=> 3001,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
>> exten=> 3001,n,Hangup()
>>
>> That dialplan is about as easy as it can get. :)
>>
>> Regards,
>>   Philipp Kempgen

Okie dokie

[outbound]
exten=> _X.,1,GotoIf([${CALLERID(num)} == "3000"]?path0|1)
exten=> _X.,2,GotoIf([${CALLERID(num)} == "3001"]?path1|1)
exten=> _X.,3,Playback(tt-monkeys)
exten=> _X.,4,Hangup

[path0]
exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
exten=> _X.,2,Hangup

[path1]
exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
exten=> _X.,2,Hangup

If you hear the monkeys...

Brett

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Re: [asterisk-users] Advanced Dial Plan

[John Faubion]

> Let's suppose that my sip extension 3000 want to call to (302).123.3211
> I need a rule in extensions.conf to match with this number, right ?

Let me see if I have this correct. You want to use the
"provider-302333-3000" for any call going out from 3000 and
"provider-30-3001" for any call going out from 3001. Basically a one to
one mapping of extension to trunk, right?

Try something like this...

===

exten=> _X./3000.,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
exten=> _X./3000.,2,Hangup

exten=> _X./3001,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
exten=> _X./3001,2,Hangup

===

This way if extension 3000 makes a call it uses the first one and if 3001
makes a call it uses the second one.

John


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[asterisk-users] [Fwd: [asterisk-dev] Realtime Queues]

[Steve Totaro]

Someone posted to the wrong list, thought I would help out.

Thanks,
Steve
--- Begin Message ---
I have heard that they have been problems with Realtime Queus in
Asterisk.  What are the problems?  Do they still exist?  Do they exist
in the most current version of 1.2?  Has anyone been able to use this
technology reliably in a high call volume environment (Over 1000 calls
per day)?
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[asterisk-users] How install chan_unicall.so!!

[sistemas]

Hi, How install chan_unicall.so in Asterisknow??

Thanks!
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Re: [asterisk-users] [asterisk-biz] Polycom Provisioning Tool

[David Boyd]

Michael,  

way cool. 

Works in WINE also :)

db

On Wed, 2007-10-24 at 23:09 -0400, Michael Munger wrote:
> Not sure if one exists, but someone had asked me for this a while ago.
> Here it is! My Polycom Provisioning Tool. Notice the version is 0.0.1.
> Just a concept program (but it works well).
> 
>  
> 
> I am open for suggestions, feature additions, and bug fixes. Email me
> with any requests. I want to improve this to make it really useful for
> the community, so let me know what you think.
> 
>  
> 
> http://www.wintrisk.com/ppt.html
>  
> 
>  
> 
> Michael Munger, dCAP
> 
> High Powered Help, Inc
> 
> [EMAIL PROTECTED]
> 
> 404-438-2128 x 101
> 
>  
> 
> 
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[asterisk-users] T.38 Faxing and Asterisk

[Paul Bryson]

I understand that Asterisk 1.4 should support T.38 pass-through, but I 
need Asterisk (or something on the Asterisk box) to act as a T.38 
endpoint.  Judging from the unclaimed $12,000USD bounty, it doesn't 
appear that Asterisk itself can do this.
http://www.voip-info.org/wiki-Asterisk+T.38+Bounty

Does anyone have any experience with this, or are able to point to an 
example of this working?

It looks like I need something along the lines of Asterisk detecting an 
incoming fax, passing it to IAXmodem (or something, I don't know what 
could do it) for conversion to T.38, and then the T.38 getting passed on 
to an appropriate T.38 supporting ATA (or other endpoint).

Or at least I think that's what we need to support these scenarios.

Scenario 1:
1. Somebody sends us a fax, and is an analog signal until it gets into 
the Telco's network where it is translated to digital audio.
2. The call comes in over our T1 to our Asterisk box.
3. Asterisk box answers and converts the incoming digital audio fax into 
T.38 data.
4. Asterisk box contacts an ATA that supports T.38 (such as the 
Grandstream HT502) and sends it the T.38 data.
5. The ATA converts the T.38 data into an analog fax signal for an 
attached fax machine.

Scenario 2:
1. Somebody sends us a fax, and is an analog signal until it gets into 
the Telco's network where it is translated to digital audio.
2. The call comes in over our T1 to our Asterisk box#1.
3. Asterisk box#1 answers and converts the incoming digital audio fax 
into T.38 data.
4. Asterisk box#1 sends Asterisk box#2 the T.38 data over a relatively 
slow/high latency connection.
5. Asterisk box#2 contacts an ATA that supports T.38 and sends it the 
T.38 data.
6. The ATA converts the T.38 data into an analog fax signal for an 
attached fax machine.

Scenario 3:
1. Our fax machine sends a fax out over an analog line attached to an 
ATA that supports T.38.
2. The ATA converts the fax over analog signal to T.38 data.
3. The ATA sends the T.38 data to Asterisk box.
4. Asterisk box converts the T.38 to fax over analog and sends the 
digital audio out over the T1.
5. At the other end of the Telco's network the digital audio is 
converted to analog and a fax machine receives the fax.

Scenario 4:
1. Somebody sends us a fax, and is an analog signal until it gets into 
the Telco's network where it is translated to digital audio.
2. The call comes in over our T1 to our Asterisk box.
3. Asterisk box answers and converts the incoming digital audio fax into 
T.38 data.
4. Asterisk box contacts a system running sipX with a SIP over UDP 
connection and sends it the T.38 data.
5. sipX contacts an Exchange 2007 Unified Messaging box and sends it the 
T.38 data.
6. Exchange 2007 converts the T.38 data to an image in an email and 
stores it in the user's inbox within Exchange.


Paul Bryson


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Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card

[Matthew Fredrickson]

Joseph Begumisa wrote:
> Has anyone had any compatibility issues with a TE110P card installed on a
> Dell Poweredge 1950?  I noted the following error on the LCD display of the
> Dell Poweredge 1950:
> 
>  
> 
> E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
> 
>  
> 
> The Dell hardware owners manual states that it means the system BIOS has
> reported a PCI parity error on a component that resides in PCI configuration
> space at bus 0, device 4, function 0 and advises that the PCI expansion card
> be removed and reseated.
> 
>  
> 
> Any suggestions on what exactly might be causing this are welcome.

This sounds like something worthy of notifying tech support of.  Can you 
try contacting them so that they can further diagnose this problem?

Thanks.

-- 
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Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card

[Matthew Fredrickson]

Steve Totaro wrote:
> Joseph Begumisa wrote:
>> Has anyone had any compatibility issues with a TE110P card installed 
>> on a Dell Poweredge 1950?  I noted the following error on the LCD 
>> display of the Dell Poweredge 1950:
>>
>>  
>>
>> E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
>>
>>  
>>
>> The Dell hardware owners manual states that it means the system BIOS 
>> has reported a PCI parity error on a component that resides in PCI 
>> configuration space at bus 0, device 4, function 0 and advises that 
>> the PCI expansion card be removed and reseated.
>>
>>  
>>
>> Any suggestions on what exactly might be causing this are welcome.
>>
>>  
>>
>> Thanks.
>>
>>  
>>
>> Joseph
>>
> My guess would be that the Digium card is causing the issue although you 
> would probably be led to believe that the Dell is not compatible with 
> the card and not visa versa.
> 
> It would be interesting to see if a Sangoma board would have that same 
> issue.  I have not had any of these compatibility issues since going 
> Sangoma.
> 
> Is this an older card or one with the "New and Improved Bus" thing? 

The TE110P has the old style PCI interface on it.  The TE120 is the 
newer one based on Voicebus.  In any case, he should contact tech 
support about this so we can resolve it.

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Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] TE210P issues

[Matthew Fredrickson]

Jerry Geis wrote:
> I have a box with a TE210P. Things work for a while then stop when 
> making call files.
> I get NOANSWER as the return code (right away).
> 
> I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1
> 
> When I try to update to newer zaptel the machine locks when loading the 
> zaptel drivers.
> 
> I tried to manually load the wct1xxp module (I think that is the one for 
> the dual T1 card???)
> and the machine locks. I am in a remote location so I cannot see if 
> anything is on the console.
> 
> I tried jumping to 1.4 and the same thing happens.
> I have updated quite a few asterisk boxes remotely and never had this 
> issue before.
> 
> Last thing I tried was "chkconfig zaptel off", reboot, then try loading 
> in new version and the same thing happened.
> It locked up.
> 
> After rebooting I put back the old zaptel and it works again for  awhile.
> 
> What shall I try?

Could you contact tech support about this?  When you purchased your 
card, you also purchased support for issues like this.  And please give 
us a chance to diagnose and fix this problem.  I suspect that they will 
be able to resolve this.

-- 
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Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] Large voicemail

[Tilghman Lesher]

On Thursday 25 October 2007 07:40:06 Pepo wrote:
> I am trying to use Asterisk as the voicemail system of the TELCO where I
> work. I wanna test with 2 mail boxes ( and later with a better
> machine/server I hope try with 7 ).
>
> How do I include in voicemail.conf the file with the mail boxes?, In a big
> system like this,is better use text files or any database?

Well, if it's the same format, you can use #include.  However, with a large
system like that, I would tend to use the database to configure mailboxes,
while using sound files directly on disk.

-- 
Tilghman

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Re: [asterisk-users] TE210P issues

[Matthew Fredrickson]

Steve Totaro wrote:
> Calling Digium.  Post your /var/log/messages and /var/log/asterisk/full 
> (just anything that looks relevant). 
> 
> Try a Sangoma card.

Or better yet, give us an opportunity to fix it.  Sangoma cards have 
problems too and I'm sure they have been going through a "trial of fire" 
trying to eliminate compatibility-type problems with their boards, but 
when you have a problem, you still have to give them (and us) a chance 
to fix it.  It's the nature of the PC world, there are a lot of 
different platforms to interoperate with.

-- 
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Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] Advanced Dial Plan

[Tilghman Lesher]

On Thursday 25 October 2007 10:36:02 Brett Crapser wrote:
> [outbound]
> exten=> _X.,1,GotoIf([${CALLERID(num)} == "3000"]?path0|1)

exten=> _X.,1,GotoIf([${CALLERID(num)} = "3000"]?path0,${EXTEN},1)

> exten=> _X.,2,GotoIf([${CALLERID(num)} == "3001"]?path1|1)

exten=> _X.,2,GotoIf([${CALLERID(num)} = "3001"]?path1,${EXTEN},1)

> exten=> _X.,3,Playback(tt-monkeys)
> exten=> _X.,4,Hangup

You do not need double '=' and you were missing the ${EXTEN}.  Also, '|'
separating arguments is now deprecated.

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Re: [asterisk-users] Compatibility Issues with dell poweredge 195 and TE110P card

[Matthew Fredrickson]

Brian Hutchinson wrote:
> You guys are scaring me! I'm building a 2950 with SAS RAID 5 on the new PERC
> and it will have 2 TE420P's.  I hope it works or my bacon will fry.

You shouldn't see any problems with those boards.  The 2950 is a common 
environment.  If I remember correctly, there used to be a problem (and I 
think it was localized to the TE110P as well, IIRC), but it was fixed a 
while ago.

> 
> On 10/25/07, Joseph Begumisa <[EMAIL PROTECTED]> wrote:
>>> Has anyone had any compatibility issues with a TE110P card installed
>>> on a Dell Poweredge 1950?I noted the following error on the LCD
>>> display of the Dell Poweredge 1950:
>>>
>>>
>>>
>>> E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
>>> Yes, I have had this problem with a dell PE1650, 1850, SC1400, and PE650.
>> I
>> have a TE410P that does it. It may not be wise, but I just ignore the
>> orange
>> blinking LCD display (or light, >depending on the model). I did try
>> reseating the card, and it "works" for a few weeks, and then goes back to
>> the same old thing.
>>
>> Yes, that happened too.  Digium has graciously offered to send me a TE120P
>> with the Digium VoiceBus technology which I will test out on the Dell
>> 1950.
>> Will post my findings thereafter.
>>
>> Joseph.
>>
>>
>>
>>
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> 
> 
> 
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Re: [asterisk-users] PRI span configuration - span remains down

[Matthew Fredrickson]

Rony Ron wrote:
> Hello,
> Quoting Digium Support:
> "The TE110P has been discontinued and replaced in our product lineup with
> the TE120P, which features many overall improvements and does not suffer
> from the HDLC Abort/Bad FCS problems that the TE110P did."

Although this is true ( :-) ) I think that it is likely his problem is 
not related to this.  Can you post a "pri intense debug span x" for the 
span in question?

Matthew Fredrickson

> On 10/25/07, David Kennedy <[EMAIL PROTECTED]> wrote:
>> Hi,
>>
>> I'm trying to connect to Telewest/Virgin Media with a TE110P using
>> asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always
>> appears as
>>
>> PRI span 1/0: Provisioned, Down, Active
>>
>> My zapata.conf is currently
>> ---
>> [channels]
>> echocancel=yes
>> echocancelwhenbridged=no
>> echotraining=yes
>> switchtype=euroisdn
>> contect=from-pri
>> signalling=pri_cpe
>> group=1
>> channel => 1-15
>> channel => 17-31
>> ---
>>
>> zaptel.conf is
>> ---
>> span=1,1,0,ccs,hdb3,crc4
>> dchan=16
>> bchan=1-15,17-31
>> loadzone=uk
>> defaultzone=uk
>> ---
>>
>> I'm in London and the server is in Manchester, so I can't look at the
>> server directly, but when we first started setting it up, apparently a
>> pair of cables were the wrong way round, so the card was in a RED
>> alarm state. We've switched the cables and now the card is OK. We did
>> have a lot of IRQ misses, so we've upgraded the kernel and now the
>> accuracy reported by zttest is about 99.98%. Telewest have checked the
>> line for faults and have reported that it's fine, but I just can't get
>> it working.
>>
>> Does anyone have any ideas/suggestions?
>>
>> Thanks,
>>
>> Dave
>>
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> 
> 
> 
> 
> 
> 
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Re: [asterisk-users] Large voicemail

[Tzafrir Cohen]

On Thu, Oct 25, 2007 at 11:31:37AM -0500, Tilghman Lesher wrote:
> On Thursday 25 October 2007 07:40:06 Pepo wrote:
> > I am trying to use Asterisk as the voicemail system of the TELCO where I
> > work. I wanna test with 2 mail boxes ( and later with a better
> > machine/server I hope try with 7 ).
> >
> > How do I include in voicemail.conf the file with the mail boxes?, In a big
> > system like this,is better use text files or any database?
> 
> Well, if it's the same format, you can use #include.  

Doesn't this break voicemail password changing?

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

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[asterisk-users] What to use instead of LookupCIDName?

[Vincent]

Hello

When using LookupCIDName, Asterisk 1.4 says that it's
deprecated, and we should use "${DB(cidname/${CALLERID(num)})}"
instead, but I don't know how to use it:

;DEPRECATED exten => s,1,LookupCIDName
;ERROR
exten => s,1,${DB(cidname/${CALLERID(num)})}

I guess I should use this as a parameter to a function, but which one?

Thank you.


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[asterisk-users] Unable to dial out over Zap - span 1 got hangup, cause 44

[David Kennedy]

Hi

I posted earlier about having issues connecting to Telewest's ISDN,
only to find out later Telewest had forgotten to turn it on -
hopefully I'm not having a similar silly problem.

My PRI span is now up and operational, but when I try to send a call
out over it, I just get congestion tones. Occasionally, I get about
one second of ring tones, only for it to cut out and play congestion.

Here's a bit of output (I've taken out the phone number)
-- Executing [EMAIL PROTECTED]:6]
Dial("SIP/charlie59-082bc890", "Zap/|3600") in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/
-- Channel 0/4, span 1 got hangup, cause 44
-- Forcing restart of channel 0/4 on span 1 since channel reported in use
-- Hungup 'Zap/4-1'
[Oct 25 17:35:22] NOTICE[20503]: cdr.c:434 ast_cdr_free: CDR on
channel 'Zap/4-1' not posted
  == Everyone is busy/congested at this time (1:0/0/1)

Additionally, once a zap channel has been used like this, it seems to
end up in stuck in this state:
PRI Flags: Resetting

Previously, someone mentioned that the TE110P card installed had a few
issues and I should be using a TE120P instead - could that be the
cause?

Thanks

Dave

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[asterisk-users] Getting SIP Response Code from HANGUPCAUSE

[Douglas Garstang]

I'd like to grab the SIP response code that comes back from an INVITE. The 
HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get 
the SIP response code instead? 

Doug.




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Re: [asterisk-users] What to use instead of LookupCIDName?

[Vincent]

On Thu, 25 Oct 2007 18:46:19 +0200, Vincent
<[EMAIL PROTECTED]> wrote:
>I guess I should use this as a parameter to a function, but which one?

Never mind, I found how to use it:

exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})


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Re: [asterisk-users] PRI span configuration - span remains down

[David Kennedy]

Hi

While I have fixed the problem from this post, I do have another
problem, and you have asked for a debug output here, so I'll go
against my better instinct and reply here :)

-- Making new call for cr 32774
-- Requested transfer capability: 0x00 - SPEECH

> [ 00 01 0e 06 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 
> 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]

> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> N(S): 007   0: 0
> N(R): 003   P: 0
> 44 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
> Protocol Discriminator: Q.931 (8)  len=44
> Call Ref: len= 2 (reference 6/0x6) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
> Speech (0)
>  Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
> (16)
>  Ext: 1  User information layer 1: A-Law (35)
> [18 03 a9 83 86]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
> Dchan: 0
>ChanSel: Reserved
>   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>   Ext: 1  Channel: 6 ]
> [6c 0c 21 83 38 34 35 38 39 39 31 30 30 31]
> Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>   Presentation: Presentation allowed of network 
> provided number (3)  '8458991001' ]
> [70 0c 80 30 32 30 38 36 35 39 32 32 39 31]
> Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown 
> Number Plan (0)  '' ]
> [a1]
> Sending Complete (len= 1)
q931.c:2881 q931_setup: call 32774 on channel 6 enters state 1 (Call Initiated)
-- Called g0/
-- T200 counter expired, What to do...
-- Retransmitting 48 bytes
voip1*CLI>
> [ 00 01 0e 07 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 
> 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
voip1*CLI>
> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> N(S): 007   0: 0
> N(R): 003   P: 1
> 44 bytes of data
-- Rescheduling retransmission (1)
voip1*CLI>
< [ 00 01 01 11 ]
voip1*CLI>
< Supervisory frame:
< SAPI: 00  C/R: 0 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 008 P/F: 1
< 0 bytes of data
-- ACKing all packets from 6 to (but not including) 8
-- ACKing packet 7, new txqueue is -1 (-1 means empty)
-- Since there was nothing left, stopping T200 counter
-- Nothing left, starting T203 counter
-- Got RR response to our frame
-- Restarting T203 counter
voip1*CLI>
< [ 02 01 06 10 08 02 80 06 5a 08 03 82 ac 18 ]
voip1*CLI>
< Informational frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
< N(S): 003   0: 0
< N(R): 008   P: 0
< 10 bytes of data
-- ACKing all packets from 7 to (but not including) 8
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
< Protocol Discriminator: Q.931 (8)  len=10
< Call Ref: len= 2 (reference 6/0x6) (Terminator)
< Message type: RELEASE COMPLETE (90)
< [08 03 82 ac 18]
< Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Public network serving the local user (2)
<  Ext: 1  Cause: Requested channel not available
(44), class = Network Congestion (resource unavailable) (2) ]
<  Cause data 1: 18 (24)
-- Processing IE 8 (cs0, Cause)
q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null)
Sending Receiver Ready (4)
voip1*CLI>
> [ 02 01 01 08 ]
voip1*CLI>
> Supervisory frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> N(R): 004 P/F: 0
> 0 bytes of data
-- Restarting T203 counter
-- Restarting T203 counter
-- Channel 0/6, span 1 got hangup, cause 44
-- Forcing restart of channel 0/6 on span 1 since channel reported in use
voip1*CLI>
> [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 80 ]
voip1*CLI>
> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000EA: 1
> N(S): 008   0: 0
> N(R): 004   P: 0
> 13 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
> Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Originator)
> Message type: RESTART (70)
> [18 03 a9 83 86]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
> Dchan: 0
>ChanSel: Reserved
>   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>   Ext: 1  Channel: 6 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated Channel 
> (0) ]
voip1*CLI>
< [ 00 01 01 12 ]
voip1*CLI>
< Supervisory frame:
< SAPI: 00  C/R: 0 EA: 0
<  TEI: 000EA: 1
< Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
< N(R): 009 P/F: 0
< 0

Re: [asterisk-users] Unable to dial out over Zap - span 1 got hangup, cause 44

[Matthew Fredrickson]

David Kennedy wrote:
> Hi
> 
> I posted earlier about having issues connecting to Telewest's ISDN,
> only to find out later Telewest had forgotten to turn it on -
> hopefully I'm not having a similar silly problem.
> 
> My PRI span is now up and operational, but when I try to send a call
> out over it, I just get congestion tones. Occasionally, I get about
> one second of ring tones, only for it to cut out and play congestion.
> 
> Here's a bit of output (I've taken out the phone number)
> -- Executing [EMAIL PROTECTED]:6]
> Dial("SIP/charlie59-082bc890", "Zap/|3600") in new
> stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called g0/
> -- Channel 0/4, span 1 got hangup, cause 44
> -- Forcing restart of channel 0/4 on span 1 since channel reported in use
> -- Hungup 'Zap/4-1'
> [Oct 25 17:35:22] NOTICE[20503]: cdr.c:434 ast_cdr_free: CDR on
> channel 'Zap/4-1' not posted
>   == Everyone is busy/congested at this time (1:0/0/1)
> 
> Additionally, once a zap channel has been used like this, it seems to
> end up in stuck in this state:
> PRI Flags: Resetting
> 
> Previously, someone mentioned that the TE110P card installed had a few
> issues and I should be using a TE120P instead - could that be the
> cause?

If your span is up ok, and you are actually getting a valid cause code 
back (as you mentioned) your card should be just fine.  It sounds like 
protocol related problems.  Are you sure you are sending the correct 
digit format out on the line?  PRIs can be very picky about it.  Some 
like the area code, some don't, and a number of other things.  Also, can 
you get an inbound call on the PRI?  That's usually the easiest first 
case to get working.

 From looking at the specs, it looks like 44 is cause "requested channel 
unavailable".  Maybe they haven't unbusied the channels yet or something 
like that.

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] Unable to dial out over Zap - span 1 got hangup, cause 44

[Matthew Fredrickson]

David Kennedy wrote:
> Hi
> 
> I posted earlier about having issues connecting to Telewest's ISDN,
> only to find out later Telewest had forgotten to turn it on -
> hopefully I'm not having a similar silly problem.
> 
> My PRI span is now up and operational, but when I try to send a call
> out over it, I just get congestion tones. Occasionally, I get about
> one second of ring tones, only for it to cut out and play congestion.
> 
> Here's a bit of output (I've taken out the phone number)
> -- Executing [EMAIL PROTECTED]:6]
> Dial("SIP/charlie59-082bc890", "Zap/|3600") in new
> stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called g0/
> -- Channel 0/4, span 1 got hangup, cause 44
> -- Forcing restart of channel 0/4 on span 1 since channel reported in use
> -- Hungup 'Zap/4-1'
> [Oct 25 17:35:22] NOTICE[20503]: cdr.c:434 ast_cdr_free: CDR on
> channel 'Zap/4-1' not posted
>   == Everyone is busy/congested at this time (1:0/0/1)
> 
> Additionally, once a zap channel has been used like this, it seems to
> end up in stuck in this state:
> PRI Flags: Resetting
> 
> Previously, someone mentioned that the TE110P card installed had a few
> issues and I should be using a TE120P instead - could that be the
> cause?

Oh yeah, and could you also post a "pri debug span x" of the call as 
well?  That should tell a lot too.

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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[asterisk-users] Features.conf and passing DTMF to the other end

[Martin Smith]

Hi folks,

We have a problem here where users are calling a remote PBX and need to
use # and * to navigate it. We were using the Tt options in Dial() so
that we could later perhaps take advantage of this feature.

Features.conf's sections are fully commented out, so I wasn't expecting
the options on Dial() (the T and t) to have any effects. Anyway... I
plan to turn off the dialing options, but I was wondering how other
people handle this?

Is there a way to say "yeah pass # through if it doesn't match locally"
or better yet, when dialing a dtmf digit, to prefix it with something to
force asterisk to ignore it and pass it along?

Am I stuck with absolutely no features that depend on # or * if I want
users to use those digits on a remote PBX?

Thanks :)

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 


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Re: [asterisk-users] TE210P issues

[Steve Totaro]

Matthew Fredrickson wrote:
> Steve Totaro wrote:
>   
>> Calling Digium.  Post your /var/log/messages and /var/log/asterisk/full 
>> (just anything that looks relevant). 
>>
>> Try a Sangoma card.
>> 
>
> Or better yet, give us an opportunity to fix it.  Sangoma cards have 
> problems too and I'm sure they have been going through a "trial of fire" 
> trying to eliminate compatibility-type problems with their boards, but 
> when you have a problem, you still have to give them (and us) a chance 
> to fix it.  It's the nature of the PC world, there are a lot of 
> different platforms to interoperate with.
>   
Matthew,

I totally agree, that is why my first sentence in the reply was to call 
Digium. 

Everyone needs at least one backup plan (I prefer three or more). 

If the system needs to go live tomorrow, Sangoma can overnight a card 
and it may not exhibit the same issue.  The card could probably be RMAed 
with no problem to Sangoma if Digium gets it fixed in the meantime. 

Sometimes systems need to be turned up yesterday, now, tomorrow, or you 
best case scenario, you have weeks of time to to play around and 
diagnose a problem.  Your plan is going to reflect your timeframe needs.

Thanks,
Steve Totaro

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Re: [asterisk-users] PRI span configuration - span remains down

[Matthew Fredrickson]

David Kennedy wrote:
> Hi
> 
> While I have fixed the problem from this post, I do have another
> problem, and you have asked for a debug output here, so I'll go
> against my better instinct and reply here :)

I just looked through your debug and can't see any obvious problems. 
It's likely you'll need to ask your telco why the other switch is 
complaining about the channel selection.

Matthew Fredrickson

> 
> -- Making new call for cr 32774
> -- Requested transfer capability: 0x00 - SPEECH
> 
>> [ 00 01 0e 06 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 
>> 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
> 
>> Informational frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>> N(S): 007   0: 0
>> N(R): 003   P: 0
>> 44 bytes of data
> -- Restarting T203 counter
> Stopping T_203 timer
> Starting T_200 timer
>> Protocol Discriminator: Q.931 (8)  len=44
>> Call Ref: len= 2 (reference 6/0x6) (Originator)
>> Message type: SETUP (5)
>> [04 03 80 90 a3]
>> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
>> Speech (0)
>>  Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
>> (16)
>>  Ext: 1  User information layer 1: A-Law (35)
>> [18 03 a9 83 86]
>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
>> Dchan: 0
>>ChanSel: Reserved
>>   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>>   Ext: 1  Channel: 6 ]
>> [6c 0c 21 83 38 34 35 38 39 39 31 30 30 31]
>> Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: 
>> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>>   Presentation: Presentation allowed of network 
>> provided number (3)  '8458991001' ]
>> [70 0c 80 30 32 30 38 36 35 39 32 32 39 31]
>> Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown 
>> Number Plan (0)  '' ]
>> [a1]
>> Sending Complete (len= 1)
> q931.c:2881 q931_setup: call 32774 on channel 6 enters state 1 (Call 
> Initiated)
> -- Called g0/
> -- T200 counter expired, What to do...
> -- Retransmitting 48 bytes
> voip1*CLI>
>> [ 00 01 0e 07 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 
>> 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
> voip1*CLI>
>> Informational frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>> N(S): 007   0: 0
>> N(R): 003   P: 1
>> 44 bytes of data
> -- Rescheduling retransmission (1)
> voip1*CLI>
> < [ 00 01 01 11 ]
> voip1*CLI>
> < Supervisory frame:
> < SAPI: 00  C/R: 0 EA: 0
> <  TEI: 000EA: 1
> < Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> < N(R): 008 P/F: 1
> < 0 bytes of data
> -- ACKing all packets from 6 to (but not including) 8
> -- ACKing packet 7, new txqueue is -1 (-1 means empty)
> -- Since there was nothing left, stopping T200 counter
> -- Nothing left, starting T203 counter
> -- Got RR response to our frame
> -- Restarting T203 counter
> voip1*CLI>
> < [ 02 01 06 10 08 02 80 06 5a 08 03 82 ac 18 ]
> voip1*CLI>
> < Informational frame:
> < SAPI: 00  C/R: 1 EA: 0
> <  TEI: 000EA: 1
> < N(S): 003   0: 0
> < N(R): 008   P: 0
> < 10 bytes of data
> -- ACKing all packets from 7 to (but not including) 8
> -- Since there was nothing left, stopping T200 counter
> -- Stopping T203 counter since we got an ACK
> -- Nothing left, starting T203 counter
> < Protocol Discriminator: Q.931 (8)  len=10
> < Call Ref: len= 2 (reference 6/0x6) (Terminator)
> < Message type: RELEASE COMPLETE (90)
> < [08 03 82 ac 18]
> < Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
> Location: Public network serving the local user (2)
> <  Ext: 1  Cause: Requested channel not available
> (44), class = Network Congestion (resource unavailable) (2) ]
> <  Cause data 1: 18 (24)
> -- Processing IE 8 (cs0, Cause)
> q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null)
> Sending Receiver Ready (4)
> voip1*CLI>
>> [ 02 01 01 08 ]
> voip1*CLI>
>> Supervisory frame:
>> SAPI: 00  C/R: 1 EA: 0
>>  TEI: 000EA: 1
>> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
>> N(R): 004 P/F: 0
>> 0 bytes of data
> -- Restarting T203 counter
> -- Restarting T203 counter
> -- Channel 0/6, span 1 got hangup, cause 44
> -- Forcing restart of channel 0/6 on span 1 since channel reported in use
> voip1*CLI>
>> [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 80 ]
> voip1*CLI>
>> Informational frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>> N(S): 008   0: 0
>> N(R): 004   P: 0
>> 13 bytes of data
> -- Restarting T203 counter
> Stopping T_203 timer
> Starting T_200 timer
>> Protocol Discriminator: Q.931 (8)  len=13
>> Call Ref: len= 2 (reference 0/0x0) (Originator)
>> Message type: RESTART (70)
>> [18 03 a9 83 86]
>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
>> Dchan: 0
>>  

Re: [asterisk-users] Compatibility Issues with dell poweredge 195 and TE110P card

[Brian Hutchinson]

The thing I'm trying to get an answer on now is getting Dell or rPath to
tell me what I have to do to get the 256M battery backed up RAM going and if
I have to do anything special due to the SAS drives on the new PERC 5/i
controller!

I'm running AsteriskNOW to build my project while I wait for the TE420P
cards to come in and ABE to be released.

Regards,

Brian

On 10/25/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote:
>
> Brian Hutchinson wrote:
> > You guys are scaring me! I'm building a 2950 with SAS RAID 5 on the new
> PERC
> > and it will have 2 TE420P's.  I hope it works or my bacon will fry.
>
> You shouldn't see any problems with those boards.  The 2950 is a common
> environment.  If I remember correctly, there used to be a problem (and I
> think it was localized to the TE110P as well, IIRC), but it was fixed a
> while ago.
>
> >
> > On 10/25/07, Joseph Begumisa <[EMAIL PROTECTED]> wrote:
> >>> Has anyone had any compatibility issues with a TE110P card installed
> >>> on a Dell Poweredge 1950?I noted the following error on the LCD
> >>> display of the Dell Poweredge 1950:
> >>>
> >>>
> >>>
> >>> E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
> >>> Yes, I have had this problem with a dell PE1650, 1850, SC1400, and
> PE650.
> >> I
> >> have a TE410P that does it. It may not be wise, but I just ignore the
> >> orange
> >> blinking LCD display (or light, >depending on the model). I did try
> >> reseating the card, and it "works" for a few weeks, and then goes back
> to
> >> the same old thing.
> >>
> >> Yes, that happened too.  Digium has graciously offered to send me a
> TE120P
> >> with the Digium VoiceBus technology which I will test out on the Dell
> >> 1950.
> >> Will post my findings thereafter.
> >>
> >> Joseph.
> >>
> >>
> >>
> >>
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> >>
> >
> >
> > 
> >
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>
> --
> Matthew Fredrickson
> Software/Firmware Engineer
> Digium, Inc.
>
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Re: [asterisk-users] PRI span configuration - span remains down

[David Kennedy]

Is there some part of the debug output I need to tell the telco about?
When I was on to them earlier today, the engineer only seemed to know
how to turn bits of their network on and off, not much about settings
I need my end etc.

Dave

On 10/25/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote:
> David Kennedy wrote:
> > Hi
> >
> > While I have fixed the problem from this post, I do have another
> > problem, and you have asked for a debug output here, so I'll go
> > against my better instinct and reply here :)
>
> I just looked through your debug and can't see any obvious problems.
> It's likely you'll need to ask your telco why the other switch is
> complaining about the channel selection.
>
> Matthew Fredrickson
>
> >
> > -- Making new call for cr 32774
> > -- Requested transfer capability: 0x00 - SPEECH
> >
> >> [ 00 01 0e 06 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 
> >> 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
> >
> >> Informational frame:
> >> SAPI: 00  C/R: 0 EA: 0
> >>  TEI: 000EA: 1
> >> N(S): 007   0: 0
> >> N(R): 003   P: 0
> >> 44 bytes of data
> > -- Restarting T203 counter
> > Stopping T_203 timer
> > Starting T_200 timer
> >> Protocol Discriminator: Q.931 (8)  len=44
> >> Call Ref: len= 2 (reference 6/0x6) (Originator)
> >> Message type: SETUP (5)
> >> [04 03 80 90 a3]
> >> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
> >> capability: Speech (0)
> >>  Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
> >> (16)
> >>  Ext: 1  User information layer 1: A-Law (35)
> >> [18 03 a9 83 86]
> >> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
> >> Dchan: 0
> >>ChanSel: Reserved
> >>   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> >>   Ext: 1  Channel: 6 ]
> >> [6c 0c 21 83 38 34 35 38 39 39 31 30 30 31]
> >> Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: 
> >> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> >>   Presentation: Presentation allowed of network 
> >> provided number (3)  '8458991001' ]
> >> [70 0c 80 30 32 30 38 36 35 39 32 32 39 31]
> >> Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: 
> >> Unknown Number Plan (0)  '' ]
> >> [a1]
> >> Sending Complete (len= 1)
> > q931.c:2881 q931_setup: call 32774 on channel 6 enters state 1 (Call 
> > Initiated)
> > -- Called g0/
> > -- T200 counter expired, What to do...
> > -- Retransmitting 48 bytes
> > voip1*CLI>
> >> [ 00 01 0e 07 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 
> >> 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
> > voip1*CLI>
> >> Informational frame:
> >> SAPI: 00  C/R: 0 EA: 0
> >>  TEI: 000EA: 1
> >> N(S): 007   0: 0
> >> N(R): 003   P: 1
> >> 44 bytes of data
> > -- Rescheduling retransmission (1)
> > voip1*CLI>
> > < [ 00 01 01 11 ]
> > voip1*CLI>
> > < Supervisory frame:
> > < SAPI: 00  C/R: 0 EA: 0
> > <  TEI: 000EA: 1
> > < Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> > < N(R): 008 P/F: 1
> > < 0 bytes of data
> > -- ACKing all packets from 6 to (but not including) 8
> > -- ACKing packet 7, new txqueue is -1 (-1 means empty)
> > -- Since there was nothing left, stopping T200 counter
> > -- Nothing left, starting T203 counter
> > -- Got RR response to our frame
> > -- Restarting T203 counter
> > voip1*CLI>
> > < [ 02 01 06 10 08 02 80 06 5a 08 03 82 ac 18 ]
> > voip1*CLI>
> > < Informational frame:
> > < SAPI: 00  C/R: 1 EA: 0
> > <  TEI: 000EA: 1
> > < N(S): 003   0: 0
> > < N(R): 008   P: 0
> > < 10 bytes of data
> > -- ACKing all packets from 7 to (but not including) 8
> > -- Since there was nothing left, stopping T200 counter
> > -- Stopping T203 counter since we got an ACK
> > -- Nothing left, starting T203 counter
> > < Protocol Discriminator: Q.931 (8)  len=10
> > < Call Ref: len= 2 (reference 6/0x6) (Terminator)
> > < Message type: RELEASE COMPLETE (90)
> > < [08 03 82 ac 18]
> > < Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
> > Location: Public network serving the local user (2)
> > <  Ext: 1  Cause: Requested channel not available
> > (44), class = Network Congestion (resource unavailable) (2) ]
> > <  Cause data 1: 18 (24)
> > -- Processing IE 8 (cs0, Cause)
> > q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null)
> > Sending Receiver Ready (4)
> > voip1*CLI>
> >> [ 02 01 01 08 ]
> > voip1*CLI>
> >> Supervisory frame:
> >> SAPI: 00  C/R: 1 EA: 0
> >>  TEI: 000EA: 1
> >> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> >> N(R): 004 P/F: 0
> >> 0 bytes of data
> > -- Restarting T203 counter
> > -- Restarting T203 counter
> > -- Channel 0/6, span 1 got hangup, cause 44
> > -- Forcing restart of channel 0/6 on span 1 since channel reported in 
> > use
> > voip1*C

[asterisk-users] Coming-off-hold delay/silence on Sipura 841 and Asterisk

[Chris Hanson]

  Hi all. Newbie to the list, been using VOIP with Sipura & Grandstream
hardphones for a few years, via a VOIP service provider (who I won't name
here). I haven't stepped up to running my own Asterisk box yet, because of
poor reliability of our Internet connection during non-business hours, but
I'm considering it in the future.

  I can't provide a ton of detail about the Asterisk version and
configuration at my service provider's end, as they admin it and I don't
know how to query what exactly it runs by normal VOIP channels. On my end I
use Sipura 841s (decent) and one Grandstream GXP2000 (total piece of junk).

  I've weathered some hiccups recently from our SP upgrading their setup and
changing things. Normal operations are pretty stable now, though I
understand from them that we are on a slightly older version of Asterisk
than is presently available due to a problem that came up with the newest.

  Recently, a new issue has surfaced that I haven't found a solution to -- a
customer calls in, we answer and talk, and put them on hold. When taking
them off hold, they are unable to hear us at all for the first couple of
seconds. After that the call proceeds as normal.

  While troubleshooting with the SP, they suggested turning the Handset Gain
level from 0 (default) down to -6, and when we did, the problem seemed to go
away. They claim this has solved the problem for others as well.

  This strikes me as a kludge, and not really finding or solving the real
problem, and I am loathe to adopt such an arbitrary hack. I was hoping to
find out if anyone else had run into a situation like this, and if so, what
the cause and solution were. I don't think the cause lies at my end, as we
have not changed our configuration at all in some time, and the problem
seemed to arise spontaneously recently.

  Now granted, it's probably my SP's responsibility to be trying to solve
this, but they seem satisfied that the Handset Gain is the solution -- I'm
the only one who is objecting to that. So, please bear with my weak
knowledge of Asterisk and speak clearly in short sentences. ;)

  Thanks in advance for any advice.
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Re: [asterisk-users] PRI span configuration - span remains down

[Rony Ron]

Hi, in meantime if you have another type of digium pri
card you can plug it into your box to confirm that it's not related to
that card!
Better eliminate any doubt about that card... it made me suffer !

BR,


On 10/25/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote:
>
> David Kennedy wrote:
> > Hi
> >
> > While I have fixed the problem from this post, I do have another
> > problem, and you have asked for a debug output here, so I'll go
> > against my better instinct and reply here :)
>
> I just looked through your debug and can't see any obvious problems.
> It's likely you'll need to ask your telco why the other switch is
> complaining about the channel selection.
>
> Matthew Fredrickson
>
> >
> > -- Making new call for cr 32774
> > -- Requested transfer capability: 0x00 - SPEECH
> >
> >> [ 00 01 0e 06 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83
> 38 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
> >
> >> Informational frame:
> >> SAPI: 00  C/R: 0 EA: 0
> >>  TEI: 000EA: 1
> >> N(S): 007   0: 0
> >> N(R): 003   P: 0
> >> 44 bytes of data
> > -- Restarting T203 counter
> > Stopping T_203 timer
> > Starting T_200 timer
> >> Protocol Discriminator: Q.931 (8)  len=44
> >> Call Ref: len= 2 (reference 6/0x6) (Originator)
> >> Message type: SETUP (5)
> >> [04 03 80 90 a3]
> >> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
> capability: Speech (0)
> >>  Ext: 1  Trans mode/rate: 64kbps,
> circuit-mode (16)
> >>  Ext: 1  User information layer 1: A-Law
> (35)
> >> [18 03 a9 83 86]
> >> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare:
> 0  Exclusive  Dchan: 0
> >>ChanSel: Reserved
> >>   Ext: 1  Coding: 0  Number Specified  Channel
> Type: 3
> >>   Ext: 1  Channel: 6 ]
> >> [6c 0c 21 83 38 34 35 38 39 39 31 30 30 31]
> >> Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> >>   Presentation: Presentation allowed of network
> provided number (3)  '8458991001' ]
> >> [70 0c 80 30 32 30 38 36 35 39 32 32 39 31]
> >> Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
> Unknown Number Plan (0)  '' ]
> >> [a1]
> >> Sending Complete (len= 1)
> > q931.c:2881 q931_setup: call 32774 on channel 6 enters state 1 (Call
> Initiated)
> > -- Called g0/
> > -- T200 counter expired, What to do...
> > -- Retransmitting 48 bytes
> > voip1*CLI>
> >> [ 00 01 0e 07 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83
> 38 34 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
> > voip1*CLI>
> >> Informational frame:
> >> SAPI: 00  C/R: 0 EA: 0
> >>  TEI: 000EA: 1
> >> N(S): 007   0: 0
> >> N(R): 003   P: 1
> >> 44 bytes of data
> > -- Rescheduling retransmission (1)
> > voip1*CLI>
> > < [ 00 01 01 11 ]
> > voip1*CLI>
> > < Supervisory frame:
> > < SAPI: 00  C/R: 0 EA: 0
> > <  TEI: 000EA: 1
> > < Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> > < N(R): 008 P/F: 1
> > < 0 bytes of data
> > -- ACKing all packets from 6 to (but not including) 8
> > -- ACKing packet 7, new txqueue is -1 (-1 means empty)
> > -- Since there was nothing left, stopping T200 counter
> > -- Nothing left, starting T203 counter
> > -- Got RR response to our frame
> > -- Restarting T203 counter
> > voip1*CLI>
> > < [ 02 01 06 10 08 02 80 06 5a 08 03 82 ac 18 ]
> > voip1*CLI>
> > < Informational frame:
> > < SAPI: 00  C/R: 1 EA: 0
> > <  TEI: 000EA: 1
> > < N(S): 003   0: 0
> > < N(R): 008   P: 0
> > < 10 bytes of data
> > -- ACKing all packets from 7 to (but not including) 8
> > -- Since there was nothing left, stopping T200 counter
> > -- Stopping T203 counter since we got an ACK
> > -- Nothing left, starting T203 counter
> > < Protocol Discriminator: Q.931 (8)  len=10
> > < Call Ref: len= 2 (reference 6/0x6) (Terminator)
> > < Message type: RELEASE COMPLETE (90)
> > < [08 03 82 ac 18]
> > < Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
> > Location: Public network serving the local user (2)
> > <  Ext: 1  Cause: Requested channel not available
> > (44), class = Network Congestion (resource unavailable) (2) ]
> > <  Cause data 1: 18 (24)
> > -- Processing IE 8 (cs0, Cause)
> > q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null)
> > Sending Receiver Ready (4)
> > voip1*CLI>
> >> [ 02 01 01 08 ]
> > voip1*CLI>
> >> Supervisory frame:
> >> SAPI: 00  C/R: 1 EA: 0
> >>  TEI: 000EA: 1
> >> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> >> N(R): 004 P/F: 0
> >> 0 bytes of data
> > -- Restarting T203 counter
> > -- Restarting T203 counter
> > -- Channel 0/6, span 1 got hangup, cause 44
> > -- Forcing restart of channel 0/6 on span 1 since channel reported
> in use
> > voip1*CLI>
> >> [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 

[asterisk-users] GUI for Asterisk 1.2 Source

[OCOSA ListAcct]

Hi,

Is there a GUI for Asterisk 1.2 compiled from source or would I need to 
upgrade to the 1.4 version to get the GUI that can be installed on 
servers complied from source? Any help is appreciated.

Otis




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Re: [asterisk-users] GUI for Asterisk 1.2 Source

[Tzafrir Cohen]

On Thu, Oct 25, 2007 at 01:46:53PM -0500, OCOSA ListAcct wrote:
> Hi,
> 
> Is there a GUI for Asterisk 1.2 compiled from source or would I need to 
> upgrade to the 1.4 version to get the GUI that can be installed on 
> servers complied from source? Any help is appreciated.

asterisk-gui[tm] requires asterisk 1.4 . Ther are a number of other
graphical user interfaces for Asterisk which work well with Asterisk 1.2.

/me recommends gvim and runs.

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

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[asterisk-users] Grandstream GXV-3000

[hin lee]

I am trying to set up a Grandstream GXV-3000 Video
phone to Asterisk ver 1.2.21.1.  The problem I'm
having is that it can call other SIP phones, but not
vice versa.  Can someone tell me where is the problem?
TIA!

Here's part of my configurations:

--
sip.conf
--
; 113 is the Grandstream phone
[113]
type=friend
username=113
secret=secret
context=default
dtmfmode = rfc2833
host = dynamic
qualify = yes
allow = h263
video=yes
videosupport=yes

; 112 is the X-Lite phone
[112]
type=friend
host=dynamic
user=112
username=112
secret=secret
allow=all
nat=no
 
-
extensions.conf
-
exten => 112,1,Dial(SIP/112)
exten => 112,2,Playback(vm-nobodyavail)
exten => 112,102,Playback(tt-allbusy)
exten => 112,103,Voicemail([EMAIL PROTECTED])

exten => 113,1,Dail(SIP/113)
exten => 113,2,Playback(vm-nobodyavail)
exten => 113,102,Playback(tt-allbusy)
exten => 113,103,Voicemail([EMAIL PROTECTED])

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Re: [asterisk-users] GUI for Asterisk 1.2 Source

[OCOSA ListAcct]

Thanks...

Tzafrir Cohen wrote:
> On Thu, Oct 25, 2007 at 01:46:53PM -0500, OCOSA ListAcct wrote:
>   
>> Hi,
>>
>> Is there a GUI for Asterisk 1.2 compiled from source or would I need to 
>> upgrade to the 1.4 version to get the GUI that can be installed on 
>> servers complied from source? Any help is appreciated.
>> 
>
> asterisk-gui[tm] requires asterisk 1.4 . Ther are a number of other
> graphical user interfaces for Asterisk which work well with Asterisk 1.2.
>
> /me recommends gvim and runs.
>
>   


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Re: [asterisk-users] Grandstream GXV-3000

[Rafael Canchola]

Hi.

Check the codec allowing, disallow=all and allow=ulaw etc.


At 02:25 p.m. 25/10/2007, hin lee wrote:
>I am trying to set up a Grandstream GXV-3000 Video
>phone to Asterisk ver 1.2.21.1.  The problem I'm
>having is that it can call other SIP phones, but not
>vice versa.  Can someone tell me where is the problem?
>TIA!
>
>Here's part of my configurations:
>
>--
>sip.conf
>--
>; 113 is the Grandstream phone
>[113]
>type=friend
>username=113
>secret=secret
>context=default
>dtmfmode = rfc2833
>host = dynamic
>qualify = yes
>allow = h263
>video=yes
>videosupport=yes
>
>; 112 is the X-Lite phone
>[112]
>type=friend
>host=dynamic
>user=112
>username=112
>secret=secret
>allow=all
>nat=no
>
>-
>extensions.conf
>-
>exten => 112,1,Dial(SIP/112)
>exten => 112,2,Playback(vm-nobodyavail)
>exten => 112,102,Playback(tt-allbusy)
>exten => 112,103,Voicemail([EMAIL PROTECTED])
>
>exten => 113,1,Dail(SIP/113)
>exten => 113,2,Playback(vm-nobodyavail)
>exten => 113,102,Playback(tt-allbusy)
>exten => 113,103,Voicemail([EMAIL PROTECTED])
>
>__
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Re: [asterisk-users] Cisco Phones

[Mojo with Horan & Company, LLC]

Can you comment on the use of these phones with asterisk with the Skinny 
images?  I think you're talking about Cisco phones converted to using 
the SIP image.

Moj

Alex Balashov wrote:
> Roy,
>
> While there is a difference in the feature set provided by the
> SIP and Skinny images for the Cisco phones, the loss is not
> appreciable in my view.  There are some differences in interface
> aesthetics as well.
>   


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Re: [asterisk-users] Grandstream GXV-3000

[Mojo with Horan & Company, LLC]

Is your snippet from extensions.conf in the [default] context? 

Are spaces OK around the '=' in sip.conf?  They might be, just an idea.

Now I notice that for friend 112, you say codecs 'allow=all'  and for 
friend 113 you say 'allow=h263' -- maybe you need to explicitly allow 
something like ulaw on friend [113] (thinking maybe they're just not 
agreeing on codecs...?)  Try adding 'disallow=all' to [113]'s 
definition, just before the 'allow=h263' and the 'allow=ulaw' i'm 
suggesting.  If, theoretically, friend 113 will ONLY use h263, does 
X-Lite support this codec?

Have you tried kicking the verbose level at the console up a little bit?


Moj

hin lee wrote:
> I am trying to set up a Grandstream GXV-3000 Video
> phone to Asterisk ver 1.2.21.1.  The problem I'm
> having is that it can call other SIP phones, but not
> vice versa.  Can someone tell me where is the problem?
> TIA!
>
> Here's part of my configurations:
>
> --
> sip.conf
> --
> ; 113 is the Grandstream phone
> [113]
> type=friend
> username=113
> secret=secret
> context=default
> dtmfmode = rfc2833
> host = dynamic
> qualify = yes
> allow = h263
> video=yes
> videosupport=yes
>
> ; 112 is the X-Lite phone
> [112]
> type=friend
> host=dynamic
> user=112
> username=112
> secret=secret
> allow=all
> nat=no
>  
> -
> extensions.conf
> -
> exten => 112,1,Dial(SIP/112)
> exten => 112,2,Playback(vm-nobodyavail)
> exten => 112,102,Playback(tt-allbusy)
> exten => 112,103,Voicemail([EMAIL PROTECTED])
>
> exten => 113,1,Dail(SIP/113)
> exten => 113,2,Playback(vm-nobodyavail)
> exten => 113,102,Playback(tt-allbusy)
> exten => 113,103,Voicemail([EMAIL PROTECTED])
>
> __
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Re: [asterisk-users] Advanced Dial Plan

[Frederico Madeira]

Thanks for advices.

The last one from Tilghman fit better for my needs.

Thanks a lot.

-- 
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br




2007/10/25, Tilghman Lesher <[EMAIL PROTECTED]>:
> On Thursday 25 October 2007 10:36:02 Brett Crapser wrote:
> > [outbound]
> > exten=> _X.,1,GotoIf([${CALLERID(num)} == "3000"]?path0|1)
>
> exten=> _X.,1,GotoIf([${CALLERID(num)} = "3000"]?path0,${EXTEN},1)
>
> > exten=> _X.,2,GotoIf([${CALLERID(num)} == "3001"]?path1|1)
>
> exten=> _X.,2,GotoIf([${CALLERID(num)} = "3001"]?path1,${EXTEN},1)
>
> > exten=> _X.,3,Playback(tt-monkeys)
> > exten=> _X.,4,Hangup
>
> You do not need double '=' and you were missing the ${EXTEN}.  Also, '|'
> separating arguments is now deprecated.
>
> --
> Tilghman
>
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Re: [asterisk-users] Advanced Dial Plan

[Steve Murphy]

On Thu, 2007-10-25 at 11:35 -0500, Tilghman Lesher wrote:
> On Thursday 25 October 2007 10:36:02 Brett Crapser wrote:
> > [outbound]
> > exten=> _X.,1,GotoIf([${CALLERID(num)} == "3000"]?path0|1)
> 
> exten=> _X.,1,GotoIf([${CALLERID(num)} = "3000"]?path0,${EXTEN},1)
> 
> > exten=> _X.,2,GotoIf([${CALLERID(num)} == "3001"]?path1|1)
> 
> exten=> _X.,2,GotoIf([${CALLERID(num)} = "3001"]?path1,${EXTEN},1)
> 
> > exten=> _X.,3,Playback(tt-monkeys)
> > exten=> _X.,4,Hangup
> 
> You do not need double '=' and you were missing the ${EXTEN}.  Also, '|'
> separating arguments is now deprecated.
> 

Although late model expression parser code allows ==, and treats it
like =,
and, & and && are interchangeable, and so also | and ||.

murf



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[asterisk-users] Kirk IP600/3 Wireless Server SIP config

[Remco Barendse]

Hi list!

Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk?

Any experiences / caveats?

If anyone would be willing to share the dump of their IP600 config file, 
i would really appreciate it.

Is there anything special i should put in my asterisk config?

Thanks !!!
Remco

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Re: [asterisk-users] Advanced Dial Plan

[Eric "ManxPower" Wieling]

Steve Murphy wrote:

> Although late model expression parser code allows ==, and treats it
> like =,
> and, & and && are interchangeable, and so also | and ||.
> 
> murf

"late model" is SVN TRUNK, 1.4, SVN BRANCH 1.4, or 1.2?

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Re: [asterisk-users] CISCO 7921G with asterisk

[Dan Austin]

Jordi wrote:
> Any one have experience with this CISCO Wireless 
> IP phone running with Asterisk??

> It doesn't support SIP protocol I believe, so I need
> to know if the skinny channel can work with the 7921.

The 7921 works fine with SVN trunk, and I think the
trivial changes required to support it also made it into
1.4 around release 1.4.7

Dan

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Re: [asterisk-users] Grandstream GXV-3000

[hin lee]

Thanks for the replies.  Found my error, it's in the
extensions.conf. The word "dial" was misspelled.  LOL!
 Felt like a fool now. 


--- "Mojo with Horan & Company, LLC"
<[EMAIL PROTECTED]> wrote:

> Is your snippet from extensions.conf in the
> [default] context? 
> 
> Are spaces OK around the '=' in sip.conf?  They
> might be, just an idea.
> 
> Now I notice that for friend 112, you say codecs
> 'allow=all'  and for 
> friend 113 you say 'allow=h263' -- maybe you need to
> explicitly allow 
> something like ulaw on friend [113] (thinking maybe
> they're just not 
> agreeing on codecs...?)  Try adding 'disallow=all'
> to [113]'s 
> definition, just before the 'allow=h263' and the
> 'allow=ulaw' i'm 
> suggesting.  If, theoretically, friend 113 will ONLY
> use h263, does 
> X-Lite support this codec?
> 
> Have you tried kicking the verbose level at the
> console up a little bit?
> 
> 
> Moj
> 
> hin lee wrote:
> > I am trying to set up a Grandstream GXV-3000 Video
> > phone to Asterisk ver 1.2.21.1.  The problem I'm
> > having is that it can call other SIP phones, but
> not
> > vice versa.  Can someone tell me where is the
> problem?
> > TIA!
> >
> > Here's part of my configurations:
> >
> > --
> > sip.conf
> > --
> > ; 113 is the Grandstream phone
> > [113]
> > type=friend
> > username=113
> > secret=secret
> > context=default
> > dtmfmode = rfc2833
> > host = dynamic
> > qualify = yes
> > allow = h263
> > video=yes
> > videosupport=yes
> >
> > ; 112 is the X-Lite phone
> > [112]
> > type=friend
> > host=dynamic
> > user=112
> > username=112
> > secret=secret
> > allow=all
> > nat=no
> >  
> > -
> > extensions.conf
> > -
> > exten => 112,1,Dial(SIP/112)
> > exten => 112,2,Playback(vm-nobodyavail)
> > exten => 112,102,Playback(tt-allbusy)
> > exten => 112,103,Voicemail([EMAIL PROTECTED])
> >
> > exten => 113,1,Dail(SIP/113)
> > exten => 113,2,Playback(vm-nobodyavail)
> > exten => 113,102,Playback(tt-allbusy)
> > exten => 113,103,Voicemail([EMAIL PROTECTED])
> >
> > __
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> > Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> > http://mail.yahoo.com 
> >
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Re: [asterisk-users] Grandstream GXV-3000

[Mojo with Horan & Company, LLC]

lol that's a good stumper :)  I missed that too!
hin lee wrote:
> Thanks for the replies.  Found my error, it's in the
> extensions.conf. The word "dial" was misspelled.  LOL!
>  Felt like a fool now. 
>
>   
>>> exten => 113,1,Dail(SIP/113)
>>>   


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Re: [asterisk-users] Cisco Phones

[Anciso, Roy]

That is correct. Our Cisco rep is sending us a 7911G and 7941G so we can
test with asterisk.  We plan on converting them over to SIP for testing.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan & Company, LLC
Sent: Thursday, October 25, 2007 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco Phones

Can you comment on the use of these phones with asterisk with the Skinny

images?  I think you're talking about Cisco phones converted to using 
the SIP image.

Moj

Alex Balashov wrote:
> Roy,
>
> While there is a difference in the feature set provided by the
> SIP and Skinny images for the Cisco phones, the loss is not
> appreciable in my view.  There are some differences in interface
> aesthetics as well.
>   


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Re: [asterisk-users] Advanced Dial Plan

[Steve Murphy]

On Thu, 2007-10-25 at 10:36 -0500, Brett Crapser wrote:
> On Thu, 25 Oct 2007, Frederico Madeira wrote:
> > Philipp
> >
> > This didn't wotk.
> >
> > Let's suppose that my sip extension 3000 want to call to (302).123.3211
> > I need a rule in extensions.conf to match with this number, right ?
> > So, I can't use rules that you advice.
> >
> > My problem is only for outbound calls.
> >
> > 2007/10/25, Philipp Kempgen <[EMAIL PROTECTED]>:
> >> Frederico Madeira wrote:
> >>
> >>> I Have in my sip.conf two extension 3000 and 3001.
> >>>
> >>> I have this rule in my extensions.conf
> >>>
> >>> exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
> >>> exten=> _X.,2,Hangup
> >>>
> >>> exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED](num)},60,Tt)
> >>> exten=> _X.,2,Hangup
> >>>
> >>>
> >>> And every calls made by my both extension was using the first rule, so
> >>> calls from  extension 3000 match with peer and work, but calls from
> >>> 3001 didn't match with peer and I got error.
> >>
> >>
> >> exten=> 3000,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
> >> exten=> 3000,n,Hangup()
> >>
> >> exten=> 3001,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
> >> exten=> 3001,n,Hangup()
> >>
> >> That dialplan is about as easy as it can get. :)
> >>
> >> Regards,
> >>   Philipp Kempgen
> 
> Okie dokie
> 
> [outbound]
> exten=> _X.,1,GotoIf([${CALLERID(num)} == "3000"]?path0|1)
> exten=> _X.,2,GotoIf([${CALLERID(num)} == "3001"]?path1|1)

oops! you need to say:

exten=> _X.,1,GotoIf(["${CALLERID(num)}" == "3000"]?path0|1)
exten=> _X.,2,GotoIf(["${CALLERID(num)}" == "3001"]?path1|1)

.. or the double quotes won't match, and you'll hear monkeys for sure!


> exten=> _X.,3,Playback(tt-monkeys)
> exten=> _X.,4,Hangup
> 
> [path0]
> exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
> exten=> _X.,2,Hangup
> 
> [path1]
> exten=> _X.,1,Dial(SIP/[EMAIL PROTECTED],60,Tt)
> exten=> _X.,2,Hangup
> 
> If you hear the monkeys...
> 
> Brett
> 
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[asterisk-users] In my messages log..

[Dominic Son]

Hi. I'm still a bit of a newb to linux, I see this in my messages log:
Asterisk init: Id "ax" respawning too fast: disabled for 5 minutes

What does this mean?
and how severe is it?



-- 
Anything else, let me know.

- Dominic


"It is not the force of a stroke that makes fine art"
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Re: [asterisk-users] Advanced Dial Plan

[Steve Murphy]

On Thu, 2007-10-25 at 15:26 -0500, Eric "ManxPower" Wieling wrote:
> Steve Murphy wrote:
> 
> > Although late model expression parser code allows ==, and treats it
> > like =,
> > and, & and && are interchangeable, and so also | and ||.
> > 
> > murf
> 
> "late model" is SVN TRUNK, 1.4, SVN BRANCH 1.4, or 1.2?
> 

Late model == trunk. Change was too late for 1.4




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Re: [asterisk-users] Kirk IP600/3 Wireless Server SIP config

[Luis Antonio Prata Barbosa]

Some days ago, I was looking for some mobility solutions...

My conclusion is Wi-Fi phones are growing up fast and I think it's only a
time question they became a standart for mobility in pbx, as well as pure IP
telephony. Even manufactures of DECT systems are preparing their products
line to Wi-fi.

Of course, DECT is a mature technology but If you could spend some time in
tests and adjusts I suggest you to think in wi-fi as an option.

Any opinions ???

Thanks.
Luis A P Barbosa

2007/10/25, Remco Barendse <[EMAIL PROTECTED]>:
>
> Hi list!
>
> Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk?
>
> Any experiences / caveats?
>
> If anyone would be willing to share the dump of their IP600 config file,
> i would really appreciate it.
>
> Is there anything special i should put in my asterisk config?
>
> Thanks !!!
> Remco
>
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[asterisk-users] Realtime on Asterisk 1.2.24

[Steve Totaro]

Does realtime work reliably on Asterisk 1.2.24?

Are there any definitive guides, I can only find bits and pieces here 
and there.  Any accurate howtos would be of great help.

I am missing func_realtime.so.  Where does this file come from?  
Asterisk or asterisk-addons?  I saw in one of the howtos that it is 
needed.  Is it needed for 1.2.X or 1.4.X.  Also, what about the switch 
lines in the .conf files.  Some howtos say you need them, others say to 
delete the whole file, that if for example, extensions.conf exisist, 
then Realtime wont load extensions.

TIA,
Steve Totaro

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[asterisk-users] Regarding ast-ax-snmpd

[dhannya.chandran]

 
Hi,
 
   I have downloaded the ast-ax-snmpd package  in Ubuntu Edgy Eft (6.10)
kernel 2.6.17  ;   Asterisk 1.2.12.1 server is already installed in
my system  and its working  fine .   As per README file I have  made
patching and after that when I  issued the "make" command in Asterisk
its  showing error like  " linux/compiler.h"   file not found ( error is
from Asterisk1.1.12.1/channel/chan_phone.c ) . Even if I have copied the
compiler.h header file  its  showing   other errors   all related to
header files  only .  May I know what is the problem .  ?
 
Asterisk is installed in the /usr/src/Asterisk1.2.12.1   path . I have
untar the ast-ax-snmpd package in the same /usr/src/ directory .Made the
patching and after that tried  " make " . Is the procedure which I have
followed is correct .  Please let me know .
 
Thanks &  Regards,
Dhannya Chandran. 



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[asterisk-users] Asterisk Recording Interface (ARI) integration with Asterisk 1.4

[Kashif Naeem]

Hello All

Has anyone integrated ARI with Asterisk 1.4 ? Is there any manual or steps
available ? Also let me know if someone know about any other similar
software.

Regards,

-- 
Kashif Naeem
Director
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]
Gmail: [EMAIL PROTECTED]
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
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Re: [asterisk-users] Realtime on Asterisk 1.2.24

[ram]

On 10/26/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> Does realtime work reliably on Asterisk 1.2.24?
>
> Are there any definitive guides, I can only find bits and pieces here
> and there.  Any accurate howtos would be of great help.
>
> I am missing func_realtime.so.  Where does this file come from?
> Asterisk or asterisk-addons?  I saw in one of the howtos that it is
> needed.  Is it needed for 1.2.X or 1.4.X.  Also, what about the switch
> lines in the .conf files.  Some howtos say you need them, others say to
> delete the whole file, that if for example, extensions.conf exisist,
> then Realtime wont load extensions.


Hi

I have not see any problem as of now
any of 1.2.X real time,

may be you can look integrated package
http://voiceone.it/

ram
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Re: [asterisk-users] Kirk IP600/3 Wireless Server SIP config

[Benny Amorsen]

> "RB" == Remco Barendse <[EMAIL PROTECTED]> writes:

RB> Hi list! Is anyone using the Kirk IP600/3 with SIP firmware
RB> connected to Asterisk?

Yes.

RB> Any experiences / caveats?

Make sure you keep the firmware updated. It improves rapidly.

RB> If anyone would be willing to share the dump of their IP600 config
RB> file, i would really appreciate it.

Sorry I'm not at work right now. If I get time later, I will.

RB> Is there anything special i should put in my asterisk config?

No, the IP600 is just like any other SIP device.


/Benny



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Re: [asterisk-users] Asterisk Recording Interface (ARI) integration with Asterisk 1.4

[ram]

On 10/26/07, Kashif Naeem <[EMAIL PROTECTED]> wrote:
>
> Hello All
>
> Has anyone integrated ARI with Asterisk 1.4 ? Is there any manual or steps
> available ? Also let me know if someone know about any other similar
> software.
>
>


Hi

Look at Trixbox its already integrated along with asterisk, and ISO image
available also

ram
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Re: [asterisk-users] In my messages log..

[Tzafrir Cohen]

On Thu, Oct 25, 2007 at 04:08:11PM -0700, Dominic Son wrote:
> Hi. I'm still a bit of a newb to linux, I see this in my messages log:
> Asterisk init: Id "ax" respawning too fast: disabled for 5 minutes
> 
> What does this mean?
> and how severe is it?

This is something to do with your distribution oior your local setup and
probably nothing to do with asterisk. The message comes from the init
process.

To see the command for label "ax" see:

  grep ^ax /etc/inittab

or actually:

  grep ^ax /etc/inittab | cut -d: -f3-

That command is probably with "respawn", which means that the init
process should restart it automatically. However init has noticed that
this process has failed too many times too quickly. This can easily
leave the system unusable in an infinite loop. So it stops respawning
that process for 5 minutes.

I suggest that you take this to the standard support channels of your
distribution.

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

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Re: [asterisk-users] PRI span configuration - span remains down

[Alejandro Kauffmann]

David Kennedy wrote:
> Hi
>
> While I have fixed the problem from this post, I do have another
> problem, and you have asked for a debug output here, so I'll go
> against my better instinct and reply here :)
>
> -- Making new call for cr 32774
> -- Requested transfer capability: 0x00 - SPEECH
>
>   
>> [ 00 01 0e 06 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 
>> 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
>> 
>
>   
>> Informational frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>> N(S): 007   0: 0
>> N(R): 003   P: 0
>> 44 bytes of data
>> 
> -- Restarting T203 counter
> Stopping T_203 timer
> Starting T_200 timer
>   
>> Protocol Discriminator: Q.931 (8)  len=44
>> Call Ref: len= 2 (reference 6/0x6) (Originator)
>> Message type: SETUP (5)
>> [04 03 80 90 a3]
>> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
>> Speech (0)
>>  Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
>> (16)
>>  Ext: 1  User information layer 1: A-Law (35)
>> [18 03 a9 83 86]
>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
>> Dchan: 0
>>ChanSel: Reserved
>>   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
>>   Ext: 1  Channel: 6 ]
>> [6c 0c 21 83 38 34 35 38 39 39 31 30 30 31]
>> Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: 
>> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>>   Presentation: Presentation allowed of network 
>> provided number (3)  '8458991001' ]
>> [70 0c 80 30 32 30 38 36 35 39 32 32 39 31]
>> Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown 
>> Number Plan (0)  '' ]
>> [a1]
>> Sending Complete (len= 1)
>> 
> q931.c:2881 q931_setup: call 32774 on channel 6 enters state 1 (Call 
> Initiated)
> -- Called g0/
> -- T200 counter expired, What to do...
> -- Retransmitting 48 bytes
> voip1*CLI>
>   
>> [ 00 01 0e 07 08 02 00 06 05 04 03 80 90 a3 18 03 a9 83 86 6c 0c 21 83 38 34 
>> 35 38 39 39 31 30 30 31 70 0c 80 30 32 30 38 36 35 39 32 32 39 31 a1 ]
>> 
> voip1*CLI>
>   
>> Informational frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>> N(S): 007   0: 0
>> N(R): 003   P: 1
>> 44 bytes of data
>> 
> -- Rescheduling retransmission (1)
> voip1*CLI>
> < [ 00 01 01 11 ]
> voip1*CLI>
> < Supervisory frame:
> < SAPI: 00  C/R: 0 EA: 0
> <  TEI: 000EA: 1
> < Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
> < N(R): 008 P/F: 1
> < 0 bytes of data
> -- ACKing all packets from 6 to (but not including) 8
> -- ACKing packet 7, new txqueue is -1 (-1 means empty)
> -- Since there was nothing left, stopping T200 counter
> -- Nothing left, starting T203 counter
> -- Got RR response to our frame
> -- Restarting T203 counter
> voip1*CLI>
> < [ 02 01 06 10 08 02 80 06 5a 08 03 82 ac 18 ]
> voip1*CLI>
> < Informational frame:
> < SAPI: 00  C/R: 1 EA: 0
> <  TEI: 000EA: 1
> < N(S): 003   0: 0
> < N(R): 008   P: 0
> < 10 bytes of data
> -- ACKing all packets from 7 to (but not including) 8
> -- Since there was nothing left, stopping T200 counter
> -- Stopping T203 counter since we got an ACK
> -- Nothing left, starting T203 counter
> < Protocol Discriminator: Q.931 (8)  len=10
> < Call Ref: len= 2 (reference 6/0x6) (Terminator)
> < Message type: RELEASE COMPLETE (90)
> < [08 03 82 ac 18]
> < Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
> Location: Public network serving the local user (2)
> <  Ext: 1  Cause: Requested channel not available
> (44), class = Network Congestion (resource unavailable) (2) ]
> <  Cause data 1: 18 (24)
> -- Processing IE 8 (cs0, Cause)
> q931.c:3503 q931_receive: call 32774 on channel 6 enters state 0 (Null)
> Sending Receiver Ready (4)
> voip1*CLI>
>   
>> [ 02 01 01 08 ]
>> 
> voip1*CLI>
>   
>> Supervisory frame:
>> SAPI: 00  C/R: 1 EA: 0
>>  TEI: 000EA: 1
>> Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
>> N(R): 004 P/F: 0
>> 0 bytes of data
>> 
> -- Restarting T203 counter
> -- Restarting T203 counter
> -- Channel 0/6, span 1 got hangup, cause 44
> -- Forcing restart of channel 0/6 on span 1 since channel reported in use
> voip1*CLI>
>   
>> [ 00 01 10 08 08 02 00 00 46 18 03 a9 83 86 79 01 80 ]
>> 
> voip1*CLI>
>   
>> Informational frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>> N(S): 008   0: 0
>> N(R): 004   P: 0
>> 13 bytes of data
>> 
> -- Restarting T203 counter
> Stopping T_203 timer
> Starting T_200 timer
>   
>> Protocol Discriminator: Q.931 (8)  len=13
>> Call Ref: len= 2 (reference 0/0x0) (Originator)
>> Message type: RESTART (70)
>> [18 03 a9 83 86]
>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
>> Dchan: 0
>>ChanSel: Reserved
>>   Ext: 1  Coding: 0  Number Specifie