[Asterisk-Users] Unknown RTP codec 100 received

[Nedi]

Hi all!  
I am frustrated. 
I am new to asterisk. My system is 
ASTLINUX    
if  receive a Fax on my sipura 
spa2000  
 
i get this: Feb 25 07:41:00 NOTICE[1708]: 
rtp.c:564 ast_rtp_read: Unknown RTP codec 100 
received
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[Asterisk-Users] Problem calling a ZAP channel with svn 10842

[John covici]

I am using asterisk CVS 10842 and a TDM 400p withanfxs and an fxo
module and when I dial the fxs channel it rings for a second and then
says no answer after 20 seconds.  I also have the latest Zaptel drivers.

Here is a log snippet.
Feb 25 00:53:11 VERBOSE[2015] logger.c: -- Executing
Dial("SIP/15712523171-09c1", "ZAP/1|20|trwW") in new stack
Feb 25 00:53:11 VERBOSE[2015] logger.c: -- Called 1
Feb 25 00:53:11 DEBUG[2015] channel.c: Driver for channel
'SIP/15712523171-09c1' does not support indication 3, emulating it
Feb 25 00:53:11 DEBUG[2015] channel.c: Scheduling timer at 160 sample
intervals
Feb 25 00:53:11 VERBOSE[2015] logger.c: -- Zap/1-1 is ringing
Feb 25 00:53:11 DEBUG[2015] channel.c: Generator got voice, switching
to phase locked mode
Feb 25 00:53:11 DEBUG[2015] channel.c: Scheduling timer at 0 sample
intervals
Feb 25 00:53:11 VERBOSE[2015] logger.c: -- Nobody picked up in
2 ms

Any assistance would be appreciated.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]
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Re: [Asterisk-Users] S100U and TigerJet

[Tzafrir Cohen]

On Fri, Feb 24, 2006 at 03:12:17PM +0100, [EMAIL PROTECTED] wrote:
> Hi all, this is another post about this problem.
> I installed from scratch a new Suse Linux  10.0, with latest stable
> asterisk.
> Moreover I add the lines to  /etc/udev/rules.d/50-udev.rules, in order to
> let the driver create the /dev/zap...
> 
> When I plug into usb port my TigerJet adapter, I see on /var/log/messages
> 
> Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using
> uhci_hcd and address 2
> Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver
> snd-usb-audio
> Feb 24 14:55:03 srvlnx05 kernel: zaptel: module not supported by Novell,
> setting U taint flag.
> Feb 24 14:55:03 srvlnx05 kernel: Zapata Telephony Interface Registered on
> major 196
> Feb 24 14:55:03 srvlnx05 kernel: wcusb: module not supported by Novell,
> setting U taint flag.
> Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver wcusb
> Feb 24 14:55:03 srvlnx05 kernel: Wildcard USB FXS Interface driver
> registered

That is a generic message the wcusb prints when it loads, even without a
valid device. Any specific message about registering a span or so?
Do you see it under /proc/zap at that point?

cat /proc/zap/*

If not, there is no poit of tweaking udev, as it simply did not get the
inputs to work with.

> 
> while lsusb shows
> Bus 001 Device 002: ID 06e6:831c Tiger Jet Network, Inc.
> Bus 001 Device 001: ID :
> 
> under /dev, I see "borning" /zap and children
> srvlnx05:/etc # dir /dev/zap/
> 
> drwxr-xr-x   2 root root  120 Feb 24 14:55 .
> drwxr-xr-x  14 root root15720 Feb 24 14:55 ..
> crw-rw   1 asterisk asterisk 196, 254 Feb 24 14:55 channel
> crw-rw   1 asterisk asterisk 196,   0 Feb 24 14:55 ctl
> crw-rw   1 asterisk asterisk 196, 255 Feb 24 14:55 pseudo
> crw-rw   1 asterisk asterisk 196, 253 Feb 24 14:55 timer
> 
> but NO channel 01 al all.
> I would like to know if anybody
> 1) ever succeded in having this configuration up and running.
> 2) ever succeded in having this configuration up and running with a *TRUE*
> S100U adapter from Digium.
> 3) If 2 is true *WHERE* it could be possible to buy this true adapter: on
> digium shop I was not able to find it.
> 
> My opinion is that it could be an issue related to the operating system: I
> think I should do something similar to what I did on
>  /etc/udev/rules.d/50-udev.rules in order to allow the creation of
> usb-related devices under /dev/zap. Unfortunately
> I don't know anything about Linux kernel enumeration process. Also, does
> exist any debugging tool for wcusb ?
> Wcusb is up and running, is the only in the system ( I removed the wcusb.ko
> natively present under the /extra directory)
> lsmod | grep wcu shows:
> 
> srvlnx05:~ # lsmod | grep wcu
> wcusb  19104  0
> zaptel187268  1 wcusb
> usbcore   112512  5 wcusb,snd_usb_audio,snd_usb_lib,uhci_hcd

wcusb has no problem of loading without a device connected.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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[Asterisk-Users] Re: auto provision of IP501 polycom

[Noah I. Miller]

Hi Matt - 

> I have the same problem.  I'm running CentOS, which comes 
> with vsftpd, do you know of anyway to do this using vsftpd?

I know what you mean.  I run TaoLinux on all our * machines, so they all
had vsftpd installed with the OS.  I had to replace it with ProFTPd
because I just couldn't figure out how to make vsftpd do the capitalized
usernames.  A little annoyance, but easy to fix.

- Noah
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Re: [Asterisk-Users] mpg123 alternative?

[sean cook]

Just to through another hat in the ring... I use madplay for mp3s...

[default]
mode=custom
directory=/var/lib/asterisk/mohmp3
application=/usr/bin/madplay -Q -o raw:- --mono -R 8000 -a -12


On Thu, 2006-02-23 at 15:23 -0600, Aaron Daniel wrote:
> I'd suggest using the format_mp3 program that's included in 
> asterisk-addons.  We switched to that after mpg123 wouldn't compile on 
> our newer 64bit machines, and it works like a charm, and you don't have 
> to change anything.
> 
> Aaron
> 
> Rich Adamson wrote:
> > Been using mpg123 for moh for the last two years or so. However, when
> > I have * config errors, often times get a endless stream of console
> > messages and need to kill the two mpg123 processes.
> > 
> > Is there an alternative to mpg123 that eliminates that issue?
> > 
> > I see references in musiconhold.conf relative to madplay, native file
> > format, asterisk-addons, etc. Not sure why the asterisk-addon approach
> > hasn't been moved into trunk, or if madplay is a better choice on this
> > fc3 trunk box.
> > 
> > Any suggestions?
> > 
> > 
> > 
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[Asterisk-Users] Queus seem to work different between IAX and ZAP channels

[David F Bakker]

Queus seem to work different between IAX and ZAP channels. I have a IAX 
trunk today which I want to switch over to ZAP lines. I have a menu 
option for a caller to be able to "locate" as person. What happens is a 
queue is setup with a ringroup in it. The ringroup is an external # with 
a 15 second timeout. With the IAX trunk this works. A caller hits the 
option, goes to the queue and listens to music on hold while the extenal 
# rings for 15 seconds. If no one picks up it puts the caller in the 
persons voicemail. With a ZAP trunk the queue calls the external # but 
does an xfer. So the caller doesnt hear music on hold but the extenal # 
ringing. The timeout doesnt apply anymore either so it seems it is doing 
a xfer. Any ideas?


-David
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Re: [Asterisk-Users] Is Asterisk a PBX?

[Paul]

Zach A wrote:

>Hi everybody,
>
>This question is confusing me for some time. From selling point of view
>to a customer, calling asterisk a PBX doesn't look right. According to
>the definitions of PBX or PABX, Asterisk is not just PBX but much more
>than that. My question is, how should I introduce Asterisk to a
>customer? I don't want to call it a PBX.
>
>Thanks
>
>Zach A.
>  
>
Think of it like the chassis that you can buy if you are building
specialty vehicles. You might see motor homes ranging from $50k to $500k
that use the same basic chassis and powertrain. Asterisk is a foundation
that you can build on. A PBX is just one of the many possibilities,

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Re: [Asterisk-Users] ParkAndAnnounce2 Feature Request

[Steven Andres]

I just tried this out and it didn't work, Steve. Using:

exten => 700,1,NoOp(Park and Announce)
exten => 700,n,Set(REFBY=${SIP_HEADER(Referred-By)})
exten => 700,n,NoOp(Referred-By: ${REFBY})
exten => 
700,n,ParkAndAnnounce(pbx-transfer:PARKED|20|SIP/${REFBY:5:5}|office,${EXTEN},1)

The Referred-By header doesn't exist so I can't use it:

-- Executing Set("SIP/4159524515-03b1", "REFBY=") in new stack
-- Executing NoOp("SIP/4159524515-03b1", "Referred-By: ") in new stack

And when the call timesout after 20 seconds, it returns to ${EXTEN} which is 
'700'. I want it to return to the phone that originally put the call on 
hold. Darn, thought I had a solution. Back to the drawing board...

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

Sent: Friday, February 24, 2006 2:58 PM
Subject: Re: [Asterisk-Users] ParkAndAnnounce2 Feature Request


Steven:

  I'm assuming your using IP phones registered to Asterisk in this
example. I don't do that but I use ParkandAnnounce for IP phones
registering to a SER server. To handle the call back part of your
question I use a snipet like:

exten => _700,5,SIPGetHeader(REFBY=Referred-By)
exten =>
_700,6,ParkAndAnnounce(parkedcall16:PARKED|7200|SIP/${REFBY:5:5}@|default|${EXTEN}|3)



Steven Andres wrote:

>Perhaps someone has accomplished these enhancements in the dialplan already
>(if you have, please share!), but here's what I would like to modify
>ParkAndAnnounce to do, if I were a skilled coder:
>
>SYNTAX:
> ParkAndAnnounce2(announce:template|timeout|dial|return_context|options)
>
>OPTIONS:
>a = Asterisk would insert SIP header of "Call-Info: Answer-After: 0" to
>  the dial command. This would allow the announcement to happen
>  over the target phone's speaker and not require answering a
>  ringing call. Only affects announcement, not call park return
>p = automatically make announcements to whomever originated the
>  parking and return parked calls to same. Caller ID Name will read
>  'Call Park at XXX' during announcement and 'Call Park Return'
>  during return. Currently it just says 'asterisk'
>
>EXAMPLE:
>   exten => 700,1,Answer
>   exten => 700,2,Wait(1)
>   exten => 700,3,ParkAndAnnounce2(pbx-transfer:PARKED|60|ignore|ignore|ap)
>
>In this example, let's assume that an outside caller reaches me at
>extension
>101. I blind-transfer on my SIP phone to extension 700. The caller hears
>music-on-hold and is parked in the next available slot (let's say 701). The
>app would then add the auto-answer header (thanks to the 'a' option) needed
>for our GXP-2000's and then place the call to SIP/101 (the device that
>originated the parking) with the CallerID Name set to 'Call Park at 701'.
>The call is auto-answered at the SIP phone (thanks to the SIP header) and
>then the dial plan plays "pbx-transfer", reads out the digits, and hangs
>up.
>After 60 seconds, the caller is returned to SIP/101 (this time without the
>auto-answer SIP header) with the CallerID Name set to 'Call Park Return'.
>Notice that when the 'p' option is used, the 'dial' and 'return' parameters
>are ignored.

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Re: [Asterisk-Users] ParkAndAnnounce2 Feature Request

[Steven Andres]

Yes, our phones are registered to Asterisk and are all GXP-2000's (no matter 
what everyone says about how bad they are--we love 'em). :)

I wasn't aware of the "REFBY" header. I will tinker with that. Sounds like 
that would solve the proposed 'p' option. Now I just need a method to do 
what my proposed 'a' option does (insert the 'Answer-After: 0' header). Any 
ideas?

Thanks for the quick post, Steve.

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

Sent: Friday, February 24, 2006 2:58 PM
Subject: Re: [Asterisk-Users] ParkAndAnnounce2 Feature Request


Steven:

  I'm assuming your using IP phones registered to Asterisk in this
example. I don't do that but I use ParkandAnnounce for IP phones
registering to a SER server. To handle the call back part of your
question I use a snipet like:

exten => _700,5,SIPGetHeader(REFBY=Referred-By)
exten =>
_700,6,ParkAndAnnounce(parkedcall16:PARKED|7200|SIP/${REFBY:5:5}@|default|${EXTEN}|3)



Steven Andres wrote:

>Perhaps someone has accomplished these enhancements in the dialplan already
>(if you have, please share!), but here's what I would like to modify
>ParkAndAnnounce to do, if I were a skilled coder:
>
>SYNTAX:
> ParkAndAnnounce2(announce:template|timeout|dial|return_context|options)
>
>OPTIONS:
>a = Asterisk would insert SIP header of "Call-Info: Answer-After: 0" to
>  the dial command. This would allow the announcement to happen
>  over the target phone's speaker and not require answering a
>  ringing call. Only affects announcement, not call park return
>p = automatically make announcements to whomever originated the
>  parking and return parked calls to same. Caller ID Name will read
>  'Call Park at XXX' during announcement and 'Call Park Return'
>  during return. Currently it just says 'asterisk'
>
>EXAMPLE:
>   exten => 700,1,Answer
>   exten => 700,2,Wait(1)
>   exten => 700,3,ParkAndAnnounce2(pbx-transfer:PARKED|60|ignore|ignore|ap)
>
>In this example, let's assume that an outside caller reaches me at 
>extension
>101. I blind-transfer on my SIP phone to extension 700. The caller hears
>music-on-hold and is parked in the next available slot (let's say 701). The
>app would then add the auto-answer header (thanks to the 'a' option) needed
>for our GXP-2000's and then place the call to SIP/101 (the device that
>originated the parking) with the CallerID Name set to 'Call Park at 701'.
>The call is auto-answered at the SIP phone (thanks to the SIP header) and
>then the dial plan plays "pbx-transfer", reads out the digits, and hangs 
>up.
>After 60 seconds, the caller is returned to SIP/101 (this time without the
>auto-answer SIP header) with the CallerID Name set to 'Call Park Return'.
>Notice that when the 'p' option is used, the 'dial' and 'return' parameters
>are ignored.

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Re: [Asterisk-Users] What business IP phone to use

[Cory Andrews]

Has anyone tried the Linksys SRW224P? 24 Port managed switch, 10/100, 2 Gig 
Uplink Ports, PoE:
 a.. Delivers reliable power over 10/100 Ethernet ports using IEEE 802.3af 
standard
 b.. Secure management via SSH/SSL and secure user control via 802.1x & MAC 
filtering
 c.. IGMP snooping, L2/L3 COS, queuing & scheduling makes solution ideal 
for Voice/Video
 d.. Intelligent traffic management with Rate Limiting, Policing ACLs, and 
Storm control
All that for around $450we have not put one of these through any heavy 
duty production stress tests, but I was amazed at the features on this thing 
for the price.


Cory J Andrews

VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
- Original Message - 
From: "mustardman29" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 


Sent: Friday, February 24, 2006 8:01 PM
Subject: RE: [Asterisk-Users] What business IP phone to use



Interesting,

So are there any sort of specifications to look for?  What your talking
about does not sound like a managed vs unmanaged issue.  More like cheap
crap vs half decent.  I would never want any switch to drop packets VoIP 
or

not.  Does not sound like QoS could help resolve that or jitter if the
conflicting packets both have SIP priority.

Managed switches used to imply higher quality but I think we are starting 
to
see cheap and crappy managed switches coming onto the market.  I would 
still
choose a $500 unmanaged switch over a $100 managed switch.  If the switch 
is

doing it's job you should never have to view what is going on in there
anyways.


-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Friday, February 24, 2006 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] What business IP phone to use


> Aha, micro seconds in networking terms is normally written
usecs or us
> (actually it's the greek letter mu as in ulaw) rather than ms which
> are milliseconds seconds - what had me puzzled was that it
was stated
> that this could harm the voice path!
>
> > The difference can also cause unnecessary delays and
therefor echo
> > in the path. For example, procurve switches typically have 13ms
> > switching time, the high-end netgears about 21ms. As soon as you
> > stack a couple of switches you are talking 26ms vs 42ms
extra delay in the path!
>
> There is then only 8 usecs between the two switches, how on earth
> would this make any difference to the voice path at all?
Let alone induce any echo...
>
> Obviously the originally poster didn't understand the
difference. And
> based on this, he's probably advising people not to use Netgear
> switches for voice, oh dear.

I'll jump in here to make a couple of comments relative to
ethernet switches.
Not all switches are created equal!!!

If you take the cover off a switch, write down the part
numbers for the chips used, and read the doc on those chips,
you'll see major differences.
(We've actually tested several switches over the past several
years in real customer's networks as well.)

Many entry level switches on the market have only minimal
buffering for inbound and outbound packets. Its not uncommon
for output buffers to be limited to one or two packets, and
as a user, you can't chnage it.

Port congestion frequently shows up when two (or more)
devices connected to a switch (assume 100 mbs for now) try to
communicate via a single upstream port (assume 100 mbs for
now). The instantanous offered traffic is essentially 200
mbs, and the switch is expected to send that traffic out via
a 100 mbs port. For those devices with minimal buffering,
packets will be dropped. For newer switches with deeper
buffers, "some" packets will be held up in the chip's
internal queue waiting to get on the outbound port's wire.
The delay in the buffer will become jitter, and depending
upon exactly how many ports are contending for the outboud
port, the jitter _can_ become noticable. (That _is_ one of
the reasons why some switch vendors support QoS.)

One can talk about "wire speed throughput", etc, and it
doesn't mean squat. Those are all marketing and sales words,
not engineering specs.

There are plenty of very well known switch vendors that
purchase switches from other manufacturers and put their
names on the front covers. Some of those have characteristics
as noted above, while others manage the buffering and queuing
much better then what their marketing/sales words imply.

Its fairly common to see engineers in large corporate
networks using workgroup switches to consolidate traffic from
multiple wiring closets, and not pay any attention whatsoever
to "dropped packets" in the switches.
That's about the time when senior mgmt intervens and asks an
external company to assess their network performance to
resolve the internal fingerpointing. Our company has
completed many of these.

The _only_ way to k

Re: [Asterisk-Users] Linksys WIP300 WiFi Phone

[BJ Weschke]

On 2/24/06, Philip Edelbrock <[EMAIL PROTECTED]> wrote:
>
> Philip Edelbrock wrote:
> >
> > Whoo hoo!  I just received my WIP300 from voipsupply.  I have to let it
> > charge before I can play with it.
> >
>
> After it charged and I started using it, I had three crashes.  Once
> during a call (exactly 3 minutes into it, according to the frozen
> display), and twice while the phone wasn't in use.  When I woke the
> phone up it had a blank white display.  To unhang the phone, I removed
> the battery.  My settings were lost after two of the crashes (it's
> possible that the first crash was before I had any significant settings
> and so didn't notice them gone).  With so many settings (multiple wifi
> configs, sip configs, email, phone preferences, etc.) it is quite
> painful to have to start over.
>
> Are there others who have this phone?  Have you had it crash like this?
>  I'm wondering if this is a firmware issue, or if I have a defective
> phone. :'(
>

 I've just plugged mine back into the charger after having used it
nearly all day. I didn't have any of the problems you've described.
Sorry you're having such bad luck with it. I'm not certain what the
phones are rated to do, but I probably got better than 3 hours talk
time on it today which is definitely the best I've gotten with any of
the WiFi phones up to this point.

 BJ

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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RE: [Asterisk-Users] What business IP phone to use

[mustardman29]

Interesting,

So are there any sort of specifications to look for?  What your talking
about does not sound like a managed vs unmanaged issue.  More like cheap
crap vs half decent.  I would never want any switch to drop packets VoIP or
not.  Does not sound like QoS could help resolve that or jitter if the
conflicting packets both have SIP priority.

Managed switches used to imply higher quality but I think we are starting to
see cheap and crappy managed switches coming onto the market.  I would still
choose a $500 unmanaged switch over a $100 managed switch.  If the switch is
doing it's job you should never have to view what is going on in there
anyways.

> -Original Message-
> From: Rich Adamson [mailto:[EMAIL PROTECTED] 
> Sent: Friday, February 24, 2006 9:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] What business IP phone to use
> 
> 
> > Aha, micro seconds in networking terms is normally written 
> usecs or us 
> > (actually it's the greek letter mu as in ulaw) rather than ms which 
> > are milliseconds seconds - what had me puzzled was that it 
> was stated 
> > that this could harm the voice path!
> > 
> > > The difference can also cause unnecessary delays and 
> therefor echo 
> > > in the path. For example, procurve switches typically have 13ms 
> > > switching time, the high-end netgears about 21ms. As soon as you 
> > > stack a couple of switches you are talking 26ms vs 42ms 
> extra delay in the path!
> > 
> > There is then only 8 usecs between the two switches, how on earth 
> > would this make any difference to the voice path at all? 
> Let alone induce any echo...
> > 
> > Obviously the originally poster didn't understand the 
> difference. And 
> > based on this, he's probably advising people not to use Netgear 
> > switches for voice, oh dear.
> 
> I'll jump in here to make a couple of comments relative to 
> ethernet switches.
> Not all switches are created equal!!!
> 
> If you take the cover off a switch, write down the part 
> numbers for the chips used, and read the doc on those chips, 
> you'll see major differences.
> (We've actually tested several switches over the past several 
> years in real customer's networks as well.)
> 
> Many entry level switches on the market have only minimal 
> buffering for inbound and outbound packets. Its not uncommon 
> for output buffers to be limited to one or two packets, and 
> as a user, you can't chnage it.
> 
> Port congestion frequently shows up when two (or more) 
> devices connected to a switch (assume 100 mbs for now) try to 
> communicate via a single upstream port (assume 100 mbs for 
> now). The instantanous offered traffic is essentially 200 
> mbs, and the switch is expected to send that traffic out via 
> a 100 mbs port. For those devices with minimal buffering, 
> packets will be dropped. For newer switches with deeper 
> buffers, "some" packets will be held up in the chip's 
> internal queue waiting to get on the outbound port's wire. 
> The delay in the buffer will become jitter, and depending 
> upon exactly how many ports are contending for the outboud 
> port, the jitter _can_ become noticable. (That _is_ one of 
> the reasons why some switch vendors support QoS.)
> 
> One can talk about "wire speed throughput", etc, and it 
> doesn't mean squat. Those are all marketing and sales words, 
> not engineering specs.
> 
> There are plenty of very well known switch vendors that 
> purchase switches from other manufacturers and put their 
> names on the front covers. Some of those have characteristics 
> as noted above, while others manage the buffering and queuing 
> much better then what their marketing/sales words imply.
> 
> Its fairly common to see engineers in large corporate 
> networks using workgroup switches to consolidate traffic from 
> multiple wiring closets, and not pay any attention whatsoever 
> to "dropped packets" in the switches.
> That's about the time when senior mgmt intervens and asks an 
> external company to assess their network performance to 
> resolve the internal fingerpointing. Our company has 
> completed many of these.
> 
> The _only_ way to know for sure what a switch is doing (eg, 
> dropping pkts) is to ensure the switches have some form of 
> network management. Even the simple Dell 2708 (eight port gig 
> switch for $100) has "some" level of mgmt in it. Certainly 
> not the best, but at least you can identify some issues.
> 
> With the pricing drops that we've all seen over the last 
> couple of years, its fairly easy to find managed switches at 
> very reasonable cost. I'd _never_ using unmanaged switches in 
> any environment where critical application data flows across 
> the net, and I'd suggest voip traffic represents "critical" 
> traffic in all production networks.
> 
> 
> 
> 
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Re: [Asterisk-Users] FW: auto provision of IP501 polycom

[Matthew T. O'Connor]

I have the same problem.  I'm running CentOS, which comes with vsftpd, 
do you know of anyway to do this using vsftpd?


Thanks,

Matt



Noah Miller wrote:
Hi Again Damon - 


I just remembered that the FTP server setup can be tricky, too.  The default
username has capitalized letters, and this doesn't work with a lot of FTP
servers.  I had to use ProFTPd to get it done.  I created a user account
called plcmspip, and added the following to /etc/proftpd.conf (or wherever
you choose to put your config file):

UserAlias   PlcmSpIp plcmspip


- Noah


-- Forwarded Message
From: Noah Miller <[EMAIL PROTECTED]>
Date: Thu, 23 Feb 2006 11:34:31 -0500
To: Asterisk Users Mailing List - Non-Commercial Discussion

Cc: <[EMAIL PROTECTED]>
Conversation: auto provision of IP501 polycom
Subject: Re: auto provision of IP501 polycom

Hi Damon - 


Has anyone been able to get the IP501 to discover the FTP server IP
address (via dhcp or dns) and download 100% of the config from a
provisioning server?


Sure, works great!  I'm not sure if you got the TFTP config from the
gentleman who suggested it, but this is really dependent on what DHCP server
you are using.  For example, we use Cisco routers, and the option to add is:

option 66 ascii "xxx.xxx.xxx.xxx"

where the xxx's are the IP address (I don't think DNS names will work).


ISC's DHCP server is a little different:

option  tftp-server-name "XXX.XXX.XXX.XXX";

Yes, it should be tftp-server-name, even if you use FTP (or HTTPS, I
believe).



- Noah
 




-- End of Forwarded Message

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Re: [Asterisk-Users] Linksys WIP300 WiFi Phone

[Philip Edelbrock]

Philip Edelbrock wrote:


Whoo hoo!  I just received my WIP300 from voipsupply.  I have to let it 
charge before I can play with it.




After it charged and I started using it, I had three crashes.  Once 
during a call (exactly 3 minutes into it, according to the frozen 
display), and twice while the phone wasn't in use.  When I woke the 
phone up it had a blank white display.  To unhang the phone, I removed 
the battery.  My settings were lost after two of the crashes (it's 
possible that the first crash was before I had any significant settings 
and so didn't notice them gone).  With so many settings (multiple wifi 
configs, sip configs, email, phone preferences, etc.) it is quite 
painful to have to start over.


Are there others who have this phone?  Have you had it crash like this? 
 I'm wondering if this is a firmware issue, or if I have a defective 
phone. :'(



Phil
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Re: [Asterisk-Users] ImportVar Syntax

[Kevin Bockman]

Steven Ringwald wrote:
I am trying to use ImportVar to get some information out of a SIP/ZAP 
channel. I cannot seem to find an example of the syntax, or what 
variables I can access.


Basically, I would like to output which person is being called. i.e: 
SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21.  The 
info that I want is stored in the channel's "Direct Bridge" variable.


I have tried: ImportVar(TEST=SIP/25-6d2a|name)
exten => s,n,ImportVar(BRIDGEPEER=${CHANNEL:0:$[${LEN(${CHANNEL})} - 
2]}\,2|BRIDGEPEER)


This is what I use.  You can see all variables related to the channel by 
using DumpChan app.  From the CLI, you can also do a show channel 
.  I remember there were some got-yas when using macros.  Take a 
look using DumpChan everywhere to make sure what variables are set 
first.  Doing a show channel from the CLI will show you all variables on 
the channel after the answer, which can vary from what is available in 
your macro at the time.



Kevin
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Re: [Asterisk-Users] ParkAndAnnounce2 Feature Request

[Steve Blair]

Steven:

 I'm assuming your using IP phones registered to Asterisk in this 
example. I don't do that but I use ParkandAnnounce for IP phones 
registering to a SER server. To handle the call back part of your 
question I use a snipet like:


exten => _700,5,SIPGetHeader(REFBY=Referred-By)
exten => 
_700,6,ParkAndAnnounce(parkedcall16:PARKED|7200|SIP/${REFBY:5:5}@proxy IP>|default|${EXTEN}|3)


-Steve

Steven Andres wrote:

We've had a regular Park function in the past but recently I found the 
ParkAndAnnounce() application and I love the idea behind it. Here's a snip 
from the wiki 
(http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce) 
so that we're all talking the same language:


|| ParkAndAnnounce(announce:template|timeout|dial|return_context)
||
|| Park a call into the parkinglot and announce the call over an extension.
||
|| announce template: colon seperated list of files to announce, the word
||   PARKED will be replaced by a say_digits of the ext the call is parked 
in

|| timeout: time in seconds before the call returns into the return context.
|| dial: The app_dial style resource to call to make the announcement.
||   Console/dsp calls the console.
|| return_context: the goto style label to jump the call back into after 
timeout.



Perhaps someone has accomplished these enhancements in the dialplan already 
(if you have, please share!), but here's what I would like to modify 
ParkAndAnnounce to do, if I were a skilled coder:


SYNTAX:
ParkAndAnnounce2(announce:template|timeout|dial|return_context|options)

OPTIONS:
   a = Asterisk would insert SIP header of "Call-Info: Answer-After: 0" to
 the dial command. This would allow the announcement to happen
 over the target phone's speaker and not require answering a
 ringing call. Only affects announcement, not call park return
   p = automatically make announcements to whomever originated the
 parking and return parked calls to same. Caller ID Name will read
 'Call Park at XXX' during announcement and 'Call Park Return'
 during return. Currently it just says 'asterisk'

EXAMPLE:
  exten => 700,1,Answer
  exten => 700,2,Wait(1)
  exten => 700,3,ParkAndAnnounce2(pbx-transfer:PARKED|60|ignore|ignore|ap)

In this example, let's assume that an outside caller reaches me at extension 
101. I blind-transfer on my SIP phone to extension 700. The caller hears 
music-on-hold and is parked in the next available slot (let's say 701). The 
app would then add the auto-answer header (thanks to the 'a' option) needed 
for our GXP-2000's and then place the call to SIP/101 (the device that 
originated the parking) with the CallerID Name set to 'Call Park at 701'. 
The call is auto-answered at the SIP phone (thanks to the SIP header) and 
then the dial plan plays "pbx-transfer", reads out the digits, and hangs up. 
After 60 seconds, the caller is returned to SIP/101 (this time without the 
auto-answer SIP header) with the CallerID Name set to 'Call Park Return'. 
Notice that when the 'p' option is used, the 'dial' and 'return' parameters 
are ignored.



Does anyone else think this is interesting/valuable enough to submit to 
Mantis as a feature request? Or am I just off on my own little world?


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Re: [Asterisk-Users] OT: VoIP over bonded link

[Rich Adamson]

> > It's stupid. Don't ever connect 2 different building with copper.
> > Just wait until you get some kind of lightening hit or electrical
> > fault, but make sure you are no where near it. Use fibre.
> 
> That's a great rule of thumb, but the reality isn't quite so black and white.
> 
> A direct lightning strike is not going to draw *any* significant current 
> through the ethernet cable, as the moment you try to pull significant 
> current, those cables will either open up or vaporize due to IR losses in 
> such small gage wire.  You'll have far more current draw through the (I'm 
> assuming) metal conduit, which is already grounded.

The above is really not true with many production switches. Any form of static
electricity (regardless of whether its sourced from lightning for people) can
and have been known to blow the ethernet interface IC inside the switch.
(I can have some of our customers ship a boat load of those to you if you
want. You pay shipping. ;)

Cabletron and SMC switches seem to be the worst, and SMC manufacturers a lot
of entry level workgroup switches for other well known companies.

> Yes, you may introduce grounding loops and these will cause other (sometimes 
> significant) issues but they have all been solved before.  The best solution 
> is to simply take a pair of media converters with a fiber patch cable between 
> them, space them out adequately and hope for the best.  You're already going 
> to have a conduction path through the power supplies of the media converters 
> but with an isolation transformer and appropriate surge arrestors it's about 
> as best as you are going to be able to do.

The above concern have been a major issue with telephone equipment (eg, central 
offices) and the telco's spend a significant amount of money burying very long 
rods in the ground and interconnectng them with the CO hardware using cables 
that are larger then 1/4" in diameter (don't remember the guage anymore).
Every row of racks include the heavy ground cabling, and rack paint (etc)
is often times scrapped off between racks to ensure a solid ground.
They use special test equipment to actually measure the implementation.

Historically, Florida locations have very poor grounding which is known to
cause telco's issues for sure.


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[Asterisk-Users] Reading sound in eagi script and recognizing DTMF sounds at thesame time ?

[Robert Rozman]

Hi,

we've connected Sphinx4 through eagi script (modified eagi example) to
Asterisk. Users can now say their wishes - but for gradual evolution we
would like to provide "older" way of DTMF navigation too - can we recognize
DTMF while reading sound in eagi ?

Any advice or examples ?

Thanks in advance,

regards,

Rob.


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[Asterisk-Users] no sound in NAT configuration scenrios

[Amir Aziz]

Hello All,     I have installed asterisk at home and it is behine firewall. I have done all the changes.I have opened all the required firewall ports. I am able to dial but when I dial to some extension on the internet I get the phone ringing but when the phone is picked up there is no sound. I have check to make sure if sound card and headphones etc are all ok. I have no idea where to start troubleshooting this. any help will be appreciated. Thankyou and Regards,     Amir
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[Asterisk-Users] 7940 and Subscriptions

[Aaron Daniel]

Does anyone know if the SIP firmware for the cisco 7940/7960's support 
subscriptions to lines for hinting?  Basically we're trying to set up 
secretary phones and I'm seeing a lot of info about how the phone has to 
subscribe to a line to do it, but not sure how/if you can do it on the 
Cisco's.


Aaron
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[Asterisk-Users] ParkAndAnnounce2 Feature Request

[Steven Andres]

We've had a regular Park function in the past but recently I found the 
ParkAndAnnounce() application and I love the idea behind it. Here's a snip 
from the wiki 
(http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce) 
so that we're all talking the same language:

|| ParkAndAnnounce(announce:template|timeout|dial|return_context)
||
|| Park a call into the parkinglot and announce the call over an extension.
||
|| announce template: colon seperated list of files to announce, the word
||   PARKED will be replaced by a say_digits of the ext the call is parked 
in
|| timeout: time in seconds before the call returns into the return context.
|| dial: The app_dial style resource to call to make the announcement.
||   Console/dsp calls the console.
|| return_context: the goto style label to jump the call back into after 
timeout.


Perhaps someone has accomplished these enhancements in the dialplan already 
(if you have, please share!), but here's what I would like to modify 
ParkAndAnnounce to do, if I were a skilled coder:

SYNTAX:
 ParkAndAnnounce2(announce:template|timeout|dial|return_context|options)

OPTIONS:
a = Asterisk would insert SIP header of "Call-Info: Answer-After: 0" to
  the dial command. This would allow the announcement to happen
  over the target phone's speaker and not require answering a
  ringing call. Only affects announcement, not call park return
p = automatically make announcements to whomever originated the
  parking and return parked calls to same. Caller ID Name will read
  'Call Park at XXX' during announcement and 'Call Park Return'
  during return. Currently it just says 'asterisk'

EXAMPLE:
   exten => 700,1,Answer
   exten => 700,2,Wait(1)
   exten => 700,3,ParkAndAnnounce2(pbx-transfer:PARKED|60|ignore|ignore|ap)

In this example, let's assume that an outside caller reaches me at extension 
101. I blind-transfer on my SIP phone to extension 700. The caller hears 
music-on-hold and is parked in the next available slot (let's say 701). The 
app would then add the auto-answer header (thanks to the 'a' option) needed 
for our GXP-2000's and then place the call to SIP/101 (the device that 
originated the parking) with the CallerID Name set to 'Call Park at 701'. 
The call is auto-answered at the SIP phone (thanks to the SIP header) and 
then the dial plan plays "pbx-transfer", reads out the digits, and hangs up. 
After 60 seconds, the caller is returned to SIP/101 (this time without the 
auto-answer SIP header) with the CallerID Name set to 'Call Park Return'. 
Notice that when the 'p' option is used, the 'dial' and 'return' parameters 
are ignored.


Does anyone else think this is interesting/valuable enough to submit to 
Mantis as a feature request? Or am I just off on my own little world?

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[Asterisk-Users] Asterisk compile error

[Adam Robins]

I'm trying to 
compile Asterisk 1.2.4 on a Redhat Enterprise system, kernel  
2.4.21-27.0.2.ELsmp
I'm getting the 
following errors and then the compile stops.
 
/usr/kerberos/lib/libgssapi_krb5.so.2: undefined 
reference to `add_error_table'/usr/kerberos/lib/libgssapi_krb5.so.2: 
undefined reference to `remove_error_table'collect2: ld returned 1 exit 
status
Can anyone 
point me in the right direction?  I can't seem to find anything 
online.
 
Thanks
 




Adam S. RobinsExecutive Vice President & CIO
PHARMACENTRA, LLC 5901B Peachtree Dunwoody 
Road, Suite 380Atlanta, GA 30328
Office:  770-395-0088 x2034Fax: 
 770-395-0989Mobile: 
770-855-1360Email:  [EMAIL PROTECTED]Web:http://www.pharmacentra.com 



 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.
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[Asterisk-Users] Asterisk Topology

[Ron McCarthy]

Hi List,

Im planning on setting up asterisk for a large scale enviorment, with multiple sites.

We will be doing quite a bit of inner office calling at each site, and
want to place a smaller scale * box at each site with no PRI's, and
have that connect to our main * servers at our data center that will
have the PRI connections. 

Can this be done? I havent seen to much of this on the mailing list, im
guessing each server would talk to the main * server via a IAX trunk or
a SIP peer. Also one other key point would then be to keep the
voicemail for each office on its local * server instead of having it go
to the data center.

My main concern is the dialplan, I guess if the peer is not local it
would then go out the IAX or SIP gateway to the main * server and then
check in its dial plan/routing table there, correct?

Any help/suggesstion on this would be great!

Thanks
Ron
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[Asterisk-Users] Generating Polarity Reversal on FXS line

[Infobox Peru]

Hello list,

Is it possible to send polarity reversal signalling on FXS lines in Asterisk using  the TDM240X cards? 

I need to send that signalling to legacy PABX using TDM boards of my Asterisk box.

The unique clue I could find was:


http://lists.digium.com/pipermail/asterisk-users/2004-December/thread.html#77646


Thanks in advance,

Daniel Pizarro
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Re: [Asterisk-Users] chanspy instability

[Kevin P. Fleming]

Matt wrote:
> I too have noticed this but received no solution =\  I was running 1.2.0

Did you try it again after updating to the latest 1.2 release? Did you
report the bug on the bug tracker and provide a backtrace so someone
could try to fix it?

If not, how did you expect a solution to be created? We aren't
telepathic, you know :-)
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Re: [Asterisk-Users] Is Asterisk a PBX?

[Melcon Moraes]

I was thinking smaller... just Communications Suite

Michael Collins wrote:

Hi everybody,

This question is confusing me for some time. From selling point of


view


to a customer, calling asterisk a PBX doesn't look right. According to
the definitions of PBX or PABX, Asterisk is not just PBX but much more
than that. My question is, how should I introduce Asterisk to a
customer? I don't want to call it a PBX.

Thanks

Zach A.



Good question.  What does Cisco call their VoIP server product line?
Maybe you could use a synonym of that.  It's hard to pin down what
Asterisk "is" because it does so much.  Can't call it "just a PBX" and
it's certainly more than a "VoIP server."  You can't even call it a
"voice communications server" because it does more than that!  Is it a
"Communications Server," or maybe a "telecom/datacom server?"  I can see
it now, yet more acronyms!  How about this: Asterisk is an AVCS/DCS
(kinda like CSU/DSU): Audio/Video Communications Server/Data
Communications Server?  


We could have lots of fun with this one... :)

-MC
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RE: [Asterisk-Users] Is Asterisk a PBX?

[Colin Anderson]

Yes, it is a PBX. It is also a softswitch. It is also a VoIP or PSTN proxy.
It is also a messaging application platform. A fax server. An SMS relay
point. etc. It is all of this, and more. Taken as a whole, Asterisk is
greater than the sum of it's parts. 

This also makes it impossible to pigeonhole Asterisk into a specific
category, since it does so many different things. How you present it to
people, IMO, depends on what they want to hear. Typical PHB's do not
understand the difference between "PBX" or "Messaging platform", so it's up
to you to present it in a manner in which they can digest. This is going to
vary depending upon the needs of the prospective adopter. You will only
confuse the hell out of them by saying "Yes, but it does THIS thing and THAT
thing as well". If the prospective adopter asks: "Will it REPLACE my PBX?"
say "Yes" - no more. If they ask: "Can I fax with it?" say Yes - no more. 

Your answer will be "YES" for 90% of the questions you will get, and then
let them formulate their own conclusions. 

Of course, for guys who already know the score and aren't PHB's, you can
blurt out everything that Asterisk does and once they finish crapping their
pants, they will ask you to implement it. 


-Original Message-
From: Zach A [mailto:[EMAIL PROTECTED]
Sent: Friday, February 24, 2006 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Is Asterisk a PBX?


Hi everybody,

This question is confusing me for some time. From selling point of view
to a customer, calling asterisk a PBX doesn't look right. According to
the definitions of PBX or PABX, Asterisk is not just PBX but much more
than that. My question is, how should I introduce Asterisk to a
customer? I don't want to call it a PBX.

Thanks

Zach A.

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Re: [Asterisk-Users] fax receive using TDM400P

[Thomas Artner]

Am Friday 24 February 2006 21:17 schrieb yrving rivas:
> Thomas: does it work in your case?
>   Do anybody have the fax working w/tdm?
>

Yes, as I wrote before, receiving faxes with a tdm400p card works perfectly!


> Thomas Artner <[EMAIL PROTECTED]> escribió:
>
>   Am Friday 24 February 2006 16:48 schrieb Anton Krall:
> > Any modification made to zapata as far as echo and gains?
> >
> > Should echocancel be on or off?
>
> i have echocancel switched on, faxdetect is on, rx- and txgain is not used.
> (commented out).
>
> my var/log/messages says:
> Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
> ...
> Zaptel Version: 1.2.4 Echo Canceller: KB1
>
>
> maybe it depends on different hardware revisions?
> i don't know...
>
>
>
> tom
>
> > |-Original Message-
> > |From: [EMAIL PROTECTED]
> > |[mailto:[EMAIL PROTECTED] On Behalf Of
> > |Thomas Artner
> > |Sent: Friday, February 24, 2006 8:25 AM
> > |To: Asterisk Users Mailing List - Non-Commercial Discussion
> > |Subject: Re: [Asterisk-Users] fax receive using TDM400P
> > |
> > |Hi!
> > |
> > |I am using tdm400 cards for receiving faxes. It worked quite
> > |out of the box. I installed spandsp for the rxfax application only.
> > |
> > |I use it as phone/fax switch:
> > |All incoming calls are answered automatically to listen
> > |whether its a fax or not. If it is a fax, the call is
> > |forwarded to the buil-in fax extension, otherwise the analog
> > |phones (all on tdm400) rings.
> > |
> > |It works without problems. Its for a small company (about a
> > |few faxes per
> > |hour)
> > |
> > |
> > |Tom
> > |
> > |Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall:
> > |> Guys.
> > |>
> > |> Ive been testing how to receive faxes using TDM400P cards
> > |
> > |and so far, after
> > |
> > |> playing with gains, echocancell and echotraining on
> > |
> > |zapata.conf.. Ive ha
> > |
> > |> dno luck, faxes come in as garbage or broken or with blank lines.
> > |>
> > |> Anybody has successfully done this? Any tips.. Also I have
> > |
> > |some ideas:
> > |> 1. Is it really possible to get fxes on a fax machine using
> > |
> > |ATAs like the
> > |
> > |> sipura 2002? Even using ulaw and pass-thru, is it possible?
> > |>
> > |> 2. Since the faxes is coming from PSTN thru the card, I
> > |
> > |guess asterisk will
> > |
> > |> always stay in the middle right? No way around this.
> > |>
> > |> 3. Im also trying to receive faxes usign a TE110P card with spandsp,
> > |> unicall and E1 R2MFC, no luck also, some stuff, garbage and
> > |
> > |broken faxes.
> > |
> > |> Anybody done this sucessfuly?
> > |>
> > |> Hope anybody can share their thoughts and insight on this.
> > |>
> > |> ___
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> > |>
> > |> Asterisk-Users mailing list
> > |> To UNSUBSCRIBE or update options visit:
> > |> http://lists.digium.com/mailman/listinfo/asterisk-users
> > |
> > |--
> > |Thomas Artner
> > |___
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> > | http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
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>
>
> -
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Re: [Asterisk-Users] chanspy instability

[Matt]

I too have noticed this but received no solution =\  I was running 1.2.0

On 2/24/06, Dov Bigio <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I had 3 users spying on a call from the queue.
> On the exact time that the 4th user called the ChanSpy extension, Asterisk
> went down!
>
> Is there something wrong with ChanSpy???
>
> Thank you
> Dov
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RE: [Asterisk-Users] Problem with T1 installation

[Michael Collins]

> Nitin Joshi wrote:
> > Hi All,
> >
> > I have installed a Digium TE110P card on an Asterisk 1.2.1 system.
Its
> > connected directly to the PSTN. But I am unable to make outbound
calls
> > on the zap channels. The light on the card is green. Asterisk CLI
> > shows all 24 channels when I give the command 'zap show channels'. I
> > also noticed that Asterisk CLI shows an incoming call every few
> > seconds on the 24th channel. This must be some kind of a timing
> > signal. This is he first time I am configuring a T1 so I must have
> > done something wrong I guess.
> 
> T1s require a D (Data) channel, unless connecting to a channel bank,
It
> should be 23 voice 1 data.  Also, I would strongly suggest moving to
1.2.4
> 
> Doug

Guys,

A T1 is a T1: 24 channels in the US.  A PRI is a service that runs on
top of a T1, that is, it uses all 24 channels, but it uses a specific
channel (usually the 24th channel) for signaling.

It sounds very much like Nitin has a PRI - the 24th channel is seeing
lots of activity which is a pretty good sign of a PRI, but not always.
It could be a regular incoming call on the 24th channel of a standard
digital trunk, sometimes called a "supertrunk" by some carriers.

Nitin, can you confirm with your telecom carrier that you do indeed have
a PRI?  You need to ask them if it is a PRI circuit or a standard
24-channel digital trunk.  It looks like the framing and coding are
correct: B8ZS and ESF, which are typical for both PRI's and digital
trunks.  Finally, if it is a PRI then you'll need to know what switch
type they use: National, 4ess, 5ess, DMS100, etc.  Some carriers call
this the "protocol variant."  

Here's a sample of a zaptel.conf for my PRI here in California (Qwest):

loadzone= us
defaultzone = us
span=1,1,0,esf,b8zs # My PRI line
bchan=1-23
dchan=24


Here is a sample zapata.conf with PRI, where the switch type is 5ess:

[channels]
language=en
context=from-pstn
signalling=pri_cpe
switchtype=5ess
rxwink=300  ; Atlas seems to use long (250ms) winks
callerid=asreceived
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
channel => 1-23

Save your existing zaptel and zapata conf files and try playing with
these settings.  You might just figure it out without calling the telco,
although if you spend more than 15 minutes tinkering you're probably in
need of telco's assistance.

-MC
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RE: [Asterisk-Users] Is Asterisk a PBX?

[Michael Collins]

> Hi everybody,
> 
> This question is confusing me for some time. From selling point of
view
> to a customer, calling asterisk a PBX doesn't look right. According to
> the definitions of PBX or PABX, Asterisk is not just PBX but much more
> than that. My question is, how should I introduce Asterisk to a
> customer? I don't want to call it a PBX.
> 
> Thanks
> 
> Zach A.

Good question.  What does Cisco call their VoIP server product line?
Maybe you could use a synonym of that.  It's hard to pin down what
Asterisk "is" because it does so much.  Can't call it "just a PBX" and
it's certainly more than a "VoIP server."  You can't even call it a
"voice communications server" because it does more than that!  Is it a
"Communications Server," or maybe a "telecom/datacom server?"  I can see
it now, yet more acronyms!  How about this: Asterisk is an AVCS/DCS
(kinda like CSU/DSU): Audio/Video Communications Server/Data
Communications Server?  

We could have lots of fun with this one... :)

-MC
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Re: [Asterisk-Users] fax receive using TDM400P

[yrving rivas]

Thomas: does it work in your case?  Do anybody have the fax working w/tdm?Thomas Artner <[EMAIL PROTECTED]> escribió:  Am Friday 24 February 2006 16:48 schrieb Anton Krall:> Any modification made to zapata as far as echo and gains?>> Should echocancel be on or off?i have echocancel switched on, faxdetect is on, rx- and txgain is not used. (commented out).my var/log/messages says:Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)...Zaptel Version: 1.2.4 Echo Canceller: KB1maybe it depends on different hardware revisions?i don't know...tom>> |-Original Message-> |From: [EMAIL PROTECTED]> |[mailto:[EMAIL PROTECTED] On Behalf Of> |Thomas Artner>
 ; |Sent:
 Friday, February 24, 2006 8:25 AM> |To: Asterisk Users Mailing List - Non-Commercial Discussion> |Subject: Re: [Asterisk-Users] fax receive using TDM400P> |> |Hi!> |> |I am using tdm400 cards for receiving faxes. It worked quite> |out of the box. I installed spandsp for the rxfax application only.> |> |I use it as phone/fax switch:> |All incoming calls are answered automatically to listen> |whether its a fax or not. If it is a fax, the call is> |forwarded to the buil-in fax extension, otherwise the analog> |phones (all on tdm400) rings.> |> |It works without problems. Its for a small company (about a> |few faxes per> |hour)> |> |> |Tom> |> |Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall:> |> Guys.> |>> |> Ive been testing how to receive faxes using TDM400P cards> |> |and s
 o far,
 after> |> |> playing with gains, echocancell and echotraining on> |> |zapata.conf.. Ive ha> |> |> dno luck, faxes come in as garbage or broken or with blank lines.> |>> |> Anybody has successfully done this? Any tips.. Also I have> |> |some ideas:> |> 1. Is it really possible to get fxes on a fax machine using> |> |ATAs like the> |> |> sipura 2002? Even using ulaw and pass-thru, is it possible?> |>> |> 2. Since the faxes is coming from PSTN thru the card, I> |> |guess asterisk will> |> |> always stay in the middle right? No way around this.> |>> |> 3. Im also trying to receive faxes usign a TE110P card with spandsp,> |> unicall and E1 R2MFC, no luck also, some stuff, garbage and> |> |broken faxes.> |> |> Anybody done this sucessfuly?>
 |>> |> Hope anybody can share their thoughts and insight on this.> |>> |> ___> |> --Bandwidth and Colocation provided by Easynews.com --> |>> |> Asterisk-Users mailing list> |> To UNSUBSCRIBE or update options visit:> |> http://lists.digium.com/mailman/listinfo/asterisk-users> |> |--> |Thomas Artner> |___> |--Bandwidth and Colocation provided by Easynews.com --> |> |Asterisk-Users mailing list> |To UNSUBSCRIBE or update options visit:> | http://lists.digium.com/mailman/listinfo/asterisk-users>> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>
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Re: [Asterisk-Users] Is Asterisk a PBX?

[Jean-Michel Hiver]

Zach A a écrit :


Hi everybody,

This question is confusing me for some time. From selling point of view
to a customer, calling asterisk a PBX doesn't look right. According to
the definitions of PBX or PABX, Asterisk is not just PBX but much more
than that. My question is, how should I introduce Asterisk to a
customer? I don't want to call it a PBX.
 


Call it a telephony e-server or something. Some people like that BS.

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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[Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.

[Chris Modesitt]

Sorry Olle, I bet you wanted this from the SIP Proxy:)

Chris


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-dev] Possible Bug in SIP Stack.

Chris Modesitt wrote:
> I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is 
> APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server.   When I 
> use Asterisk version 10.0.10 everything works perfectly, however when I 
> use 1.2.4 I lose the ability to receive calls from the PSTN.  All I get 
> is the following error in my SIP Proxies error logs:
> 
>  
> 
> SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, 
> don't know where to send responseSIP/2.0 180 Ringing
> 
> From: "MODESITT,CHRIS " 
>
;tag=4fdc9d0e-1e600f94-ed7e6
23f
> 
> To: ;tag=as4fc8aa8a
> 
> Call-ID: [EMAIL PROTECTED]
> 
> CSeq: 5466974 INVITE
> 
> User-Agent: Asterisk PBX
> 
>  
> 
> I still can make outbound calls with no-problems, any ideas?
> 
>  
Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 
1.2.4 so we can compare them and see what happened?

Thanks
/Olle
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Sip read: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
To: 
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
Remote-Party-Id: "MODESITT,CHRIS " ;screen=yes;id-type=subscriber;party=calling;privacy=off
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
Max-Forwards: 69
Contact: 
Supported: replaces
Content-Type: application/sdp
Accept: application/sdp
Accept-Encoding: 
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
Content-Length: 241

v=0
o=voicecore1 509633636 509633636 IN IP4 63.98.126.237
s=Session SDP
c=IN IP4 63.98.126.237
t=0 0
m=audio 43486 RTP/AVP 0 101
a=silenceSupp:off
a=ecan:b on g168
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=rtpmap:0 PCMU/8000

16 headers, 11 lines
Using latest request as basis request
Sending to 67.137.28.10 : 5060 (non-NAT)
Found peer 'dialout'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 63.98.126.237:43486
Found description format telephone-event
Found description format PCMU
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Looking for 8019324299 in from-pstn
list_route: hop: 
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 67.137.28.10:5060
  dialparties.agi: callerid = MODESITT,CHRIS
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
To: ;tag=as1eb1110f
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 67.137.28.10:5060
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
To: ;tag=as1eb1110f
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 67.137.28.10:5060
We're at 132.1.42.180 port 13582
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
To: ;tag=as1eb1110f
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 2254 2254 IN IP4 132.1.42.180
s=session
c=IN IP4 132.1.42.180
t=0 0
m=audio 13582 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off 

[Asterisk-Users] chanspy instability

[Dov Bigio]

Hi,
 
I had 3 users spying on a call from the 
queue.
On the exact time that the 4th user called the 
ChanSpy extension, Asterisk went down!
 
Is there something wrong with 
ChanSpy???
 
Thank you
Dov
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[Asterisk-Users] Is Asterisk a PBX?

[Zach A]

Hi everybody,

This question is confusing me for some time. From selling point of view
to a customer, calling asterisk a PBX doesn't look right. According to
the definitions of PBX or PABX, Asterisk is not just PBX but much more
than that. My question is, how should I introduce Asterisk to a
customer? I don't want to call it a PBX.

Thanks

Zach A.

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[Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.

[Chris Modesitt]

Sorry Olle, I bet you wanted this from the SIP Proxy:)

Chris


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-dev] Possible Bug in SIP Stack.

Chris Modesitt wrote:
> I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is 
> APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server.   When I 
> use Asterisk version 10.0.10 everything works perfectly, however when I 
> use 1.2.4 I lose the ability to receive calls from the PSTN.  All I get 
> is the following error in my SIP Proxies error logs:
> 
>  
> 
> SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, 
> don't know where to send responseSIP/2.0 180 Ringing
> 
> From: "MODESITT,CHRIS " 
>
;tag=4fdc9d0e-1e600f94-ed7e6
23f
> 
> To: ;tag=as4fc8aa8a
> 
> Call-ID: [EMAIL PROTECTED]
> 
> CSeq: 5466974 INVITE
> 
> User-Agent: Asterisk PBX
> 
>  
> 
> I still can make outbound calls with no-problems, any ideas?
> 
>  
Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 
1.2.4 so we can compare them and see what happened?

Thanks
/Olle
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Sip read: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
To: 
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
Remote-Party-Id: "MODESITT,CHRIS " ;screen=yes;id-type=subscriber;party=calling;privacy=off
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
Max-Forwards: 69
Contact: 
Supported: replaces
Content-Type: application/sdp
Accept: application/sdp
Accept-Encoding: 
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
Content-Length: 241

v=0
o=voicecore1 509633636 509633636 IN IP4 63.98.126.237
s=Session SDP
c=IN IP4 63.98.126.237
t=0 0
m=audio 43486 RTP/AVP 0 101
a=silenceSupp:off
a=ecan:b on g168
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=rtpmap:0 PCMU/8000

16 headers, 11 lines
Using latest request as basis request
Sending to 67.137.28.10 : 5060 (non-NAT)
Found peer 'dialout'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 63.98.126.237:43486
Found description format telephone-event
Found description format PCMU
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Looking for 8019324299 in from-pstn
list_route: hop: 
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 67.137.28.10:5060
  dialparties.agi: callerid = MODESITT,CHRIS
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
To: ;tag=as1eb1110f
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 67.137.28.10:5060
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
To: ;tag=as1eb1110f
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 67.137.28.10:5060
We're at 132.1.42.180 port 13582
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.137.28.10;branch=z9hG4bK125da0ff92aa0fb40559dbdd9, 
SIP/2.0/UDP 63.98.126.237:5060;branch=z9hG4bK03635e104072f0ae
From: "MODESITT,CHRIS " ;tag=5a670be4-1e606464-ed7e623f
To: ;tag=as1eb1110f
Call-ID: [EMAIL PROTECTED]
CSeq: 5684092 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 2254 2254 IN IP4 132.1.42.180
s=session
c=IN IP4 132.1.42.180
t=0 0
m=audio 13582 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off 

[Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.

[Chris Modesitt]

I have included two files, one from asterisk 1.0.10 and one from 1.2.4. 

Thanks Olle

Chris Modesitt


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-dev] Possible Bug in SIP Stack.

Chris Modesitt wrote:
> I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is 
> APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server.   When I 
> use Asterisk version 10.0.10 everything works perfectly, however when I 
> use 1.2.4 I lose the ability to receive calls from the PSTN.  All I get 
> is the following error in my SIP Proxies error logs:
> 
>  
> 
> SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, 
> don't know where to send responseSIP/2.0 180 Ringing
> 
> From: "MODESITT,CHRIS " 
>
;tag=4fdc9d0e-1e600f94-ed7e6
23f
> 
> To: ;tag=as4fc8aa8a
> 
> Call-ID: [EMAIL PROTECTED]
> 
> CSeq: 5466974 INVITE
> 
> User-Agent: Asterisk PBX
> 
>  
> 
> I still can make outbound calls with no-problems, any ideas?
> 
>  
Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 
1.2.4 so we can compare them and see what happened?

Thanks
/Olle
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amp-development*CLI> sip debug peer 3000
SIP Debugging Enabled for IP: 208.187.197.66:16945
  dialparties.agi: callerid = MODESITT,CHRIS
We're at 132.1.42.180 port 12018
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED]:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 24 Feb 2006 19:41:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 2197 2197 IN IP4 132.1.42.180
s=session
c=IN IP4 132.1.42.180
t=0 0
m=audio 12018 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 208.187.197.66:16945
amp-development*CLI> 

Sip read: 
SIP/2.0 100 Trying
To: 
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


8 headers, 0 lines
amp-development*CLI> 

Sip read: 
SIP/2.0 180 Ringing
To: ;tag=28364ccb72fa8d15i0
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


8 headers, 0 lines
amp-development*CLI> 

Sip read: 
SIP/2.0 200 OK
To: ;tag=28364ccb72fa8d15i0
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
Contact: 3000 
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 237
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

=0p-development*CLI> 
o=- 375013 375013 IN IP4 208.187.197.66
s=-
c=IN IP4 208.187.197.66
t=0 0
m=audio 18412 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

12 headers, 12 lines
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 208.187.197.66:18412
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: 
set_destination: Parsing  for address/port to send 
to
set_destination: set destination to 208.187.197.66, port 16945
Transmitting:
ACK sip:[EMAIL PROTECTED]:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK2559f914
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
To: ;tag=28364ccb72fa8d15i0
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 208.187.197.66:16945
set_destination: Parsing  for address/port to send 
to
set_destination: set destination to 208.187.197.66, port 16945
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK05081e12
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
To: ;tag=28364ccb72fa8d15i0
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: A

[Asterisk-Users] RE: [asterisk-dev] Possible Bug in SIP Stack.

[Chris Modesitt]

I have included two files, one from asterisk 1.0.10 and one from 1.2.4. 

Thanks Olle

Chris Modesitt


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-dev] Possible Bug in SIP Stack.

Chris Modesitt wrote:
> I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is 
> APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server.   When I 
> use Asterisk version 10.0.10 everything works perfectly, however when I 
> use 1.2.4 I lose the ability to receive calls from the PSTN.  All I get 
> is the following error in my SIP Proxies error logs:
> 
>  
> 
> SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, 
> don't know where to send responseSIP/2.0 180 Ringing
> 
> From: "MODESITT,CHRIS " 
>
;tag=4fdc9d0e-1e600f94-ed7e6
23f
> 
> To: ;tag=as4fc8aa8a
> 
> Call-ID: [EMAIL PROTECTED]
> 
> CSeq: 5466974 INVITE
> 
> User-Agent: Asterisk PBX
> 
>  
> 
> I still can make outbound calls with no-problems, any ideas?
> 
>  
Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 
1.2.4 so we can compare them and see what happened?

Thanks
/Olle
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amp-development*CLI> sip debug peer 3000
SIP Debugging Enabled for IP: 208.187.197.66:16945
  dialparties.agi: callerid = MODESITT,CHRIS
We're at 132.1.42.180 port 12018
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED]:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 24 Feb 2006 19:41:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 2197 2197 IN IP4 132.1.42.180
s=session
c=IN IP4 132.1.42.180
t=0 0
m=audio 12018 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 208.187.197.66:16945
amp-development*CLI> 

Sip read: 
SIP/2.0 100 Trying
To: 
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


8 headers, 0 lines
amp-development*CLI> 

Sip read: 
SIP/2.0 180 Ringing
To: ;tag=28364ccb72fa8d15i0
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 0


8 headers, 0 lines
amp-development*CLI> 

Sip read: 
SIP/2.0 200 OK
To: ;tag=28364ccb72fa8d15i0
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK02b961b2
Contact: 3000 
Server: Sipura/SPA2002-3.1.2(a)
Content-Length: 237
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

=0p-development*CLI> 
o=- 375013 375013 IN IP4 208.187.197.66
s=-
c=IN IP4 208.187.197.66
t=0 0
m=audio 18412 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

12 headers, 12 lines
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 208.187.197.66:18412
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: 
set_destination: Parsing  for address/port to send 
to
set_destination: set destination to 208.187.197.66, port 16945
Transmitting:
ACK sip:[EMAIL PROTECTED]:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK2559f914
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
To: ;tag=28364ccb72fa8d15i0
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 208.187.197.66:16945
set_destination: Parsing  for address/port to send 
to
set_destination: set destination to 208.187.197.66, port 16945
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:16945 SIP/2.0
Via: SIP/2.0/UDP 132.1.42.180:5060;branch=z9hG4bK05081e12
From: "MODESITT,CHRIS " ;tag=as6a3d9b11
To: ;tag=28364ccb72fa8d15i0
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: A

[Asterisk-Users] Re: Missing 31 DTMF tones over ZAP

[Matt King]

We're using a TE205P.  lsmod indicates that it's using the wct4xxp driver.

Hope this helps; I'll give it a try with disabled vpmdtmf.

   Matt.

C F Wrote:

what zap device are you using?
IIRC disalbing the vpmdtmf on a 406 or 411 might help you. I think
it's done in wctxx4p.c

On 2/24/06, Matt King <[EMAIL PROTECTED]> wrote:


Hello,

I'm posting this to the list in case others run into the same issue.

I've recently been connecting * to a legacy Avaya InDEX switch over
E1 ISDN PRI here in the UK.  Everything was working OK, except that DTMF
digits were not being recognised by * when sent by the Avaya switch to
the * system.  Instead, the background noise of the call centre would be
silenced while users hit the keys on their phones - echo tests and
RecordFile produced a flatline response.

I had at first thought that the Avaya switch may not be sending
them, however this was working when * was not in the call path.

With further testing, I've found out that it is in fact only the
first 31 DTMF tones that are missing - those following are picked up
OK.  I've got no idea why this should happen, and have kludged a fix by
having the Avaya switch send out 31 'fake' tones before the user starts
entering data (using Translation inside Route List).   If anyone has
come across this before and knows of a 'proper' fix, or even what might
be causing the issue, I'd be very grateful for the information.

Hope this helps,

   Matt King, M.A. Oxon.
   Managing Director, Orderly Software Ltd.


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[Asterisk-Users] RE: [Asterisk-Users ] RE: Monitor a call in progress. (Steve Totaro)

[Max Glucksmann]

Steve,

You wrote this referring to monitoring a call in Asterisk, how about from an
IP phones LCD display screen:

>1.  go to www.google.com
>2.  type "asterisk monitor application"
>3.  click on the first result
>4.  read and implement
>5.  google is your friend 
>
>I hope I made myself clear too ;-P

Moreover, which phone can we use? We have a call shop cashier attended
feature for call shops, but still need to display the call to the booth
user...

Regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
 
Venezuela
Teléfono: (0500) MAXITEL – ext. 1011001
Fax: (0212) 953-0769
 
USA
Phone: 1 (877) 467-2877 – ext. 1011001
Fax: (954) 671-6800
BEGIN:VCARD
VERSION:2.1
N:Glucksmann;Max
FN:Max Glucksmann (Fax del trabajo)
ORG:ComTel Networks, Corp.
TITLE:Director
TEL;WORK;VOICE:+1 (877) 467-2877
TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835)
TEL;CELL;VOICE:+58 (414) 250-0909
TEL;WORK;FAX:+1 (954) 671-6800
TEL;HOME;FAX:+58 (212) 285-3320
ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de=
 Am=E9rica
EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800
REV:20051212T222729Z
END:VCARD
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[Asterisk-Users] ImportVar Syntax

[Steven Ringwald]

I am trying to use ImportVar to get some information out of a SIP/ZAP 
channel. I cannot seem to find an example of the syntax, or what 
variables I can access.


Basically, I would like to output which person is being called. i.e: 
SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21.  The 
info that I want is stored in the channel's "Direct Bridge" variable.


I have tried: ImportVar(TEST=SIP/25-6d2a|name)

which doesn't seem to do anything. Looking through the code, the thing 
that I am looking for is:


c->_bridge->name (in function "handle_showchan).

The voip-info page for ImportVar returns an error, and I couldn't find 
any occurance of ImportVar, except in pbx.c.


Thanks in advance!

Steve

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Re: [Asterisk-Users] Missing 31 DTMF tones over ZAP

[C F]

what zap device are you using?
IIRC disalbing the vpmdtmf on a 406 or 411 might help you. I think
it's done in wctxx4p.c

On 2/24/06, Matt King <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I'm posting this to the list in case others run into the same issue.
>
> I've recently been connecting * to a legacy Avaya InDEX switch over
> E1 ISDN PRI here in the UK.  Everything was working OK, except that DTMF
> digits were not being recognised by * when sent by the Avaya switch to
> the * system.  Instead, the background noise of the call centre would be
> silenced while users hit the keys on their phones - echo tests and
> RecordFile produced a flatline response.
>
> I had at first thought that the Avaya switch may not be sending
> them, however this was working when * was not in the call path.
>
> With further testing, I've found out that it is in fact only the
> first 31 DTMF tones that are missing - those following are picked up
> OK.  I've got no idea why this should happen, and have kludged a fix by
> having the Avaya switch send out 31 'fake' tones before the user starts
> entering data (using Translation inside Route List).   If anyone has
> come across this before and knows of a 'proper' fix, or even what might
> be causing the issue, I'd be very grateful for the information.
>
> Hope this helps,
>
>Matt King, M.A. Oxon.
>Managing Director, Orderly Software Ltd.
>
>
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Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

[Chuck Bunn]

Hi,

I support multiple context on one asterisk server. I have a situation 
where there is a spa that has seperate voicemail and extensions and a 
resturant on the same campus that has different extensions and 
voicemail. They both use the same asterisk server but I do need the 
ability to transfer a caller from the spa to the resturant and vise 
versa. There are seperate phone lines comming in for the spa and 
resturant as well.


Thanks

Moises Silva wrote:

it seems im not undestanding your question then. Could you provide a 
practical example?


On 2/24/06, *Chuck Bunn* < [EMAIL PROTECTED] 
> wrote:


Hi,

Okay but then how do you transfer across contexts then?

Thanks

Moises Silva wrote:

> you need to set a TRANSFER_CONTEXT, either for the transferer or
> transferee channel. I dont know why, but res_features give
priority to
> the transferee TRANSFER_CONTEXT, if not found, then use the
transferer
> TRANSFER_CONTEXT. That context is used to match the extension to
dial.
> So you can set this var to any context you want.
>
> Regards
>
> On 2/23/06, *Chuck Bunn* < [EMAIL PROTECTED]

> >> wrote:
>
> Hi,
>
> Is setting the variable _TRANSFER_CONTEXT required in
> features.conf for
> Asterisk 1.2.4? It appears from the Wiki that transfers across
> contexts
> are not possible when this is set. If it is not set can one
trasfer
> across contexts??
>
> Thanks
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>
>
>
>
> --
> "Su nombre es GNU/Linux, no solamente Linux, mas info en
> http://www.gnu.org";
>
>

>
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>
>

>
>No virus found in this incoming message.
>Checked by AVG Free Edition.
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2/24/2006
>
>

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[Asterisk-Users] disallow, allow codes

[Dov Bigio]

Hi,
 
On the general section of my sip.conf I had a 
disallow=all.
 
Then I put disallow=all, allow=g729, allow=ulaw on 
my users.
 
It didn't work until I removed the disallow=all 
from the header.
 
I know disallow=all in the header is totally 
useless in this case (since I have it for every user), but anyway, is this the 
correct behavior?
 
Thank you
Dov
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[Asterisk-Users] problems with dialing

[Will Glass-Husain]

Hi,
 
We're having problems dialing out to Asterisk from 
our Grandstream GXP-200 phones.  About 2 of 3 times, when we dial, nothing 
happens.  Looking at the console in max debug mode, there are no messages 
except the following:
 
Feb 24 10:29:20 WARNING[2475]: chan_sip.c:1208 
retrans_pkt: Maximum retries exceeded on transmission 9913b47bcd7[EMAIL PROTECTED] for seqno 4524 
(Critical Response)
 
Note: Early dial is set to Yes. DTMF is via SIP 
info.
 
The phones are connected via a wireless bridge, 
range extender, and router to the asterisk box.  Pinging the phone from the Asterisk box reveals a fairly long 
latency:
 
64 bytes from 192.168.10.100: icmp_seq=1 ttl=250 
time=1110 ms64 bytes from 192.168.10.100: icmp_seq=2 ttl=250 time=114 
ms64 bytes from 192.168.10.100: icmp_seq=3 ttl=250 time=21.8 ms64 bytes 
from 192.168.10.100: icmp_seq=4 ttl=250 time=33.4 ms64 bytes from 
192.168.10.100: icmp_seq=5 ttl=250 time=4.46 ms64 bytes from 192.168.10.100: 
icmp_seq=6 ttl=250 time=57.4 ms
 
Could this be the source of the problem?  If 
so, would appreciate tips on how to work around this.
 
Thanks in advance, WILL
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[Asterisk-Users] Missing 31 DTMF tones over ZAP

[Matt King]

Hello,

   I'm posting this to the list in case others run into the same issue.
  
   I've recently been connecting * to a legacy Avaya InDEX switch over 
E1 ISDN PRI here in the UK.  Everything was working OK, except that DTMF 
digits were not being recognised by * when sent by the Avaya switch to 
the * system.  Instead, the background noise of the call centre would be 
silenced while users hit the keys on their phones - echo tests and 
RecordFile produced a flatline response.


   I had at first thought that the Avaya switch may not be sending 
them, however this was working when * was not in the call path.


   With further testing, I've found out that it is in fact only the 
first 31 DTMF tones that are missing - those following are picked up 
OK.  I've got no idea why this should happen, and have kludged a fix by 
having the Avaya switch send out 31 'fake' tones before the user starts 
entering data (using Translation inside Route List).   If anyone has 
come across this before and knows of a 'proper' fix, or even what might 
be causing the issue, I'd be very grateful for the information.


   Hope this helps,

  Matt King, M.A. Oxon.
  Managing Director, Orderly Software Ltd.

  
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Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

[Moises Silva]

it seems im not undestanding your question then. Could you provide a practical example?On 2/24/06, Chuck Bunn <
[EMAIL PROTECTED]> wrote:Hi,Okay but then how do you transfer across contexts then?
ThanksMoises Silva wrote:> you need to set a TRANSFER_CONTEXT, either for the transferer or> transferee channel. I dont know why, but res_features give priority to> the transferee TRANSFER_CONTEXT, if not found, then use the transferer
> TRANSFER_CONTEXT. That context is used to match the extension to dial.> So you can set this var to any context you want.>> Regards>> On 2/23/06, *Chuck Bunn* <
[EMAIL PROTECTED]> [EMAIL PROTECTED]>> wrote:>> Hi,>> Is setting the variable _TRANSFER_CONTEXT required in
> features.conf for> Asterisk 1.2.4? It appears from the Wiki that transfers across> contexts> are not possible when this is set. If it is not set can one trasfer> across contexts??
>> Thanks> ___> --Bandwidth and Colocation provided by Easynews.com> <
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> --> "Su nombre es GNU/Linux, no solamente Linux, mas info en> http://www.gnu.org">>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users>>>
>>No virus found in this incoming message.>Checked by AVG Free Edition.>Version: 7.1.375 / Virus Database: 268.1.0/269 - Release Date: 2/24/2006>>___
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RE: [Asterisk-Users] What business IP phone to use

[Rich Adamson]

> Aha, micro seconds in networking terms is normally written usecs or us
> (actually it's the greek letter mu as in ulaw) rather than ms which are
> milliseconds seconds - what had me puzzled was that it was stated that this
> could harm the voice path!
> 
> > The difference can also cause unnecessary delays and therefor echo in the
> > path. For example, procurve switches typically have 13ms switching time,
> > the high-end netgears about 21ms. As soon as you stack a couple of
> > switches you are talking 26ms vs 42ms extra delay in the path!
> 
> There is then only 8 usecs between the two switches, how on earth would this
> make any difference to the voice path at all? Let alone induce any echo... 
> 
> Obviously the originally poster didn't understand the difference. And based
> on this, he's probably advising people not to use Netgear switches for
> voice, oh dear.  

I'll jump in here to make a couple of comments relative to ethernet switches.
Not all switches are created equal!!!

If you take the cover off a switch, write down the part numbers for the
chips used, and read the doc on those chips, you'll see major differences.
(We've actually tested several switches over the past several years in
real customer's networks as well.)

Many entry level switches on the market have only minimal buffering for
inbound and outbound packets. Its not uncommon for output buffers to be
limited to one or two packets, and as a user, you can't chnage it.

Port congestion frequently shows up when two (or more) devices connected
to a switch (assume 100 mbs for now) try to communicate via a single
upstream port (assume 100 mbs for now). The instantanous offered traffic
is essentially 200 mbs, and the switch is expected to send that traffic
out via a 100 mbs port. For those devices with minimal buffering, packets
will be dropped. For newer switches with deeper buffers, "some" packets
will be held up in the chip's internal queue waiting to get on the
outbound port's wire. The delay in the buffer will become jitter, and
depending upon exactly how many ports are contending for the outboud
port, the jitter _can_ become noticable. (That _is_ one of the reasons
why some switch vendors support QoS.)

One can talk about "wire speed throughput", etc, and it doesn't mean
squat. Those are all marketing and sales words, not engineering specs.

There are plenty of very well known switch vendors that purchase switches
from other manufacturers and put their names on the front covers. Some
of those have characteristics as noted above, while others manage the
buffering and queuing much better then what their marketing/sales words
imply.

Its fairly common to see engineers in large corporate networks using
workgroup switches to consolidate traffic from multiple wiring closets,
and not pay any attention whatsoever to "dropped packets" in the switches.
That's about the time when senior mgmt intervens and asks an external
company to assess their network performance to resolve the internal 
fingerpointing. Our company has completed many of these.

The _only_ way to know for sure what a switch is doing (eg, dropping pkts)
is to ensure the switches have some form of network management. Even the
simple Dell 2708 (eight port gig switch for $100) has "some" level of
mgmt in it. Certainly not the best, but at least you can identify some 
issues.

With the pricing drops that we've all seen over the last couple of years,
its fairly easy to find managed switches at very reasonable cost. I'd
_never_ using unmanaged switches in any environment where critical
application data flows across the net, and I'd suggest voip traffic
represents "critical" traffic in all production networks.


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Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

[Chuck Bunn]

Hi,

Okay but then how do you transfer across contexts then?

Thanks

Moises Silva wrote:

you need to set a TRANSFER_CONTEXT, either for the transferer or 
transferee channel. I dont know why, but res_features give priority to 
the transferee TRANSFER_CONTEXT, if not found, then use the transferer 
TRANSFER_CONTEXT. That context is used to match the extension to dial. 
So you can set this var to any context you want.


Regards

On 2/23/06, *Chuck Bunn* <[EMAIL PROTECTED] 
> wrote:


Hi,

Is setting the variable _TRANSFER_CONTEXT required in
features.conf for
Asterisk 1.2.4? It appears from the Wiki that transfers across
contexts
are not possible when this is set. If it is not set can one trasfer
across contexts??

Thanks
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Re: [Asterisk-Users] Problem with T1 installation

[Anthony Rodgers]

Are you sure you're supposed to be using E&M?

On Feb 24, 2006, at 5:39 AM, Nitin Joshi wrote:


Hi All,
 
I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its 
connected directly to the PSTN. But I am unable to make outbound calls 
on the zap channels. The light on the card is green. Asterisk CLI 
shows all 24 channels when I give the command 'zap show channels'. I 
also noticed that Asterisk CLI shows an incoming call every few 
seconds on the 24th channel. This must be some kind of a timing 
signal. This is he first time I am configuring a T1 so I must have 
done something wrong I guess.

 
These are the commands I used to load the zap module:
 
modprobe zaptel
modprobe wcte11xp
ztcfg -vvv
 
---
 
my zaptel.conf is as follows:
 
span=1,1,0,esf,b8zs
e&m=1-24
loadzone = us
defaultzone=us
--
 
the zapata.conf is as follows:
 
[trunkgroups]
[channels]
 
group=1
language=en
signalling=em_w
usecallerid=yes
callerid=asreceived
context=default
echocancel=64
echocancelwhenbridged=yes
rxgain=1.0
txgain=1.0
channel => 1-2
group=2
language=en
signalling=em_w
usecallerid=yes
callerid=asreceived
context=default
echocancel=64
echocancelwhenbridged=yes
rxgain=1.0
txgain=1.0
channel => 3-24
--
 
In extensions.conf  i have specified the following line:
 
[default]
exten => _ZX,1,Dial(zap/g1/${EXTEN},15,tr)
 
--
When I try to dial using the T1 line I get the following error :
 
Feb 24 06:56:53 NOTICE[5724]: app_dial.c:1010 dial_exec_full: Unable 
to create channel of type 'Zap' (cause 0 - Unknown)

  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/7180-a103' status is 'CHANUNAVAIL'

 
Any ideas guys?
 
Thanks and regards,
Nitin Joshi.
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Re: [Asterisk-Users] GPS-enabled cell phone/PDA

[Juergen K. Zick]

Some more recent phones have the possibility to be connected to seperate 
GSM-boxes. E.g. there is a plug-in for the (older) Nokia 9210(i)/9290(i) 
Communicators and most of the Symbian phones with Bluetooth support can be 
connected to any Bluetooth-enabled GPS-mouse ...


I think, getting the position data with a defined accuracy is not the 
problem. I'm quite satisfied with the location delivered by the CB channels 
of the base stations. Crucial is indeed, what kind of location based 
service you want to build and how the data gets to the server ... With 
flatrate contracts regarding SMS or GPRS-data it's not even a real question 
of costs anymore ...


But we slowly are getting completely OT for ASTERISK ;-) ...

For more info about "context awareness" and "location based services" 
probably take some time to read what some colleagues here are doing in 
research http://www.ist-mobilife.org/


-Jürgen






> Its my understanding the cell phone coordinates are sent to the cell phone
> provider and their equipment reads (and holds) that data. Its not part
> of any data available to you in any form unless you talk to the cell
> provider and convience them you have a valid need. Highly unlikely in
> the US anyway. Even if you could convience them to provide it, they
> would likely demaand some sort of out-of-band data transmission facility.




GSM networks have the Cell ID available to the phone, however that's not
much use without the location of the cellsite.

There are now location based services, whereby you can query the network
and they'll give out an approximate location (most cells are sectored
[6 sectors per cell) which gives a direction, the cell also knows what
power the phone is transmitting with, and the power it's received so can
make a good approximation of where the phone is (within 60 degrees
angle). However it's likely a phone will be picked up by several cells,
so the network can triangulate and make a better aproximation.

Making the information available to end-users is problematic due to
privacy issues, unless the user explicitly agrees to give the info away.

With GPS units, the info is stored in the phone and can send it out
using SMS or other means.

-
It was my impression that only a handful of cellphones have full GPS units 
in them.  Benefon and some Motorola units made for the former Nextel come 
to mind.  The Benefon units do send SMS reports, and in fact, I have 
written code to control and track these units via SMS using a Nokia 31 GSM 
terminal.  Unfortunately, aside from their unique GPS/SMS capability, the 
Benefons are not very attractive products, in my opinion.  And they are 
expensive.  The Motorola units contain Java machines and a well defined 
API for accessing the location data.  I have not worked with them.  There 
have undoubtedly been changes in the marketplace since I did this work 
about 2 years ago.


As I understand it (but don't have thorough knowledge and could be 
mistaken), other units generally only receive GPS satellite signals and 
relay the data to cellular provider networks where the actual location 
calculation is done.  This can be done with assistance of data obtained 
based on tower proximity, which jumpstarts the iterative process of 
approximation.  I think it is called assisted GPS or some such...

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[Asterisk-Users] incoming peer register problem

[Miguel]

Hi, i have several incoming sip peers (mostly ciscos) , with 1.0 i 
always registered them like this:



register => @prepago-in

[prepago-in]
type=friend
host=192.168.10.120
context = from-external
dtmfmode=rfc2833
insecure=very ; required for incoming FWD calls


Now with 1.2.4 it doesnt work any more, this is what i see in the CLI 
console



Feb 24 11:40:18 WARNING[11142]: chan_sip.c:3207 sip_register: Format for 
registration is user[:secret[:[EMAIL PROTECTED]:port][/contact] at line 154



i dont need a user  and pass in the ciscos, what should i put for "user"?

thanks
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[Asterisk-Users] Call quality problems

[Michael Welter]

I'm having difficulty with an Asterisk system.  The external party has 
very good call quality, but the internal party hears clipping and drop outs.


The WAN comes in from the Cisco IAD and into a LAN switch (DLink 
DGS-1005D w/ 802.1p) where the two public IPs are switched to different 
devices.  One device is a FireBox device controlling a separate LAN with 
VPNs.  The other device is eth0 on the Asterisk system.


On the Asterisk eth1 is a 3Com 2226 LAN switch which connects Polycom 
IP501 phones.  There are no PCs on this voice LAN.  All ports on all LAN 
switches indicate full duplex.  The quality problem doesn't appear to be 
volume related (a single call still has problems).


The Polycom IP501s use SIP to the PBX, and the PBX uses SIP to the provider.

The normal WWV time signal consists of a constant tone that is 
interrupted every second by a click.  On the Polycom, each click can be 
heard, the tone starts, but the tone is clipped and there is silence 
until the next click.


I've verified that QoS is enabled in the IAD.

I would appreciate your thoughts.

Thanks,

--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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Re: [Asterisk-Users] Voicemail 0 for operator call routing

[Bruce]

Paul Tinsley wrote:
Does anyone know of a way to specify what extension is dialed when 0 
is pressed in the voicemail system.  I have a situation where there is 
more than one secretary and they want the 0 to redirect to the 
appropriate secretary for the two groups of people.

So an example would be:
555-1234 -> voicemail -> Secretary 1
555-1235 -> voicemail -> Secretary 2

Any help would be greatly appreciated.
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You can set up a db value for each extension as to what secretary group 
they belong to. When someone 0's out, have the secretary key looked up 
and then dialed, if no value is found have it dial a default secretary.


Bruce

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Re: [Asterisk-Users] Problem with T1 installation

[Doug Lytle]

Brian Roy wrote:


 
Not totally true. A PRI is 23b 1d. A DS1 (US) is a 24 channel circuit.
 
Nitin - When you stop/start asterisk does it load all 24 channels? Any 
errors? How about zap show channel 1 in the CLI?


Learn something new every day.

Doug

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RE: [Asterisk-Users] What business IP phone to use

[mustardman29]

Anything under 1ms is so far below the threshold of perceivable sound
quality, echo, delay etc. that it's a mute point to discuss IMHO.  Not even
in any cumulative effect it may have.

I can certainly see the advantages of SNMP for remote troubleshooting but
hard to justify for small offices with less than 10 extensions.  A good
quality unmanaged switch is all you need IMHO.  Not a cheap plastic Dlink or
Linksys you buy at your local wallmart mind you.

> -Original Message-
> From: Conrad Wood [mailto:[EMAIL PROTECTED] 
> Sent: Friday, February 24, 2006 3:02 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] What business IP phone to use
> 
> On Fri, 2006-02-24 at 10:54 +1100, David Ankers wrote:
> > Are you sure those switch figures are right? 16ms delay in 
> the switch 
> > path sounds a bit long. Cisco's mid-range switches like the 
> 2950 have 
> > switching times measured in micro seconds. Then again a 
> 2626 procurve 
> > is only around $700.
> 
> I meant micro-seconds, yes - my apologies.
> The 26xx series are ok, but I had specifically the 4108 in 
> mind when I said 'good experience'.
> 
> 
> 
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Re: [Asterisk-Users] fax receive using TDM400P

[Thomas Artner]

Am Friday 24 February 2006 16:48 schrieb Anton Krall:
> Any modification made to zapata as far as echo and gains?
>
> Should echocancel be on or off?


i have echocancel switched on, faxdetect is on, rx- and txgain is not used. 
(commented out).

my var/log/messages says:
Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
...
Zaptel Version: 1.2.4 Echo Canceller: KB1


maybe it depends on different hardware revisions?
i don't know...



tom

>
> |-Original Message-
> |From: [EMAIL PROTECTED]
> |[mailto:[EMAIL PROTECTED] On Behalf Of
> |Thomas Artner
> |Sent: Friday, February 24, 2006 8:25 AM
> |To: Asterisk Users Mailing List - Non-Commercial Discussion
> |Subject: Re: [Asterisk-Users] fax receive using TDM400P
> |
> |Hi!
> |
> |I am using tdm400 cards for receiving faxes. It worked quite
> |out of the box. I installed spandsp for the rxfax application only.
> |
> |I use it as phone/fax switch:
> |All incoming calls are answered automatically to listen
> |whether its a fax or not. If it is a fax, the call is
> |forwarded to the buil-in fax extension, otherwise the analog
> |phones (all on tdm400) rings.
> |
> |It works without problems. Its for a small company (about a
> |few faxes per
> |hour)
> |
> |
> |Tom
> |
> |Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall:
> |> Guys.
> |>
> |> Ive been testing how to receive faxes using TDM400P cards
> |
> |and so far, after
> |
> |> playing with gains, echocancell and echotraining on
> |
> |zapata.conf.. Ive ha
> |
> |> dno luck, faxes come in as garbage or broken or with blank lines.
> |>
> |> Anybody has successfully done this? Any tips.. Also I have
> |
> |some ideas:
> |> 1. Is it really possible to get fxes on a fax machine using
> |
> |ATAs like the
> |
> |> sipura 2002? Even using ulaw and pass-thru, is it possible?
> |>
> |> 2. Since the faxes is coming from PSTN thru the card, I
> |
> |guess asterisk will
> |
> |> always stay in the middle right? No way around this.
> |>
> |> 3. Im also trying to receive faxes usign a TE110P card with spandsp,
> |> unicall and E1 R2MFC, no luck also, some stuff, garbage and
> |
> |broken faxes.
> |
> |> Anybody done this sucessfuly?
> |>
> |> Hope anybody can share their thoughts and insight on this.
> |>
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> |
> |--
> |Thomas Artner
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Re: [Asterisk-Users] fax receive using TDM400P

[Lee Howard]

Anton Krall wrote:


Any modification made to zapata as far as echo and gains?
 



As a rule, don't let anything manipulate the audio at all... even echo 
cancellation.  That said, I have seen situations where gain had to be 
increased.


Should echocancel be on or off? 



Off, most definitely off.  I can't imagine an echo cancellor being 
capable of knowing what is echo and what isn't echo in a fax call.


Lee.

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Re: [Asterisk-Users] fax receive using TDM400P

[Lee Howard]

Anton Krall wrote:


Well, I have the same effect on my TDM as in the E1 using unicall... Faxes
get here as garbage :( 



I really would like to see sometime some audio recordings made by 
IAXmodem for people that had problems with TDMs and faxing with rxfax/txfax.


Not that I have some hope of IAXmodem overcoming the odds, but because 
I'd like to actually see the what the TDM is doing to the audio.


Lee.
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[Asterisk-Users] Trouble Chan Spy

[David Guarnido]

Hi list,

 

I got a question:

 

When I try to ChanSpy a SIP channel I only listen one
channel, for example,

 

I call from 302 extension and I have two active channels:

 

SIP/r1-voip-1b7b   
(None)  
Up  Bridged Call(SIP/302-f1f1)

SIP/302-f1f1
[EMAIL PROTECTED] Up  Dial(SIP/[EMAIL PROTECTED]|4

 

When I try to spy this call from another extension:

 

1.SIP/301-fecc
[EMAIL PROTECTED] Up  ChanSpy(SIP/302)

2.SIP/r1-voip-1b7b   
(None)  
Up  Bridged Call(SIP/302-f1f1)

3.SIP/302-f1f1
[EMAIL PROTECTED] Up  Dial(SIP/[EMAIL PROTECTED]|4

 

I got 3 active channels, the one spying, the one that
places the call and the one that receives the call.

My problem is in the spying channel I can only hear
the one that receives the call (3) but I cannot hear the channel (2):

 

Thanks for your help,






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Re: [Asterisk-Users] GPS-enabled cell phone/PDA

[Bill Michaelson]

Date: Fri, 24 Feb 2006 14:56:54 +
From: Steve Kennedy <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] GPS-enabled cell phone/PDA

On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote:



  > Its my understanding the cell phone coordinates are sent to the cell phone
> provider and their equipment reads (and holds) that data. Its not part
> of any data available to you in any form unless you talk to the cell
> provider and convience them you have a valid need. Highly unlikely in
> the US anyway. Even if you could convience them to provide it, they
> would likely demaand some sort of out-of-band data transmission facility.
  


GSM networks have the Cell ID available to the phone, however that's not
much use without the location of the cellsite.

There are now location based services, whereby you can query the network
and they'll give out an approximate location (most cells are sectored
[6 sectors per cell) which gives a direction, the cell also knows what
power the phone is transmitting with, and the power it's received so can
make a good approximation of where the phone is (within 60 degrees
angle). However it's likely a phone will be picked up by several cells,
so the network can triangulate and make a better aproximation.

Making the information available to end-users is problematic due to
privacy issues, unless the user explicitly agrees to give the info away.

With GPS units, the info is stored in the phone and can send it out
using SMS or other means.


-
It was my impression that only a handful of cellphones have full GPS
units in them.  Benefon and some Motorola units made for the former
Nextel come to mind.  The Benefon units do send SMS reports, and in
fact, I have written code to control and track these units via SMS
using a Nokia 31 GSM terminal.  Unfortunately, aside from their unique
GPS/SMS capability, the Benefons are not very attractive products, in
my opinion.  And they are expensive.  The Motorola units contain Java
machines and a well defined API for accessing the location data.  I
have not worked with them.  There have undoubtedly been changes in the
marketplace since I did this work about 2 years ago.

As I understand it (but don't have thorough knowledge and could be
mistaken), other units generally only receive GPS satellite signals and
relay the data to cellular provider networks where the actual location
calculation is done.  This can be done with assistance of data obtained
based on tower proximity, which jumpstarts the iterative process of
approximation.  I think it is called assisted GPS or some such...


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Re: [Asterisk-Users] Analyzer for Milliwatt

[Matt Roth]

Andrew Kohlsmith wrote:

What is being discussed here is basically what I was planning on doing for an 
automatic VOIP quality check.  Using miliwatt and analyzing it for 
pop/jitter/etc as well as sending other known waveforms and comparing what 
was received to what was expected and coming up with some "quality" number 
which would be fed back to the dialplan to adjust the least-cost routing 
paths.  Essentially come up with a "least cost but still good quality" 
routing.  :-)


I've done absolutely nothing other than a little research and a lot of 
thinking about how to do it though.  I did some research on digital click/pop 
removal for records as a way to detect poor quality, and then also some 
monkeying around with coppice's excellent DSP routines in spandsp.


-A.


Andrew,

This sounds like a programming project.  Something like a stripped down 
softphone (or possibly a plugin to an existing phone) with the ability 
to analyze the Milliwatt signal for variations/quality problems.  The 
ability to analyze other known waveforms would add a lot of value.


I suggest proposing your ideas to the -dev list or #asterisk-dev on 
FreeNode.  Someone else (I can't recall who) is working with SIPP in 
order to get it to pass the full RTP stream, instead of just the SIP 
signaling.  I believe that analyzing the quality of the RTP stream is 
still an open issue.  If it could be handled on a 1-to-1 basis by the 
call endpoints, it sounds like an elegant and scalable solution.


Currently, testing the scalability of an Asterisk system is a bit of a 
black art.  We did some work with an Abacus 5000 
, 
but they have a couple of significant drawbacks.  It was capable of 
originating and terminating hundreds of SIP calls, but it could only do 
audio quality analysis on up to 64 of them.  It is also a VERY expensive 
piece of equipment.


I'm very interested in your project, because our production system will 
push the vertical scalability of Asterisk.  So far we've handled 100 
concurrent calls with digital recording on a single server in a live 
environment with no quality issues, but the number of calls is going to 
increase to the 400-500 range as we add clients to the box.  The ability 
to test the results of the increased number of calls prior to going live 
could save me a LOT of headaches.  As such, your project is of 
significant value to myself as well as the community at large.  Please 
pursue it with the development community, and don't hesitate to contact 
me if needed.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call

[Mahilal Silva]

Mike,
Were you able to get this working?
Even after with a entry in the dialplan.xml does not work for me.
 
Thanks,
Ken 
On 6/20/05, Michael J. Tubby B.Sc (Hons) G8TIC <[EMAIL PROTECTED]> wrote:
Andrew,I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?When you say "mapped", dou mean that it needs an explicit entry in the
dialplan.xml like:    Mike- Original Message -
From: "Andrew Latham" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"<
asterisk-users@lists.digium.com>Sent: Thursday, June 16, 2005 2:53 PMSubject: Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get #towork during a call# and * are mapped later in the SIP(Default/MAC).cnf it has a section
in the manual if you want to see why.On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC <[EMAIL PROTECTED]>wrote:>> Gents,>> I've built an Asterisk system to replace our PBX at work and have Cisco
> 7960 phones (SIP 7.4) running with Asterisk 1.0.7.>> How to I get Asterisk to recognise the '#' being pressed during a call?>> In sip.conf I have entries likle this:>> [2001]
> type=friend> context=local-phone> auth=md5> username=2001> secret=xyzzy> callerid=Jack Tubby <2001>> host=dynamic> nat=no
> canreinvite=no> dtmfmode=rfc2833> incominglimit=2> [EMAIL PROTECTED]> disallow=all> allow=alaw> allow=ulaw> callgroup=2> pickupgroup=2
>> and in the SIPDefault.cnf for the phones I have:>> # Inband DTMF Settings (0-disable, 1-enable (default))> dtmf_inband: 1>> # Out of band DTMF Settings (none-disable, avt-avt enable (default),
> avt_always - always avt )> dtmf_outofband: avt>> # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default),> 4-3db up, 5-6dB up)> dtmf_db_level: 3>
> DTMF works for voicemail and for remote services over both analogue Zap> channels and digital (ISDN) channels.>> Asterisk doesn't appear to be 'monitoring' the audio so I can't get to> Asterisk
> features like Asterisk's transfer, parked calls and one-tuch-record...>> Am I missing something?>>> Mike>>> ___
> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:>> http://lists.digium.com/mailman/listinfo/asterisk-users>>
--Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)WWW: http://lathama.comEmail: [EMAIL PROTECTED] - 
[EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email!___
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Re: [Asterisk-Users] Problem with T1 installation

[Brian Roy]

On 2/24/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
T1s require a D (Data) channel, unless connecting to a channel bank, Itshould be 23 voice 1 data.  Also, I would strongly suggest moving to 
1.2.4
 
 
Not totally true. A PRI is 23b 1d. A DS1 (US) is a 24 channel circuit.
 
Nitin - When you stop/start asterisk does it load all 24 channels? Any errors? How about zap show channel 1 in the CLI?
 
-Brian
 
  
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Re: [Asterisk-Users] Asterisk Contact Center

[Matt Florell]

I have talked with of a couple people(don't remember their names) who
had this developed on a contract basis for the 1.0 Asterisk code tree,
they did not want to release it to GPL because of how much it cost
them and the fact that their code supposedly won't run on 1.2, but it
is technically possible and has been talked about many times on the
list.

There was even a feature request for this over 2 years ago, it was
dismissed as being too hard:
http://bugs.digium.com/view.php?id=633

There was talk of this last month on the dev list:
http://threebit.net/mail-archive/asterisk-dev/msg2.html

Maybe it's time for somebody to organize a bounty for it:
http://www.voip-info.org/wiki-Asterisk+bounty


MATT---


On 2/24/06, Stephen Arulraj <[EMAIL PROTECTED]> wrote:
>  Can the asterisk support a "coaching function" for the Supervisor to tap
> onto a call and coach the agent as she speaks to the customer without the
> customer hearing it.?
>
>  Customer database management softward (or CRM) – is this being included?
>
>  Best regards
>  Stephen
>
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Re: [Asterisk-Users] mpg123 alternative?

[Faris Raouf]

Thank you Lee, Dave, Rich, Joel and of course also Kevin. Between your 
various messages I finally understand what's happening and how it works, 
and have actually converted everything to alaw, ulaw, slin and gsm and 
am not actually using the mp3 side of things at all anymore. The 
difference is very noticeable in terms of MOH quality except when using 
g729 on the link between Asterisk and the phone - the sound quality 
seems worse there.


I have two related questions though which I'm hoping someone can help with:

We use alaw, ulaw, gsm and g729 between phones and asterisk. Sox can 
convert files to ulaw, alaw and gsm (not to mention slin) but what about 
g729? Is there such a thing as a format that won't need transcoding when 
using g729 links, or is this not something that is possible?


And what is the signed linear (slin) format used for?

Thanks,

Faris.




Lee Archer wrote:

Check out the musiconhold.conf.sample in the asterisksource/configs
folder.

Lee 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 23 February 2006 18:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] mpg123 alternative?

Ah! Now this is actually something I've not been able to get my head
around:

 > Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk,
which  > has its own MP3 player.

Can anybody tell me where this built-in MP3 player is in 1.2.x/how do I
use it ?

I still seem to have the usual two mpg123 processes running with 1.2.4,
with whatever music on hold is set in musiconhold.conf

I'm sure it is very obvious, but I can't for the life of me figure out
what I'm supposed to do to use the built-in MP3 player facilities.


I just have the following in my musiconhold.conf:

[default]
mode=mp3
directory=/var/lib/asterisk/mohmp3
random=yes


Faris.





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RE: [Asterisk-Users] GPS-enabled cell phone/PDA

[David Ankers]

In the UK this is common; several websites enable you to track a cell phone
online:

http://www.traceamobile.co.uk/

and another:

http://www.followus.co.uk/

Works the same way that Steve stated... The police here in Australia have
been using this since the late 90s. 

Interesting article:

http://www.guardian.co.uk/g2/story/0,,1699080,00.html




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent: Saturday, 25 February 2006 1:57 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] GPS-enabled cell phone/PDA

On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote:

> Its my understanding the cell phone coordinates are sent to the cell phone
> provider and their equipment reads (and holds) that data. Its not part
> of any data available to you in any form unless you talk to the cell
> provider and convience them you have a valid need. Highly unlikely in
> the US anyway. Even if you could convience them to provide it, they
> would likely demaand some sort of out-of-band data transmission facility.

GSM networks have the Cell ID available to the phone, however that's not
much use without the location of the cellsite.

There are now location based services, whereby you can query the network
and they'll give out an approximate location (most cells are sectored
[6 sectors per cell) which gives a direction, the cell also knows what
power the phone is transmitting with, and the power it's received so can
make a good approximation of where the phone is (within 60 degrees
angle). However it's likely a phone will be picked up by several cells,
so the network can triangulate and make a better aproximation.

Making the information available to end-users is problematic due to
privacy issues, unless the user explicitly agrees to give the info away.

With GPS units, the info is stored in the phone and can send it out
using SMS or other means.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

[Moises Silva]

you need to set a TRANSFER_CONTEXT, either for the transferer or
transferee channel. I dont know why, but res_features give priority to
the transferee TRANSFER_CONTEXT, if not found, then use the transferer
TRANSFER_CONTEXT. That context is used to match the extension to dial.
So you can set this var to any context you want.

RegardsOn 2/23/06, Chuck Bunn <[EMAIL PROTECTED]> wrote:
Hi,Is setting the variable _TRANSFER_CONTEXT required in features.conf forAsterisk 1.2.4? It appears from the Wiki that transfers across contextsare not possible when this is set. If it is not set can one trasfer
across contexts??Thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
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RE: [Asterisk-Users] fax receive using TDM400P

[Anton Krall]

Any modification made to zapata as far as echo and gains?

Should echocancel be on or off? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Thomas Artner
|Sent: Friday, February 24, 2006 8:25 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] fax receive using TDM400P
|
|Hi!
|
|I am using tdm400 cards for receiving faxes. It worked quite 
|out of the box. I installed spandsp for the rxfax application only.
|
|I use it as phone/fax switch:
|All incoming calls are answered automatically to listen 
|whether its a fax or not. If it is a fax, the call is 
|forwarded to the buil-in fax extension, otherwise the analog 
|phones (all on tdm400) rings.
|
|It works without problems. Its for a small company (about a 
|few faxes per
|hour)
|
|
|Tom
|
|
|
|
|Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall:
|> Guys.
|>
|> Ive been testing how to receive faxes using TDM400P cards 
|and so far, after
|> playing with gains, echocancell and echotraining on 
|zapata.conf.. Ive ha
|> dno luck, faxes come in as garbage or broken or with blank lines.
|>
|> Anybody has successfully done this? Any tips.. Also I have 
|some ideas:
|>
|> 1. Is it really possible to get fxes on a fax machine using 
|ATAs like the
|> sipura 2002? Even using ulaw and pass-thru, is it possible?
|>
|> 2. Since the faxes is coming from PSTN thru the card, I 
|guess asterisk will
|> always stay in the middle right? No way around this.
|>
|> 3. Im also trying to receive faxes usign a TE110P card with spandsp,
|> unicall and E1 R2MFC, no luck also, some stuff, garbage and 
|broken faxes.
|> Anybody done this sucessfuly?
|>
|> Hope anybody can share their thoughts and insight on this.
|>
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|
|-- 
|Thomas Artner
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RE: [Asterisk-Users] What business IP phone to use

[Douglas Garstang]

Polycom does support Asterisk, Asterisk Business Edition.

-Original Message-
From: Michael Graves [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 23, 2006 6:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] What business IP phone to use


On Wed, 22 Feb 2006 18:02:27 -0800, mustardman29 wrote:
>Just the person I have been looking for.  If you don't mind, would it be
>possible to get your opinion on feature for feature comparisons between the
>501 and 480i CT(not including cordless phone).
>
>Things like programmable buttons, display, dialing button quality, and most
>importantly, handset and speakerphone quality.
>
>Any info would be greatly appreciated.

I used the IP600 for about a year on my desk, and several IP500s
elsewhere around the place. It's a home office but I work from home
full time so it's a real working office environment.

I found that the physical quality of the Polycom phones was absolutely
top notch. They're a joy to use. Completely professional and very
reliable. But they're not perfect. They're a little harder to
provision. They're very configurable but that also adds to the
complexity. I had mine TFTP loading firmware and a common speed dial
directory from an XML file on my Astlinux server. The phones take a
fair amount of time to boot and force a reboot when you change many of
their settings. You can spend an afternoon repeatedly rebooting the
phone as you manually work out its initial configuration. Of course
Polycom doesn't support Asterisk, but others seem to fill this void
well enough.

The IP600 and IP500 are very similar but the differences are
considerable. The IP600 supports 6 line buttons and has a much better
LCD. Higher resolution, but still not backlit. Once you've used the 600
it'll be hard to go
back to the 500 just because the display is not as nice. The IP500
provides only 3 line buttons. Both phones support multiple
registrations.

The Aastra 480 is the only thing that I've seen that comes close to the
Polycom's. Physically it's just about as solid. Not quite as hefty in
the hand, but very nice. The LCD display is backlit. This is a major
advantage if you ever work in dim lighting. All other
manufacturers...LISTEN UP...this is a really big deal! I can't believe
how long its taken for someone to realise this fact.

Aastra configuration was a LOT easier both manually on the phone and
remotely. The on-phone menus are very easy to navigate and I almost
didn't bother setting up the central provisioning. With only a few
phones I could get by without it. Firmware and configs can be loaded
via tftp, ftp or http.

The on-phone directory and call logs are comparable on all three the I
have used. Actually, I prefer the way SNOM phones handle this as they
require fewer button presses. The Aastra phone makes it especially easy
to delete an entire call log with only a couple of button presses.

The 480 supports up to 9 lines with any 4 active at on time, or so I'm
told. I have mine registered for four lines so that incomming PSTN,
FWD, Gizmo and Skype calls each ring a different line. The latest
firmware supposedly support BLF indications but I've not used this.
It's really easy to assign speed dials to the six programmable keys on
the LCD. In fact, almost all of the buttons can be reassigned to new
functions. Also you can write XML applications that put the LCD to work
as an interactive menu.

Mostly I live and die by speakerphone quality. I think that the
Polycom's have a little edge on the Aastra phone, but not by much. If I
need to rework my entire system I'll probably migrate to all Aastra
phones.

Audio quality using the handset is excellent on all of them. Even on
the cordless handset with the 480i CT.

They all support POE...which I use to keep the phone system up during
power failures. I had to buy the injectors separately for the Aastra &
IP600 phones. The IP500s came with injector cables. 

The big dissappointment in my SIP phone testing was the Zultys 4x5. It
just feels cheap and many functions are too counterintuitive. I really
like the idea of the local FXO but they were never able to tell me how
to get the FXO port forwarded to the PBX for VM. Zultys provides no end
user support except through dealers and the dealers I dealt with didn't
know much about the specifics of the Zultys firmware.

Also, I'm curious about the newest SNOM phones. Some time ago I used a
SNOM 200 and like the way the web based I/F was integrated into the use
of the phone beyond simply configuration. You could access the speed
dials and place a call from the web I/F. You could also dial the phone
from a link or shortcut to a url pointed at the phone. That's a fair
substitute for desktop TAPI. If they've taken this any further it could
be very good.

I've not tried any of the lesser phones like Grandstream or Linksys.
Life's too short to use a cheap phoneat least if your budget
permits better.

Michael Graves

--
Michael Graves   

RE: [Asterisk-Users] fax receive using TDM400P

[Anton Krall]

Well, I have the same effect on my TDM as in the E1 using unicall... Faxes
get here as garbage :( 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Rich Adamson
|Sent: Friday, February 24, 2006 7:28 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] fax receive using TDM400P
|
|> Ive been testing how to receive faxes using TDM400P cards 
|and so far, 
|> after playing with gains, echocancell and echotraining on 
|> zapata.conf.. Ive ha dno luck, faxes come in as garbage or 
|broken or with blank lines.
|> 
|> Anybody has successfully done this? Any tips.. Also I have 
|some ideas:
|> 
|> 1. Is it really possible to get fxes on a fax machine using 
|ATAs like 
|> the sipura 2002? Even using ulaw and pass-thru, is it possible?
|> 
|> 2. Since the faxes is coming from PSTN thru the card, I 
|guess asterisk 
|> will always stay in the middle right? No way around this.
|> 
|> 3. Im also trying to receive faxes usign a TE110P card with spandsp, 
|> unicall and E1 R2MFC, no luck also, some stuff, garbage and broken 
|> faxes. Anybody done this sucessfuly?
|> 
|> Hope anybody can share their thoughts and insight on this.
|
|Using the TDM400 card for any form of fax'ing (or modem use) 
|is well known to be unreliable and, in most cases, totally 
|unusable. The issue has been discussed many times over the 
|last two years or so. There are no known workarounds.
|
|Its my understanding that lots of folks have spandsp working 
|via T1 and/or PRI interfaces. The issues associated with the 
|TDM400 card do not apply to the T1 cards.
|
|
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Re: [Asterisk-Users] Keep getting message in logs that pbx.c cannot find extension context 'default'

[Moises Silva]

do you have a defaultcontext=something parameter in sip.conf [general] section?? If not, the default is... em  "default"

RegardsOn 2/23/06, Chuck Bunn <[EMAIL PROTECTED]> wrote:
Hi,I am getting repeated messages in my logs with the following:*Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default'Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be
handled, bad request: [EMAIL PROTECTED]Feb 23 07:56:12 NOTICE[2470] pbx.c: Cannot find extension context 'default'
Feb 23 07:56:12 DEBUG[2470] chan_sip.c: SIP message could not behandled, bad request: [EMAIL PROTECTED]Feb 23 07:56:14 NOTICE[2470] 
pbx.c: Cannot find extension context 'default'Feb 23 07:56:14 DEBUG[2470] chan_sip.c: SIP message could not behandled, bad request: [EMAIL PROTECTED]
*I do not have a default context used in my extensions.conf - I use othernames. Am I required to have a context named 'default'??Thanks___
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Re: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

[Faris Raouf]

Thank you Armin. This is extremely useful.

Faris.


Armin Schindler wrote:
There are three possibilities to set the CallingPartyNumber (own number for 
outgoing):


1) Set(CALLERID(number)=12345)
   before Dial()

2) Dial(CAPI/contr1/12345:${EXTEN}/)

3) Dial(CAPI/contr1/${EXTEN}/d,...) and 'defaultcid=12345' in capi.conf
   with this defaultcid you can set a number for each interface in capi.conf
   and activate that by the /d option. This is useful when you combined more 
   than one interface into one group, but need to use a correct (and 
   different) number on dialout with e.g. 'g1', because the dialplan 
   doesn't know which interface will be used.


Armin

On Thu, 23 Feb 2006, Faris Raouf wrote:

Thanks for that Peter!

I think your message solved my problem: I set the master number to be in group
1 (group=1) in capi.conf and called Dial with CAPI/g1 and it worked perfectly.

However, with group=1 in capi.conf for the master number, at the moment no
matter what I do I'm getting the master number presented as the CLI. This is
fine by me because it is exactly what I want, but it is all very confusing :-)

Faris.


Peter Braidwood wrote:

I have ISDN2 and 10 MSN numbers and also have just upgraded to 1.2.4 and
chan_capi-cm and have this working completely perfectly

Capi.conf

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=en

[ISDN1]
isdnmode=msn
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=from-isdn
group=1
devices=2

bit of extensions.conf, I dial 9 for an outside line

[pstn]

exten => _9./321,1,Dial(CAPI/g1/01234567890:${EXTEN:1})
exten => _9./322,1,Dial(CAPI/g1/01234567891:${EXTEN:1})
exten => _9./323,1,Dial(CAPI/g1/01234567892:${EXTEN:1})
exten => _9./324,1,Dial(CAPI/g1/01234567893:${EXTEN:1})
exten => _9./326,1,Dial(CAPI/g1/01234567894:${EXTEN:1})
exten => _9./327,1,Dial(CAPI/g1/01234567895:${EXTEN:1})
exten => _9./328,1,Dial(CAPI/g1/01234567896:${EXTEN:1})
exten => _9./350,1,Dial(CAPI/g1/01234567897:${EXTEN:1})
exten => _9./351,1,Dial(CAPI/g1/01234567898:${EXTEN:1})
exten => _9./352,1,Dial(CAPI/g1/01234567899:${EXTEN:1})

So when extension 326 dials out the cli that is presented would be
01234567894

Contact me off list if you want any further help.

Peter Braidwood


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: 23 February 2006 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] chan_capi-cm 0.6.4 random outgoing MSN problem

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm
0.6.4

When making outgoing calls I don't seem to have any control over the CLI

that is presented to the called party -- it can be any one of the MSNs
allocated to the line, allocated on what seems to be a random basis.

This is on a BT Business Highway line (which is essentially an ISDN2e
line with two built-in analog ports), configured with 8MSNs alongside the
single the "master" digital telephone number for the line.

With a much older version of chan_capi-cm (0.5.4 I think) and Asterisk
1.0.9 it was always the "master" number that was presented, and that is
actually what I want.

Obviously the format of capi.conf has changed between these two versions

of chan_capi-cm, so maybe I'm doing something wrong. Any suggestions
would be appreciated.

Here's my capi.conf (actual numbers changed to protect the innocent!)

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
; ulaw=yes;set this, if you live in u-law world instead of
; a-law

[123456]
;  Master number for line - i.e. number for line before MSNs were
allocated
;  and which still works and still accepts incoming calls.
isdnmode=msn
msn=01234123456
; incomingmsn=*
incomingmsn=123456
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
; echotail=64
; callgroup=1
; deflect=12345678
devices=2

[123457]
; first MSN
msn=01234123457
; incomingmsn=*
incomingmsn=123457
isdnmode=msn
controller=1
softdtmf=1
accountcode=
context=isdn-in
echosquelch=1
echocancel=yes
; echotail=64
; callgroup=1
; deflect=12345678
devices=2

{repeated for next 7 MSNs}


And in extensions.conf I have:

[globals]
ISDN1=CAPI/123456


[mysip]

; GET OUTSIDE LINE (ISDN1 - dial 9)
ignorepat => 9
exten => exten => _9.,1,Dial(${ISDN1}/${EXTEN:1}/b)
exten => _9.,2,Playback(busy)
exten => _9.,3,Hangup

*

I've tried using ISDN1=CAPI/contr1 but it makes no difference.
I've tried leaving out the isdnmode=msn but it makes no difference
I've tried entering 01234123456 as the msn= line on all of the msn
entries in capi.conf but it makes no difference either.

And now I'm out of ideas and any help would be appreciated.

Thanks,

Faris.

p.s. sorry if this message is HTML. I've switched to using Thunderbird
and it is confusing the heck out of me. It claims this is a plain text
message but it doesn't look like plain text to me from this end!



_

[Asterisk-Users] Re: What business IP phone to use

[andrew matthews]

maybe you didn't want suggestions, but too bad :).

My favorite up until recently was the polycom 501 and I found it was
good quality and clear calls and priced well. but the production of te
phone is slowing down so I bought a few linksysspa941. and iVll
tell you I have a new  favorite phone. its slick, provisioning is a
breeeze and the call quality with built in qos is fantastic.

I wasn't a big fan of grandstream products they seem to be cheaply
made and i've had a few fail. but they do work.

talking about my biased opinion I don't have onee, i'm a hobby
programmer who works for a company that resells voip services and we
use polycom and linksys. I just provide support for all phones so I
kow how things work and don't work.

I hope this helps. thanks

andrew

On 2/21/06, mustardman29 <[EMAIL PROTECTED]> wrote:
>
>
> I have been struggling with this issue for about a year now.  There were
> just too many IP phones to choose from at all sorts of price points and not
> enough information about any of them.  Now I am looking at the situation
> again and if anything it has gotten worse.  There are even more phones and
> all sorts of opinions.  For every person that says phone x is great there is
> someone else complaining about it.
>
> I ended up buying a Grandstream GXP2000 and an Aastra 9133i to test so I
> pretty much know what those two phones are about.  Lot's of people talking
> about Polycom phones but they still seem to have their problems and since
> they don't officially support Asterisk I have my concerns.  I really don't
> want to have to keep buying phones to find out for myself as it get's
> expensive real fast.
>
> Is there any unbiased comparison of various phones and features anywhere.
> If someone wrote a book I'd buy it but it would probably be obsolete before
> it was published with the rate of new IP phone introductions and firmware
> revisons.  I hear some people praising the GXP2000 phones and I gotta wonder
> what they are smokin (regardless of firmware revison) so I just don't know
> who to believe anymore.
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RE: [Asterisk-Users] spandsp debug or logging

[Anton Krall]

Done..

They don't show much but they do show some problems with lost lines or
something 

Thx Bartosz  

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Bartosz Piec
|Sent: Friday, February 24, 2006 2:54 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] spandsp debug or logging
|
|Anton Krall wrote:
|> Maybe this is a stupid question but how to you enable debubg or 
|> logging on spandsp? I see you can do that for RXFAX but when people 
|> tell you to enable debug on spandsp, do they mean this with rxfax or 
|> how do you do it with spandsp?
|
|You can do it writing:
|
|exten => s,1,rxfax(/fax/file/path|debug)
|
|or the same with txfax. The logs are then written to (default) 
|/var/log/asterisk/full
|
|--
|Best regards,
|Bartosz Piec
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Re: [Asterisk-Users] S100U and TigerJet

[asterisk]

no chance, also with your scipt

 ztcfg -vvv

Zaptel Configuration
==

Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)

1 channels configured.

ZT_CHANCONFIG failed on channel 1: No such device or address (6)



   
 Jerry Glomph  
 Black 
 <[EMAIL PROTECTED]  To 
 lomph.com>Asterisk Users Mailing List -   
   Non-Commercial Discussion   
 24/02/2006 15.29 
cc 
   [EMAIL PROTECTED]
   Subject 
   Re: [Asterisk-Users] S100U and  
   TigerJet
   
   
   
   
   
   




udev drove me absolutely bat-shit in this regard; udev is a horror in many
respects.   Here's how I solved the problem, reliably:

I run this script at boot-time:

#!/bin/bash
 mkdir -p /dev/zap
 rm -f /dev/zap/ctl
 rm -f /dev/zap/channel
 rm -f /dev/zap/pseudo
 rm -f /dev/zap/timer
 rm -f /dev/zap/253
 rm -f /dev/zap/252
 rm -f /dev/zap/251
 rm -f /dev/zap/250
 mknod /dev/zap/ctl c 196 0
 mknod /dev/zap/timer c 196 253
 mknod /dev/zap/channel c 196 254
 mknod /dev/zap/pseudo c 196 255
 N=1; \
 while [ $N -lt 250 ]; do \
 rm -f /dev/zap/$N; \
 mknod /dev/zap/$N c 196 $N; \
 N=$[$N+1]; \
 done

Have had zero problems with this.



On Fri, 24 Feb 2006, [EMAIL PROTECTED] wrote:

> Hi all, this is another post about this problem.
> I installed from scratch a new Suse Linux  10.0, with latest stable
> asterisk.
> Moreover I add the lines to  /etc/udev/rules.d/50-udev.rules, in order to
> let the driver create the /dev/zap...
>
> When I plug into usb port my TigerJet adapter, I see on /var/log/messages
>
> Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using
> uhci_hcd and address 2
> Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver
> snd-usb-audio
> Feb 24 14:55:03 srvlnx05 kernel: zaptel: module not supported by Novell,
> setting U taint flag.
> Feb 24 14:55:03 srvlnx05 kernel: Zapata Telephony Interface Registered on
> major 196
> Feb 24 14:55:03 srvlnx05 kernel: wcusb: module not supported by Novell,
> setting U taint flag.
> Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver wcusb
> Feb 24 14:55:03 srvlnx05 kernel: Wildcard USB FXS Interface driver
> registered
>
> while lsusb shows
> Bus 001 Device 002: ID 06e6:831c Tiger Jet Network, Inc.
> Bus 001 Device 001: ID :
>
> under /dev, I see "borning" /zap and children
> srvlnx05:/etc # dir /dev/zap/
>
> drwxr-xr-x   2 root root  120 Feb 24 14:55 .
> drwxr-xr-x  14 root root15720 Feb 24 14:55 ..
> crw-rw   1 asterisk asterisk 196, 254 Feb 24 14:55 channel
> crw-rw   1 asterisk asterisk 196,   0 Feb 24 14:55 ctl
> crw-rw   1 asterisk asterisk 196, 255 Feb 24 14:55 pseudo
> crw-rw   1 asterisk asterisk 196, 253 Feb 24 14:55 timer
>
> but NO channel 01 al all.
> I would like to know if anybody
> 1) ever succeded in having this configuration up and running.
> 2) ever succeded in having this configuration up and running with a
*TRUE*
> S100U adapter from Digium.
> 3) If 2 is true *WHERE* it could be possible to buy this true adapter: on
> digium shop I was not able to find it.
>
> My opinion is that it could be an issue related to the operating system:
I
> think I should do something similar to what I did on
> /etc/udev/rules.d/50-udev.rules in order to allow the creation of
> usb-related devices under /dev/zap. Unfortunately
> I don't know anything about Linux kernel enumeration process. Also, does
> exist any debugging tool for wcusb ?
> Wcusb is up and running, is the only in the system ( I removed the
wcusb.ko
> natively present under the /extra directory)
> lsmod | grep wcu shows:
>
> srvlnx05:~ # lsmod | grep wcu
> wcusb  19104  0
> zaptel187268  1 wcusb
> usbcore   112512  5 wcusb,snd_usb_a

Re: [Asterisk-Users] GPS-enabled cell phone/PDA

[Rusty Dekema]

In the US, Sprint's CDMA network will do the fancy GPS+AFLT business,
but like someone else mentioned, it only sends the location data back
to Sprint's network. There is an API that you can use to access this
data for your handsets, but you have to pay some amount of money for
each location fix.

Sprint's iDEN phones (formerly Nextel) contain GPS units that can be
accessed from the phone's serial port, and I am pretty sure that the
GPS data can be accessed from a J2ME applet running in the phone. Such
an applet could then make an appropriate HTTP request to a web/app
server you run, in order to upload the data. However, the GPS data
received using this method is obtained using _only_ GPS, with no AFLT
or other form of assistance from the cellular network. The
significance of that, of course, is that you will not be able to get a
GPS fix in locations where a "regular" GPS receiver can't get a fix,
such as indoors in most cases.

-Rusty



On 2/23/06, Michael Welter <[EMAIL PROTECTED]> wrote:
> I would like to capture the lat/lon coordinates from a GPS-enabled cell
> phone or PDA.  Is this possible?  Must I subscribe to this information
> from the cellphone network provider, or can I capture it without charge?
>
> What devices will broadcast the coordinates?  Is there a device that
> will broadcast its position inband that can be captured by Asterisk?
> Can an SMS message include coordinates?
>
> The subject will willingly carry the device and will be aware that his
> location is being monitored, so privacy rights are not an issue.  The
> subject will make periodic calls to the Asterisk server in order to
> record his movements.
>
> Does anyone have experience in this area?
>
> Thanks,
> Mike
>
>
> --
> Michael Welter
> Telecom Matters Corp.
> Denver, Colorado US
> +1.303.414.4980
> [EMAIL PROTECTED]
> www.TelecomMatters.net
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[Asterisk-Users] Beer meeting at Fosdem

[Olivier.taylor]

Hi Olle,

Will u be there for the speech of Jan Janak?
If yes, you will find a guy, 1m83, with a bear and a red suit, it's me.
You also can call me on my mobile to fix the voip beer (0032495283361).

We will try to have Jan and other guys

Olivier

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RE: [Asterisk-Users] fax receive using TDM400P

[Rob Danz]

I wrestled with this for a long time, as have many others and it just
doesn't work with spandsp and asterisk alone.

Use iaxmodem and hylafax in conjunction with asterisk... it works like a
champ.  I have a single POTS line coming in so I get voice & fax with a
single number using fax detect.  

http://iaxmodem.sourceforge.net/



-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED] 
Sent: Friday, February 24, 2006 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] fax receive using TDM400P

> Ive been testing how to receive faxes using TDM400P cards and so far,
after
> playing with gains, echocancell and echotraining on zapata.conf.. Ive ha
dno
> luck, faxes come in as garbage or broken or with blank lines.
> 
> Anybody has successfully done this? Any tips.. Also I have some ideas:
> 
> 1. Is it really possible to get fxes on a fax machine using ATAs like the
> sipura 2002? Even using ulaw and pass-thru, is it possible?
> 
> 2. Since the faxes is coming from PSTN thru the card, I guess asterisk
will
> always stay in the middle right? No way around this.
> 
> 3. Im also trying to receive faxes usign a TE110P card with spandsp,
unicall
> and E1 R2MFC, no luck also, some stuff, garbage and broken faxes. Anybody
> done this sucessfuly?
> 
> Hope anybody can share their thoughts and insight on this.

Using the TDM400 card for any form of fax'ing (or modem use) is well known
to be unreliable and, in most cases, totally unusable. The issue has been
discussed many times over the last two years or so. There are no known
workarounds.

Its my understanding that lots of folks have spandsp working via T1
and/or PRI interfaces. The issues associated with the TDM400 card do
not apply to the T1 cards.




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[Asterisk-Users] Re: [asterisk-dev] Possible Bug in SIP Stack.

[Olle E Johansson]

Chris Modesitt wrote:
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is 
APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server.   When I 
use Asterisk version 10.0.10 everything works perfectly, however when I 
use 1.2.4 I lose the ability to receive calls from the PSTN.  All I get 
is the following error in my SIP Proxies error logs:


 

SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, 
don't know where to send responseSIP/2.0 180 Ringing


From: "MODESITT,CHRIS " 
;tag=4fdc9d0e-1e600f94-ed7e623f


To: ;tag=as4fc8aa8a

Call-ID: [EMAIL PROTECTED]

CSeq: 5466974 INVITE

User-Agent: Asterisk PBX

 


I still can make outbound calls with no-problems, any ideas?

 
Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 
1.2.4 so we can compare them and see what happened?


Thanks
/Olle
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RE: [Asterisk-Users] What business IP phone to use

[Michael Graves]

On Wed, 22 Feb 2006 18:02:27 -0800, mustardman29 wrote:
>Just the person I have been looking for.  If you don't mind, would it be
>possible to get your opinion on feature for feature comparisons between the
>501 and 480i CT(not including cordless phone).
>
>Things like programmable buttons, display, dialing button quality, and most
>importantly, handset and speakerphone quality.
>
>Any info would be greatly appreciated.

I used the IP600 for about a year on my desk, and several IP500s
elsewhere around the place. It's a home office but I work from home
full time so it's a real working office environment.

I found that the physical quality of the Polycom phones was absolutely
top notch. They're a joy to use. Completely professional and very
reliable. But they're not perfect. They're a little harder to
provision. They're very configurable but that also adds to the
complexity. I had mine TFTP loading firmware and a common speed dial
directory from an XML file on my Astlinux server. The phones take a
fair amount of time to boot and force a reboot when you change many of
their settings. You can spend an afternoon repeatedly rebooting the
phone as you manually work out its initial configuration. Of course
Polycom doesn't support Asterisk, but others seem to fill this void
well enough.

The IP600 and IP500 are very similar but the differences are
considerable. The IP600 supports 6 line buttons and has a much better
LCD. Higher resolution, but still not backlit. Once you've used the 600
it'll be hard to go
back to the 500 just because the display is not as nice. The IP500
provides only 3 line buttons. Both phones support multiple
registrations.

The Aastra 480 is the only thing that I've seen that comes close to the
Polycom's. Physically it's just about as solid. Not quite as hefty in
the hand, but very nice. The LCD display is backlit. This is a major
advantage if you ever work in dim lighting. All other
manufacturers...LISTEN UP...this is a really big deal! I can't believe
how long its taken for someone to realise this fact.

Aastra configuration was a LOT easier both manually on the phone and
remotely. The on-phone menus are very easy to navigate and I almost
didn't bother setting up the central provisioning. With only a few
phones I could get by without it. Firmware and configs can be loaded
via tftp, ftp or http.

The on-phone directory and call logs are comparable on all three the I
have used. Actually, I prefer the way SNOM phones handle this as they
require fewer button presses. The Aastra phone makes it especially easy
to delete an entire call log with only a couple of button presses.

The 480 supports up to 9 lines with any 4 active at on time, or so I'm
told. I have mine registered for four lines so that incomming PSTN,
FWD, Gizmo and Skype calls each ring a different line. The latest
firmware supposedly support BLF indications but I've not used this.
It's really easy to assign speed dials to the six programmable keys on
the LCD. In fact, almost all of the buttons can be reassigned to new
functions. Also you can write XML applications that put the LCD to work
as an interactive menu.

Mostly I live and die by speakerphone quality. I think that the
Polycom's have a little edge on the Aastra phone, but not by much. If I
need to rework my entire system I'll probably migrate to all Aastra
phones.

Audio quality using the handset is excellent on all of them. Even on
the cordless handset with the 480i CT.

They all support POE...which I use to keep the phone system up during
power failures. I had to buy the injectors separately for the Aastra &
IP600 phones. The IP500s came with injector cables. 

The big dissappointment in my SIP phone testing was the Zultys 4x5. It
just feels cheap and many functions are too counterintuitive. I really
like the idea of the local FXO but they were never able to tell me how
to get the FXO port forwarded to the PBX for VM. Zultys provides no end
user support except through dealers and the dealers I dealt with didn't
know much about the specifics of the Zultys firmware.

Also, I'm curious about the newest SNOM phones. Some time ago I used a
SNOM 200 and like the way the web based I/F was integrated into the use
of the phone beyond simply configuration. You could access the speed
dials and place a call from the web I/F. You could also dial the phone
from a link or shortcut to a url pointed at the phone. That's a fair
substitute for desktop TAPI. If they've taken this any further it could
be very good.

I've not tried any of the lesser phones like Grandstream or Linksys.
Life's too short to use a cheap phoneat least if your budget
permits better.

Michael Graves

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245





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Re: [Asterisk-Users] GPS-enabled cell phone/PDA

[Steve Kennedy]

On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote:

> Its my understanding the cell phone coordinates are sent to the cell phone
> provider and their equipment reads (and holds) that data. Its not part
> of any data available to you in any form unless you talk to the cell
> provider and convience them you have a valid need. Highly unlikely in
> the US anyway. Even if you could convience them to provide it, they
> would likely demaand some sort of out-of-band data transmission facility.

GSM networks have the Cell ID available to the phone, however that's not
much use without the location of the cellsite.

There are now location based services, whereby you can query the network
and they'll give out an approximate location (most cells are sectored
[6 sectors per cell) which gives a direction, the cell also knows what
power the phone is transmitting with, and the power it's received so can
make a good approximation of where the phone is (within 60 degrees
angle). However it's likely a phone will be picked up by several cells,
so the network can triangulate and make a better aproximation.

Making the information available to end-users is problematic due to
privacy issues, unless the user explicitly agrees to give the info away.

With GPS units, the info is stored in the phone and can send it out
using SMS or other means.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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[Asterisk-Users] Re: [asterisk-dev] Possible Bug in SIP Stack.

[Olle E Johansson]

Chris Modesitt wrote:
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is 
APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server.   When I 
use Asterisk version 10.0.10 everything works perfectly, however when I 
use 1.2.4 I lose the ability to receive calls from the PSTN.  All I get 
is the following error in my SIP Proxies error logs:


 

SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, 
don't know where to send responseSIP/2.0 180 Ringing


From: "MODESITT,CHRIS " 
;tag=4fdc9d0e-1e600f94-ed7e623f


To: ;tag=as4fc8aa8a

Call-ID: [EMAIL PROTECTED]

CSeq: 5466974 INVITE

User-Agent: Asterisk PBX

 


I still can make outbound calls with no-problems, any ideas?

 
Can you get SIP debug logs from a call setup with Asterisk 1.0.10 and 
1.2.4 so we can compare them and see what happened?


Thanks
/Olle
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[Asterisk-Users] Re: Explain Yellow Alarm in a Legacy Integration

[Geoff Manning]

On 2/23/06, Geoff Manning <[EMAIL PROTECTED]> wrote:
How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration?We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P.Twice today (first time in over a month) we received a Yellow Alarm on the TE110P.  I have been able to clear it easily by restarting zaptel.
Thanks in advance!

So we had another Yellow Alarm last night and I have retrieved the logs from the Mitel. It had a Red Alarm.Here seems to be the order of events:Mitel PBX: ³2006-FEB-24 02:44:54  T1/BRI card at 02 06 00 00    ³
 ³  has exceeded the service loss frame threshold ³ ³2006-FEB-24 02:44:54  Tot alarm went from No Alarm to MAJOR ³ ³  Alarm level change due to  Bay 02 trunks  ³
 ³2006-FEB-24 02:44:54  T1/BRI card at 02 06 00 00    ³ ³  removed from service & transmitting yellow alarm  ³Asterisk:Feb 24 02:45:43 WARNING[24210] chan_zap.c: Detected alarm on channel 1: Yellow Alarm 
Mitel PBX (This is when we manually reset the card on the Asterisk to clear the alarm): ³2006-FEB-24 05:25:45  T1/BRI card at 02 06 00 00    ³ ³  is in red alarm condition due to loss of sync ³
 ³2006-FEB-24 05:26:08  T1/BRI card at 02 06 00 00    ³ ³  alarm condition is now cleared    ³ ³2006-FEB-24 05:26:08  Tot alarm went from MAJOR    to No Alarm  ³
 ³  Alarm level change due to  Bay 02 trunks  Asterisk:Feb 24 05:26:54 NOTICE[24210] chan_zap.c: Alarm cleared on channel 1So it seems the Mitel is reaching a loss threshold and setting yellow alarm. Asterisk is in turn detecting the yellow alarm. I guess it's a problem with the Mitel then. We've had problems with it in the past but they cleared up and we hadn't had an issue in months. Nothing has changed at either end but we've been hit with issues for the last 3 days.
Here is what I have found about the alarms:Red 
  Alarm
  This is 
  a local equipment alarm. It indicates that the incoming signal has been 
  corrupted for a number of seconds. The red alarm shows up visually on 
  the equipment that detects the failure. This equipment will then begin 
  sending a yellow alarm as its outbound signal. 
  
  Yellow 
  Alarm
  The yellow 
  alarm alerts the network that a failure has been detected. The yellow 
  alarm pattern has a number of different definitions. The most common 
  D4 definition is to set 1 bit of every channel to a ZERO. 

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Re: [Asterisk-Users] Problem with T1 installation

[Doug Lytle]

Nitin Joshi wrote:

Hi All,
 
I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its 
connected directly to the PSTN. But I am unable to make outbound calls 
on the zap channels. The light on the card is green. Asterisk CLI 
shows all 24 channels when I give the command 'zap show channels'. I 
also noticed that Asterisk CLI shows an incoming call every few 
seconds on the 24th channel. This must be some kind of a timing 
signal. This is he first time I am configuring a T1 so I must have 
done something wrong I guess.


T1s require a D (Data) channel, unless connecting to a channel bank, It 
should be 23 voice 1 data.  Also, I would strongly suggest moving to 1.2.4


Doug

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[Asterisk-Users] Possible Bug in SIP Stack.

[Chris Modesitt]

I currently use asterisk version 1.0.10 with AMP 1.0.010,
our setup is APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk
server.   When I use Asterisk version 10.0.10 everything works
perfectly, however when I use 1.2.4 I lose the ability to receive calls from the
PSTN.  All I get is the following error in my SIP Proxies error logs:

 

SIPSession::proxyResponseImmediately(): Failed to retrieve
next Via, don't know where to send responseSIP/2.0 180 Ringing

From: "MODESITT,CHRIS " ;tag=4fdc9d0e-1e600f94-ed7e623f

To: ;tag=as4fc8aa8a

Call-ID: [EMAIL PROTECTED]

CSeq: 5466974 INVITE

User-Agent: Asterisk PBX

 

I still can make outbound calls with no-problems, any ideas?

 

Thanks

 

Chris






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Re: [Asterisk-Users] S100U and TigerJet

[Jerry Glomph Black]

udev drove me absolutely bat-shit in this regard; udev is a horror in many 
respects.   Here's how I solved the problem, reliably:


I run this script at boot-time:

#!/bin/bash
mkdir -p /dev/zap
rm -f /dev/zap/ctl
rm -f /dev/zap/channel
rm -f /dev/zap/pseudo
rm -f /dev/zap/timer
rm -f /dev/zap/253
rm -f /dev/zap/252
rm -f /dev/zap/251
rm -f /dev/zap/250
mknod /dev/zap/ctl c 196 0
mknod /dev/zap/timer c 196 253
mknod /dev/zap/channel c 196 254
mknod /dev/zap/pseudo c 196 255
N=1; \
while [ $N -lt 250 ]; do \
rm -f /dev/zap/$N; \
mknod /dev/zap/$N c 196 $N; \
N=$[$N+1]; \
done

Have had zero problems with this.



On Fri, 24 Feb 2006, [EMAIL PROTECTED] wrote:


Hi all, this is another post about this problem.
I installed from scratch a new Suse Linux  10.0, with latest stable
asterisk.
Moreover I add the lines to  /etc/udev/rules.d/50-udev.rules, in order to
let the driver create the /dev/zap...

When I plug into usb port my TigerJet adapter, I see on /var/log/messages

Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using
uhci_hcd and address 2
Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver
snd-usb-audio
Feb 24 14:55:03 srvlnx05 kernel: zaptel: module not supported by Novell,
setting U taint flag.
Feb 24 14:55:03 srvlnx05 kernel: Zapata Telephony Interface Registered on
major 196
Feb 24 14:55:03 srvlnx05 kernel: wcusb: module not supported by Novell,
setting U taint flag.
Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver wcusb
Feb 24 14:55:03 srvlnx05 kernel: Wildcard USB FXS Interface driver
registered

while lsusb shows
Bus 001 Device 002: ID 06e6:831c Tiger Jet Network, Inc.
Bus 001 Device 001: ID :

under /dev, I see "borning" /zap and children
srvlnx05:/etc # dir /dev/zap/

drwxr-xr-x   2 root root  120 Feb 24 14:55 .
drwxr-xr-x  14 root root15720 Feb 24 14:55 ..
crw-rw   1 asterisk asterisk 196, 254 Feb 24 14:55 channel
crw-rw   1 asterisk asterisk 196,   0 Feb 24 14:55 ctl
crw-rw   1 asterisk asterisk 196, 255 Feb 24 14:55 pseudo
crw-rw   1 asterisk asterisk 196, 253 Feb 24 14:55 timer

but NO channel 01 al all.
I would like to know if anybody
1) ever succeded in having this configuration up and running.
2) ever succeded in having this configuration up and running with a *TRUE*
S100U adapter from Digium.
3) If 2 is true *WHERE* it could be possible to buy this true adapter: on
digium shop I was not able to find it.

My opinion is that it could be an issue related to the operating system: I
think I should do something similar to what I did on
/etc/udev/rules.d/50-udev.rules in order to allow the creation of
usb-related devices under /dev/zap. Unfortunately
I don't know anything about Linux kernel enumeration process. Also, does
exist any debugging tool for wcusb ?
Wcusb is up and running, is the only in the system ( I removed the wcusb.ko
natively present under the /extra directory)
lsmod | grep wcu shows:

srvlnx05:~ # lsmod | grep wcu
wcusb  19104  0
zaptel187268  1 wcusb
usbcore   112512  5 wcusb,snd_usb_audio,snd_usb_lib,uhci_hcd


thank's all for attention.
Andrea


Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

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Re: [Asterisk-Users] fax receive using TDM400P

[Thomas Artner]

Hi!

I am using tdm400 cards for receiving faxes. It worked quite out of the box. I 
installed spandsp for the rxfax application only.

I use it as phone/fax switch:
All incoming calls are answered automatically to listen whether its a fax or 
not. If it is a fax, the call is forwarded to the buil-in fax extension, 
otherwise the analog phones (all on tdm400) rings.

It works without problems. Its for a small company (about a few faxes per 
hour)


Tom




Am Freitag, 24. Februar 2006 07:10 schrieb Anton Krall:
> Guys.
>
> Ive been testing how to receive faxes using TDM400P cards and so far, after
> playing with gains, echocancell and echotraining on zapata.conf.. Ive ha
> dno luck, faxes come in as garbage or broken or with blank lines.
>
> Anybody has successfully done this? Any tips.. Also I have some ideas:
>
> 1. Is it really possible to get fxes on a fax machine using ATAs like the
> sipura 2002? Even using ulaw and pass-thru, is it possible?
>
> 2. Since the faxes is coming from PSTN thru the card, I guess asterisk will
> always stay in the middle right? No way around this.
>
> 3. Im also trying to receive faxes usign a TE110P card with spandsp,
> unicall and E1 R2MFC, no luck also, some stuff, garbage and broken faxes.
> Anybody done this sucessfuly?
>
> Hope anybody can share their thoughts and insight on this.
>
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-- 
Thomas Artner
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Re: [Asterisk-Users] OT: VoIP over bonded link

[Andrew Kohlsmith]

On Thursday 23 February 2006 13:57, Bob Goddard wrote:
> It's stupid. Don't ever connect 2 different building with copper.
> Just wait until you get some kind of lightening hit or electrical
> fault, but make sure you are no where near it. Use fibre.

That's a great rule of thumb, but the reality isn't quite so black and white.

A direct lightning strike is not going to draw *any* significant current 
through the ethernet cable, as the moment you try to pull significant 
current, those cables will either open up or vaporize due to IR losses in 
such small gage wire.  You'll have far more current draw through the (I'm 
assuming) metal conduit, which is already grounded.

Yes, you may introduce grounding loops and these will cause other (sometimes 
significant) issues but they have all been solved before.  The best solution 
is to simply take a pair of media converters with a fiber patch cable between 
them, space them out adequately and hope for the best.  You're already going 
to have a conduction path through the power supplies of the media converters 
but with an isolation transformer and appropriate surge arrestors it's about 
as best as you are going to be able to do.

Electrical faults are *easily* dealt with with appropriate fusing, surge 
arrestors, isolation and plain old common sense.

I work in the power electronics industry; we regularly deal with lightning 
strikes (both direct and "close call" style) and while there is very little 
to protect you from a direct strike (we use station-class arrestors) there is 
a LOT you can do to minimize grounding or loop problems when wiring between 
buildings.  Sometimes fiber just doesn't cut it, so no, it's "not just 
stupid."

-A.
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RE: [Asterisk-Users] What business IP phone to use

[Conrad Wood]

On Sat, 2006-02-25 at 00:21 +1100, David Ankers wrote:
> Aha, micro seconds in networking terms is normally written usecs or us
> (actually it's the greek letter mu as in ulaw) rather than ms which are
> milliseconds seconds - what had me puzzled was that it was stated that this
> could harm the voice path!
> 
> > The difference can also cause unnecessary delays and therefor echo in the
> > path. For example, procurve switches typically have 13ms switching time,
> > the high-end netgears about 21ms. As soon as you stack a couple of
> > switches you are talking 26ms vs 42ms extra delay in the path!
> 
> There is then only 8 usecs between the two switches, how on earth would this
> make any difference to the voice path at all? Let alone induce any echo... 
> 
> Obviously the originally poster didn't understand the difference. And based
> on this, he's probably advising people not to use Netgear switches for
> voice, oh dear.  
> 
> 

Agree , previous statement was incorrect and I should probably not post
late at night ;-)
A few microseconds delay in the path obviously doesn't cause extra echo.
Thank you for pointing that out.

== Conrad



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[Asterisk-Users] S100U and TigerJet

[asterisk]

Hi all, this is another post about this problem.
I installed from scratch a new Suse Linux  10.0, with latest stable
asterisk.
Moreover I add the lines to  /etc/udev/rules.d/50-udev.rules, in order to
let the driver create the /dev/zap...

When I plug into usb port my TigerJet adapter, I see on /var/log/messages

Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using
uhci_hcd and address 2
Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver
snd-usb-audio
Feb 24 14:55:03 srvlnx05 kernel: zaptel: module not supported by Novell,
setting U taint flag.
Feb 24 14:55:03 srvlnx05 kernel: Zapata Telephony Interface Registered on
major 196
Feb 24 14:55:03 srvlnx05 kernel: wcusb: module not supported by Novell,
setting U taint flag.
Feb 24 14:55:03 srvlnx05 kernel: usbcore: registered new driver wcusb
Feb 24 14:55:03 srvlnx05 kernel: Wildcard USB FXS Interface driver
registered

while lsusb shows
Bus 001 Device 002: ID 06e6:831c Tiger Jet Network, Inc.
Bus 001 Device 001: ID :

under /dev, I see "borning" /zap and children
srvlnx05:/etc # dir /dev/zap/

drwxr-xr-x   2 root root  120 Feb 24 14:55 .
drwxr-xr-x  14 root root15720 Feb 24 14:55 ..
crw-rw   1 asterisk asterisk 196, 254 Feb 24 14:55 channel
crw-rw   1 asterisk asterisk 196,   0 Feb 24 14:55 ctl
crw-rw   1 asterisk asterisk 196, 255 Feb 24 14:55 pseudo
crw-rw   1 asterisk asterisk 196, 253 Feb 24 14:55 timer

but NO channel 01 al all.
I would like to know if anybody
1) ever succeded in having this configuration up and running.
2) ever succeded in having this configuration up and running with a *TRUE*
S100U adapter from Digium.
3) If 2 is true *WHERE* it could be possible to buy this true adapter: on
digium shop I was not able to find it.

My opinion is that it could be an issue related to the operating system: I
think I should do something similar to what I did on
 /etc/udev/rules.d/50-udev.rules in order to allow the creation of
usb-related devices under /dev/zap. Unfortunately
I don't know anything about Linux kernel enumeration process. Also, does
exist any debugging tool for wcusb ?
Wcusb is up and running, is the only in the system ( I removed the wcusb.ko
natively present under the /extra directory)
lsmod | grep wcu shows:

srvlnx05:~ # lsmod | grep wcu
wcusb  19104  0
zaptel187268  1 wcusb
usbcore   112512  5 wcusb,snd_usb_audio,snd_usb_lib,uhci_hcd


thank's all for attention.
Andrea


Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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Re: [Asterisk-Users] Analyzer for Milliwatt

[Paul]

Andrew Kohlsmith wrote:

>On Friday 24 February 2006 07:56, Paul wrote:
>  
>
>>Maybe the first approach should be to setup a test extension for
>>recording the tone. The idea is to get best resolution possible in real
>>time. Then process it as much as needed to get the info you want. Such
>>an approach would give you more flexibility. For example, you could
>>automatically place periodic test calls to various servers and have the
>>recordings then forwarded to one server for analysis. That would
>>minimize the impact on production asterisk servers.
>>
>>
>
>What is being discussed here is basically what I was planning on doing for an 
>automatic VOIP quality check.  Using miliwatt and analyzing it for 
>pop/jitter/etc as well as sending other known waveforms and comparing what 
>was received to what was expected and coming up with some "quality" number 
>which would be fed back to the dialplan to adjust the least-cost routing 
>paths.  Essentially come up with a "least cost but still good quality" 
>routing.  :-)
>
>I've done absolutely nothing other than a little research and a lot of 
>thinking about how to do it though.  I did some research on digital click/pop 
>removal for records as a way to detect poor quality, and then also some 
>monkeying around with coppice's excellent DSP routines in spandsp.
>  
>
I guess the best information would be obtained by recording in the codec
format. That means being sure to prevent transcoding. I'm not sure if
that can be done with simple dialplan programming.

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Re: [Asterisk-Users] What business IP phone to use

[Paul]

I have seen some very expensive switches fail. Nice thing about lower
cost devices is that you can afford to have spares. If you stick to a
standard way of labeling and connecting wires you can use good open
source monitoring software to detect switch failure. If you allow people
to randomly connect to a bank of switches it is not so easy to quickly
find and remedy such problems.

The more expensive switches are good if you are going to take advantage
of the features they offer. I have recently seen situations like
employees installing things like camera and itunes software that caused
local network problems. Managed switches allowed immediate remote
disconnection of the workstations. At this customer site the fancy
switches are used for all workstations and some 3rd party
servers(security video system is a good example). However, the
customer-owned servers I installed are plugged into a $40 switch. Those
servers are properly managed so there is no need for the features found
in the more expensive switches.

David Ankers wrote:

>Aha, micro seconds in networking terms is normally written usecs or us
>(actually it's the greek letter mu as in ulaw) rather than ms which are
>milliseconds seconds - what had me puzzled was that it was stated that this
>could harm the voice path!
>
>  
>
>>The difference can also cause unnecessary delays and therefor echo in the
>>path. For example, procurve switches typically have 13ms switching time,
>>the high-end netgears about 21ms. As soon as you stack a couple of
>>switches you are talking 26ms vs 42ms extra delay in the path!
>>
>>
>
>There is then only 8 usecs between the two switches, how on earth would this
>make any difference to the voice path at all? Let alone induce any echo... 
>
>Obviously the originally poster didn't understand the difference. And based
>on this, he's probably advising people not to use Netgear switches for
>voice, oh dear.  
>
>
>
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Watkins,
>Bradley
>Sent: Friday, 24 February 2006 10:08 PM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: RE: [Asterisk-Users] What business IP phone to use
>
>It must be microseconds that is being quoted, as even the 2626 that you
>mention lists a less than 13.3 microsecond latency.
>
>- Brad
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of David Ankers
>Sent: Thursday, February 23, 2006 6:54 PM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: RE: [Asterisk-Users] What business IP phone to use
>
>
>Are you sure those switch figures are right? 16ms delay in the switch path
>sounds a bit long. Cisco's mid-range switches like the 2950 have switching
>times measured in micro seconds. Then again a 2626 procurve is only around
>$700.
>
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Conrad Wood
>Sent: Friday, 24 February 2006 7:50 AM
>To: asterisk-users@lists.digium.com
>Subject: RE: [Asterisk-Users] What business IP phone to use
>
>
>  
>
>>Simple formula:
>>
>>1. Total Revenue
>>2. % of revenue derived from phone usage
>>3. =Cost of downtime by using SoHo or consumer gear.
>>
>>It's not a question of if a SoHo or low cost device will screw up, it 
>>is a question of when. This is 23 years of experience talking.
>>
>>Where I work, the value of #3 above is $16 Cdn a *second*. We are 
>>below
>>
>>
>500
>  
>
>>employees, so we fall into the SMB segment. Sometimes I'm appalled by 
>>statements that a $700 switch or a $400 phone isn't worth it. Huh?? 
>>Maybe
>>
>>
>in
>
>Absolutely right! for something as critical as switches & cabling I always
>recommend to spend real money. Don't ever try to save money any equipment
>that is required to operate the business. (Had very good experience with HP
>procurves over the last 10 years or so). There is no point buying netgear or
>other low-cost switches for a business ever. The cost saving of being able
>to pin-point a cabling/NIC/bandwidth problem down to the port on the switch
>easily and quickly is wonderful. Combined with SNMP and all the other
>goodies good switches come with, our clients save a lot of money by paying
>me less for my time ( d'oh ;-) ). The difference can also cause unnecessary
>delays and therefor echo in the path. For example, procurve switches
>typically have 13ms switching time, the high-end netgears about 21ms. As
>soon as you stack a couple of switches you are talking 26ms vs 42ms extra
>delay in the path!
>
>I see no reason however to spend $400 on a single phone though, because if a
>single phone breaks, it's not going to bring your business to a standstill,
>is it? (I guess unless you only have one in the first place ;-) )
>
>conrad
>  
>

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Re: [Asterisk-Users] GPS-enabled cell phone/PDA

[Juergen K. Zick]

Hi There,

this is very much dependent from your provider, your PDA/cell phone and the 
network. For GSM networks in Europe e.g. the providers have different types 
of information available through the CB channels of their base stations.
This data can always be read and stored in your SMARTPHONE/PDA and when 
that has GPS data, then this data as well ...
One nice examples are celltrack or gsmmon9210 for SYMBIAN based phones. 
What you do on the phone with the data is your business ;-) ..
There are web-based databases available which show the exact location of 
the next station you're connected to. If you have GPS locally, than you 
have not to rely on thie cell data.
Cell data inside cities can give your location as exact as to 100m, in 
rural areas it can be up to 5 km I suppose.


Of course you can send the received data via SMS to other systems or with 
GPRS or WLAN access more or less online to Internet based services.


With TDMA or IDEN phone systems which are used outside of Europe I have no 
experiences at all, sorry ...


-- Jürgen



> I would like to capture the lat/lon coordinates from a GPS-enabled cell
> phone or PDA.  Is this possible?  Must I subscribe to this information
> from the cellphone network provider, or can I capture it without charge?
>
> What devices will broadcast the coordinates?  Is there a device that
> will broadcast its position inband that can be captured by Asterisk?
> Can an SMS message include coordinates?
>
> The subject will willingly carry the device and will be aware that his
> location is being monitored, so privacy rights are not an issue.  The
> subject will make periodic calls to the Asterisk server in order to
> record his movements.
>
> Does anyone have experience in this area?

Its my understanding the cell phone coordinates are sent to the cell phone
provider and their equipment reads (and holds) that data. Its not part
of any data available to you in any form unless you talk to the cell
provider and convience them you have a valid need. Highly unlikely in
the US anyway. Even if you could convience them to provide it, they
would likely demaand some sort of out-of-band data transmission facility.




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Re: [Asterisk-Users] OT: VoIP over bonded link

[Bob Goddard]

On Thursday 23 Feb 2006 20:34, Colin Anderson wrote:
> >It's stupid. Don't ever connect 2 different building with copper.
> >Just wait until you get some kind of lightening hit or electrical
> >fault, but make sure you are no where near it. Use fibre.
>
> Thanks for the reply. Unfortunately, the conduit for the provisioning of
> the new building is unsuitable for fibre (too many sharp bends) and we
> can't core out the concrete and put in a new conduit because of obstacles
> in the way that make laying new conduit impractical, so we are stuck with
> (existing) copper. We already have copper-to-copper connections of
> different types (electrical, security etc) between the buildings so a
> lightning strike is going to hose us no matter what.

In that case, put opto-couplers in place to protect both ends.
Fibre/ethernet transceivers at both ends with a short run of
fibre will protect both ends. Lightening strikes are only one
problem, look to see what happens when one building attempts
to ground itself through the copper cable to the other side.
I would also question the legality of connecting both building
with what I assume is mains electricity.


B

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[Asterisk-Users] Problem with T1 installation

[Nitin Joshi]

Hi All,
 
I have installed a Digium TE110P card on an 
Asterisk 1.2.1 system. Its connected directly to the PSTN. But I am unable to 
make outbound calls on the zap channels. The light on the card is green. 
Asterisk CLI shows all 24 channels when I give the command 'zap show channels'. 
I also noticed that Asterisk CLI shows an incoming call every few seconds on the 
24th channel. This must be some kind of a timing signal. This is he first time I 
am configuring a T1 so I must have done something wrong I guess.
 
These are the commands I used to load the zap 
module:
 
modprobe zaptel
modprobe wcte11xp
ztcfg -vvv
 
---
 
my zaptel.conf is as 
follows:
 
span=1,1,0,esf,b8zse&m=1-24loadzone = 
usdefaultzone=us
--
 
the zapata.conf is as 
follows:
 
[trunkgroups][channels]
 
group=1language=ensignalling=em_wusecallerid=yescallerid=asreceivedcontext=defaultechocancel=64echocancelwhenbridged=yesrxgain=1.0txgain=1.0channel 
=> 
1-2group=2language=ensignalling=em_wusecallerid=yescallerid=asreceivedcontext=defaultechocancel=64echocancelwhenbridged=yesrxgain=1.0txgain=1.0channel 
=> 3-24
--
 
In extensions.conf  i have 
specified the following line:
 
[default]
exten => 
_ZX,1,Dial(zap/g1/${EXTEN},15,tr)
 
--

When I try to dial using the T1 line I 
get the following error :
 
Feb 24 06:56:53 NOTICE[5724]: 
app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 
- Unknown)  == Everyone is busy/congested at this time 
(1:0/0/1)  == Auto fallthrough, channel 'SIP/7180-a103' status is 
'CHANUNAVAIL'
 
Any ideas guys?
 
Thanks and regards,
Nitin 
Joshi.
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Re: [Asterisk-Users] fax receive using TDM400P

[Rich Adamson]

> Ive been testing how to receive faxes using TDM400P cards and so far, after
> playing with gains, echocancell and echotraining on zapata.conf.. Ive ha dno
> luck, faxes come in as garbage or broken or with blank lines.
> 
> Anybody has successfully done this? Any tips.. Also I have some ideas:
> 
> 1. Is it really possible to get fxes on a fax machine using ATAs like the
> sipura 2002? Even using ulaw and pass-thru, is it possible?
> 
> 2. Since the faxes is coming from PSTN thru the card, I guess asterisk will
> always stay in the middle right? No way around this.
> 
> 3. Im also trying to receive faxes usign a TE110P card with spandsp, unicall
> and E1 R2MFC, no luck also, some stuff, garbage and broken faxes. Anybody
> done this sucessfuly?
> 
> Hope anybody can share their thoughts and insight on this.

Using the TDM400 card for any form of fax'ing (or modem use) is well known
to be unreliable and, in most cases, totally unusable. The issue has been
discussed many times over the last two years or so. There are no known
workarounds.

Its my understanding that lots of folks have spandsp working via T1
and/or PRI interfaces. The issues associated with the TDM400 card do
not apply to the T1 cards.


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Re: [Asterisk-Users] Analyzer for Milliwatt

[Andrew Kohlsmith]

On Friday 24 February 2006 07:56, Paul wrote:
> Maybe the first approach should be to setup a test extension for
> recording the tone. The idea is to get best resolution possible in real
> time. Then process it as much as needed to get the info you want. Such
> an approach would give you more flexibility. For example, you could
> automatically place periodic test calls to various servers and have the
> recordings then forwarded to one server for analysis. That would
> minimize the impact on production asterisk servers.

What is being discussed here is basically what I was planning on doing for an 
automatic VOIP quality check.  Using miliwatt and analyzing it for 
pop/jitter/etc as well as sending other known waveforms and comparing what 
was received to what was expected and coming up with some "quality" number 
which would be fed back to the dialplan to adjust the least-cost routing 
paths.  Essentially come up with a "least cost but still good quality" 
routing.  :-)

I've done absolutely nothing other than a little research and a lot of 
thinking about how to do it though.  I did some research on digital click/pop 
removal for records as a way to detect poor quality, and then also some 
monkeying around with coppice's excellent DSP routines in spandsp.

-A.
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Re: [Asterisk-Users] Polycom IP 601 Buddy Watch doesn't work after Asterisk reload

[BJ Weschke]

On 2/24/06, Marco Maiolini <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I configured Buddy Watch function on my Polycom IP 601. It works well, until 
> I make a reload of Asterisk. After reload, if I give the "show hints" command 
> in Asterisk's CLI, it says that there are no watcher for the extensions that 
> I configured.
>
> Before the reload in the CLI appears:
>
> -= Registered Asterisk Dial Plan Hints =-
>
> 3002 : SIP/3002 State:Idle
>   Watchers 1
>
> 3006 : SIP/3006 State:Idle
>Watchers 1
>
> 3003 : SIP/3003 State:Unavailable 
> Watchers 1
>
> 3001 : SIP/3001 State:Idle
> Watchers 1
>
> 3000 : SIP/3000 State:Idle
>  Watchers 1
>
>
> After the reload in the CLI appears:
>
> -= Registered Asterisk Dial Plan Hints =-
>
> 3002 : SIP/3002 State:Idle   
> Watchers 0
>
> 3006 : SIP/3006 State:Idle   
> Watchers 0
>
> 3003 : SIP/3003 State:Unavailable 
> Watchers 0
>
> 3001 : SIP/3001 State:Idle
> Watchers 0
>
> 3000 : SIP/3000 State:Idle
> Watchers 0
>
>
> Asterisk sends a SIP NOTIFY message in which the field Subscription-State is: 
> "terminated; reason=probation" and the phone responds with a ACK.
>
> I have then to restart the phone to reactivate the Buddy Watch function.
>
> Is there anybody that can help me with this problem? Is it a problem of the 
> PBX  or a problem of the phone?
>

 It is a phone issue as the phone is supposed to try and resubscribe
after 60 seconds which is an attribute in that message, but it
doesn't. However, bug 6047 in Mantis has some code to try and provide
a workaround for this issue. Testing would be greatly appreciated.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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