RE: [asterisk-users] fxotune unable to set impedence

[Yuan LIU]

From: "Yuan LIU" <[EMAIL PROTECTED]>

How can I fix this?  Or does fxotune only tune TDM400?  (My TDM400P shows a 
mere 1.2% echo.)  Could it do "authentic" X100P?


I just didn't want to accept fxotune.c's claim about working only with TDM.  
Several other users indicated that they were not able to tune X100P.  
There's also a README.debian note that specifically indicated exclusion of 
X100P.


Yuan Liu


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RE: [asterisk-users] Diva Server V-BRI-2 and internal numbers

[Armin Schindler]

On Thu, 14 Dec 2006, Gregory Duchatelet wrote:
> > It looks like 107 is busy ;-)
> > Please increase verbosity, like
> >   set verbose 5
> >   capi debug
> > to see what is happening.
> 
> Hi Armin,
> 
> Verbose was at 30 :)
> 107 is not busy since i can call it from 102, which is another internal
> phone. All internal phones are busy for Asterisk...
> 
> Here is the log with verbose at 100 and capi debug enabled :
 
I should have seen that in your first email: your dialstring is
wrong. You set 'b' as callerid for that call. I assume you want to have
early-B3, so you should do
  Dial(CAPI/ISDN1/107/b)

Armin

 
> -- Executing Dial("SIP/Greg-081f5a10", "CAPI/ISDN1/b:107||rtT") in new
> stack
>> data = ISDN1/b:107
>> parsed dialstring: 'ISDN1' 'b' '107' ''
>> capi request for interface 'ISDN1'
>> parsed dialstring: 'ISDN1' 'b' '107' ''
>   == ISDN1: Call CAPI/ISDN1/107-1e   (pres=0x00, ton=0x00)
> CONNECT_REQ ID=001 #0x1002 LEN=0047
>   Controller/PLCI/NCCI= 0x1
>   CIPValue= 0x1
>   CalledPartyNumber   = <80>107
>   CallingPartyNumber  = <00 80>b
>   CalledPartySubaddress   = default
>   CallingPartySubaddress  = default
>   BProtocol
>B1protocol = 0x1
>B2protocol = 0x1
>B3protocol = 0x0
>B1configuration= default
>B2configuration= default
>B3configuration= default
>GlobalConfiguration= default
>   BC  = default
>   LLC = default
>   HLC = default
>   AdditionalInfo
>BChannelinformation= <00 00>
>Keypadfacility = default
>Useruserdata   = default
>Facilitydataarray  = default
>SendingComplete= default
> 
> -- Called ISDN1/b:107
>> CAPI devicestate requested for ISDN1/107
>> CAPI devicestate requested for ISDN1/107
> CONNECT_CONF ID=001 #0x1002 LEN=0014
>   Controller/PLCI/NCCI= 0x101
>   Info= 0x0
> 
> -- ISDN1: received CONNECT_CONF PLCI = 0x101
> INFO_IND ID=001 #0x11a8 LEN=0016
>   Controller/PLCI/NCCI= 0x101
>   InfoNumber  = 0x18
>   InfoElement = <8a>
> 
> INFO_RESP ID=001 #0x11a8 LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- ISDN1: info element CHANNEL IDENTIFICATION 8a
> INFO_IND ID=001 #0x11a9 LEN=0015
>   Controller/PLCI/NCCI= 0x101
>   InfoNumber  = 0x800d
>   InfoElement = default
> 
> INFO_RESP ID=001 #0x11a9 LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- ISDN1: info element SETUP ACK
> INFO_IND ID=001 #0x11ab LEN=0015
>   Controller/PLCI/NCCI= 0x101
>   InfoNumber  = 0x8002
>   InfoElement = default
> 
> INFO_RESP ID=001 #0x11ab LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- ISDN1: info element CALL PROCEEDING
> -- CAPI/ISDN1/107-1e is proceeding passing it to SIP/Greg-081f5a10
> INFO_IND ID=001 #0x11ad LEN=0037
>   Controller/PLCI/NCCI= 0x101
>   InfoNumber  = 0x1c
>   InfoElement = <91 a1 13 02 02 8f> <02 01 22>0<0a a1
> 05>0<03 02 01 00 82 01 01>
> 
> INFO_RESP ID=001 #0x11ad LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- ISDN1: info element FACILITY
> INFO_IND ID=001 #0x11ae LEN=0017
>   Controller/PLCI/NCCI= 0x101
>   InfoNumber  = 0x8
>   InfoElement = <81 d8>
> 
> INFO_RESP ID=001 #0x11ae LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- ISDN1: info element CAUSE 81 d8
> INFO_IND ID=001 #0x11af LEN=0015
>   Controller/PLCI/NCCI= 0x101
>   InfoNumber  = 0x8045
>   InfoElement = default
> 
> INFO_RESP ID=001 #0x11af LEN=0012
>   Controller/PLCI/NCCI= 0x101
> 
> -- ISDN1: info element DISCONNECT
> -- ISDN1: Disconnect case 1
> -- CAPI/ISDN1/107-1e is busy
>   == ISDN1: CAPI Hangingup
> -- ISDN1: activehangingup (cause=88)
> DISCONNECT_REQ ID=001 #0x1003 LEN=0018
>   Controller/PLCI/NCCI= 0x101
>   AdditionalInfo
>BChannelinformation= default
>Keypadfacility = default
>Useruserdata   = default
>Facilitydataarray  = default
>SendingComplete= default
> 
>   == Everyone is busy/congested at this time (1:1/0/0)
> -- Executing Hangup("SIP/Greg-081f5a10", "") in new stack
>   == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on
> 'SIP/Greg-081f5a10' in macro 'appel_sortant'
>   == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on
> 'SIP/Greg-081f5a10'

[asterisk-users] fxotune unable to set impedence

[Yuan LIU]

My SM56 (Motorola X100P clone) has echo as hight as 38%, according to 
fxotune -d.  But when trying to take action, it fxotune simply says it 
can't.


./fxotune -i3 -
Running with parameters:
   doset=0
   docalibrate=1
   dodump=0
   startdev=1
   stopdev=252
   calibtype=2
   waveformtype=-1
   delaytosilence=0
   silencegoodfor=18
   dialstr=5
   debug=4
Tuning module /dev/zap/1
Unable to set impedance on fd 4
Failure!
--

How can I fix this?  Or does fxotune only tune TDM400?  (My TDM400P shows a 
mere 1.2% echo.)  Could it do "authentic" X100P?


I use zaptel-1.2.10, fxotune.c Rev. 1348.

Yuan Liu


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Re: [Asterisk-Users] Siemens Gigaset SL75

[Alberto Pastore]

Joao Pereira ha scritto:
Do you know if it has 802.1x authentication as it is defined in 
EDUroam (  http://www.eduroam.org/ )   ?
I never found a WiFi phone working with 802.1x  I tested ZyXel 
Prestige 2000 but the sound was bad and it doesnt support 802.1x :(


Thanks
Joao Pereira


Well, I have not tested 802.1x with eap/radius so far,
but wpa/psk works quite well in a multi-AP roamed environment.
The phone definitely has 802.1x authentication,
at least that's what user guide and phone menus report.


So far, this is the best wi-fi phone I ever put my hands on
others phone I tested were (with my humble opinion on them):

Zyxel P2000  -> discrete but poor wi-fi options
Linksys WIP300 -> crappy, slow, freezing,
 battery lasts less than 30' talking
UTStarcom F1000 -> sufficient, bad display/menus, poor audio
UTStarcom F1000G -> ultimate crap, firmware is really bad,
   frequent disconnection from wifi net
Samsung WIP6000 -> good phone, but available in Italy
  only as Telecom Italia rebranded and locked
  plus it's a little too small, cellphone-like
Nokia eSeries (60,70) -> great smartphones but BAD BAD BAD sip stack
and/or wi-fi integration, hoping for a
fixing firmware update from nokia guys


On the contrary SL75 has:

- the right size for a cordless phone
- a comfortable charging cradle
- multilanguage interface (including Italian! yup!)
 & quick reference guide
- standard call functions (call waiting, hold, transfer
 3-way conference)
- a battery that lasts at least 2 hours talking
 (personally tested!)
- soft plastic case which should prolong the phone's life
 with absent-minded employees, in case of dropping
- good roaming features with manual setting of
 RSSI threshold
- the best price/performance ratio

What the phone really lacks is a TFTP-like automated
provisioning capability. Unfortunately, as far as I know
you can configure it only manually via web.


I'll be testing radius eap-md5 in the next few days.

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[asterisk-users] FYI Panasonic Wireless Phone MWI

[Doug Crompton]

Last week I asked about MWI indicators on wireless phones that would work
with Asterisk. I sent a message off to Panasonic asking them about it
because in their ads they specifically stated that the indicator works
with and requires phone company voicemail subscription.

The is the model TG5631.

Specs here...

http://www.amazon.com/Panasonic-KX-TG5631S-GigaRange-Cordless-Answering/dp/B000F4C2CA

and this was the response

Go figure!

Dear MR CROMPTON

Thank you for contacting Panasonic.

The purpose of the message button of your phone system is to inform you
that you have new
messages in your answering system.  That indicator will not work for your
voicemail. We do not have any phone system that has a message alert
indicator that will work both for your voicemail and your answering
machine.

We hope this information is useful to you.

Thank You,
Panasonic Consumer Support


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Re: [asterisk-users] Bandwidth.com on asterisk

[Andrew Joakimsen]

What do they provide you? You normally wouldnt install asterisk "on their
system" unless you are leasing a server from them.

On 12/15/06, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:


Does anybody know how to setup bandwidth.com trunk on asterisk. They
provide bandwidth services to asterisk.org, but don't know how to setup up
asterisk on their system.

--
Zeeshan A Zakaria
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[asterisk-users] Bandwidth.com on asterisk

[Zeeshan Zakaria]

Does anybody know how to setup bandwidth.com trunk on asterisk. They provide
bandwidth services to asterisk.org, but don't know how to setup up asterisk
on their system.

--
Zeeshan A Zakaria
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Re: [asterisk-users] IBM Server / USB Ports

[Leo Ann Boon]

Matt wrote:

So you are saying that the card is on it's own IRQ and is not sharing
anything with anything?  I realize the eth0 and usb are sharing, but
am not too concerned about that.
What's your zttest result and did zttool reported any irq misses? If 
zttest is mostly >99.98%, then the zap device is fine.


See
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting


Leo

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Re: [asterisk-users] On-Hold

[Bartosz Wegrzyn - maillists]

this is ata , simple flash button works,

thx

> If using a VOIP phone there should be a button. If using an ATA the
> instructions should be in the manual of the ATA (you also may be able to
> look in the web interface of the device). I forgot how to do it if you are
> using ZAP. Have a look on the wiki.
>
> - Original Message -
> From: "Bartosz Wegrzyn - maillists" <[EMAIL PROTECTED]>
> To: 
> Sent: Thursday, December 14, 2006 10:30 PM
> Subject: [asterisk-users] On-Hold
>
>
> Hello,
>
> When in conversation, how can I put somebody on hold?
>
> thx
>
>
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RE: [asterisk-users] Broadvoice registration problems

[Bartosz Wegrzyn - maillists]

I am not sure how far I will go with that, but
I did a capture and explained in detail what is the problem,

I hope that somebody there will forward it to high level support maybe,
who knows, it is so hard to get help when something strange happens,

from my experience with broadvoice, everything works always great, but
from time to time they change something on their side, and I have to fix it
myself by some changes in the configuration (if possible)
this time I have no idea , but it does not work,

we will see,

thx

> Any ideas?
> Did anyone experience something like that?
>
> Thx
>
> Yes, unfortunately, all the time.  There answer is if it works with a sip
> softphone client than it's not their problem.  It does work with the
> softphone client.
>
>
>
> -Original Message-
> From: Bartosz Wegrzyn - maillists [mailto:[EMAIL PROTECTED]
> Sent: Thursday, December 14, 2006 3:24 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Broadvoice registration problems
>
> Hello,
>
> I have two broadvoice accounts.
> Lately, very often my broadvoice accounts are in unregistered state.
> When I log into asterisk I see:
>
> voip*CLI> sip show registry
> Host Username Refresh State
> sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent
> sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent
>
> xxx and yyy are my ohone numbers,
>
> when I do tcpdump on my router, I see only packets comming out my router
> to broadvoice but nothing comming back on port 5060
>
> 23:06:41.398952 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
> 147.135.12.128.5060: UDP, length 411
> 23:06:42.306975 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
> 147.135.12.128.5060: UDP, length 411
> 23:06:42.398951 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
> 147.135.12.128.5060: UDP, length 411
> 23:06:44.306932 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
> 147.135.12.128.5060: UDP, length 411
> 23:06:44.398921 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
> 147.135.12.128.5060: UDP, length 411
>
> the only way to get registered is to change the proxy in the host file,
> after some time same story happens, I am in register sent state and
> nothing is coming back from the broadvoice, changing proxy in the host
> file again solves problem,
>
> For me it sounds like broadvoice servers does not want talk with my
> servers for some unknown reason,
>
> I called them but their support is unqualified, and I could not get any
> answer,
>
> Any ideas?
> Did anyone experience something like that?
>
> thx
>
>
> Bartosz Wegrzyn
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>
> --
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.432 / Virus Database: 268.15.18/586 - Release Date:
> 12/13/2006
> 6:13 PM
>
>
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[asterisk-users] bridging calls on a samsung pbx from asterisk

[James Harper]

I have the following configuration:

VoIP Provider <> Asterisk <> Samsung PBX <---> PSTN
  ^
Asterisk has a few VoIP extensions connected, but most of the extensions
are hanging of the Samsung PBX.

Asterisk has 1 NT and 1 TE interface, which are connected to a TE and NT
interface respectively on the Samsung PBX. This allows us to use the
VoIP extensions on Asterisk to make calls to the PSTN, and the Samsung
extensions to make calls via the VoIP provider (dial 0 to get PSTN
dialtone, and 80 to get VoIP 'dialtone').

One feature I'd like to add is to be able to use a workstation (outlook
contacts etc) to start a call between an extension and an outside number
(or another extension?), so that Asterisk would ring your extension, and
when you picked it up it would then dial the outside line and then
connect the channels together.

I believe the above feature is all a solved problem under Asterisk, but
in my case a Samsung extension to PSTN call would tie up 2 ISDN channels
(1 on the NT interface, 1 on the TE interface) for the duration of the
call... is it possible for Asterisk to instruct the PBX to internally
bridge the calls together, and thus get Asterisk out of the call path?
 
I guess it would have to be a function of the ISDN protocol to allow
this sort of thing... I'm pretty sure I read somewhere that you can do
this if they are both on the same ISDN interface but these are on
different interface, and actually different adapters - the NT is a
HFCS-USB and the TE is a NetJet.

Thanks

James

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Re: [asterisk-users] IBM Server / USB Ports

[Matt]

So you are saying that the card is on it's own IRQ and is not sharing
anything with anything?  I realize the eth0 and usb are sharing, but
am not too concerned about that.

On 12/14/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote:

Matt wrote:
> I see that the digium card doesn't share the IRQ however Digium
> has recommended diabled USB still... additionally the Digium card is
> on 169 which isn't a valid IRQ.. how can I find out what it is sharing
> with?
the tdm card is not sharing an interrupt with your USB. It's your LAN card.

169 is valid if you're running on uniprocessor IO-APIC or SMP kernel.

Guess you have to look elsewhere for the source of your crackling.

Try unloading the USB modules from the kernel, i.e.
rmmod uhci_hcd

Leo


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Re: [asterisk-users] (no subject)

[Henry.L.Coleman]

You might want to take a look at the new 4 port FXO from Grandstream
I haven't had one yet to evaluate but assuming it works it is very price
competative and off-loads all the analog (TDM) stuff from your PC
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


> I have been using the sangoma A200 with echo cancelation and I have been
> real happy.
>
> - Original Message -
> From: "Todd- Asterisk" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Thursday, December 14, 2006 3:23 PM
> Subject: [asterisk-users] (no subject)
>
>
>> Hello everyone! I'm planning on setting up a new system shortly and
>> can't
>> pick the right card...  We will have 2 or 3 lines coming in and  7
>> extensions (GXP2k's).  Should I just get 2 or 3 X100P cards?  Or do  I
>> need the Sangoma A20200 or even the A20200D (Echo cancelation)...   I
>> was
>> thinking I'd use a Dell 2.0 GHz machine as the server...  If  anyone has
>> suggestions as to the benifits/problems of each card  choice, I'd love
>> to
>> hear it.
>>  thanks
>>   Todd
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Re: [asterisk-users] Fast Busy

[Henry.L.Coleman]

Sounds like you have a disconnect supervision problem.

Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


> We currently have a pri coming into our asterisk system. Most of the
> time, the did numbers that we call into it work great. However,
> occationally, we get fast busies, but we noticed those busies were not
> due to anyone being on the line, etc...
>
> Any ideas what could cause this? Is this a congestion thing? Is there
> something I should add to the dial plan or configuration of the card to
> fix this?
>
> Thanks,
> Rob
>
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Re: [asterisk-users] On-Hold

[Dovid B]

If using a VOIP phone there should be a button. If using an ATA the 
instructions should be in the manual of the ATA (you also may be able to 
look in the web interface of the device). I forgot how to do it if you are 
using ZAP. Have a look on the wiki.


- Original Message - 
From: "Bartosz Wegrzyn - maillists" <[EMAIL PROTECTED]>

To: 
Sent: Thursday, December 14, 2006 10:30 PM
Subject: [asterisk-users] On-Hold


Hello,

When in conversation, how can I put somebody on hold?

thx


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Re: [asterisk-users] IBM Server / USB Ports

[Leo Ann Boon]

Matt wrote:

I see that the digium card doesn't share the IRQ however Digium
has recommended diabled USB still... additionally the Digium card is
on 169 which isn't a valid IRQ.. how can I find out what it is sharing
with?

the tdm card is not sharing an interrupt with your USB. It's your LAN card.

169 is valid if you're running on uniprocessor IO-APIC or SMP kernel.

Guess you have to look elsewhere for the source of your crackling.

Try unloading the USB modules from the kernel, i.e.
rmmod uhci_hcd

Leo


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Re: [asterisk-users] Zaptel under FC6

[John Novack]

Howard Lowndes wrote:

How old is your mobo?

I have that same problem and I think it because the TDM card will only 
work with PCI 2.2 or later and, although lspci finds the card, udev is 
not installing the zap devices.


Which is why those in the know who don't care to hear Digium's stock 
answer "try another motherboard" choose Sangoma every time

None of that foolishness
Sangoma gives REAL support, not to mention a 5 year Warranty

John Novack


Rudolf Ladyzhenskii wrote:

Thanks for suggestion.

Just tried that Was surprised with download size of only 72k. Anyway,
command work, but I still have same ptoblem.
I tried to run modprobe -- it failed.
Tried to run service zaptel start and get:

service zaptel start
No functioning zap hardware found in /proc/zaptel, loading ztdummy
Loading ztdummy: FATAL: Module ztdummy not found.
  [FAILED]
Running ztcfg:  Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

  [FAILED]
.

I guess, modules are mot there. Running find / -name zaptel* did not
find any modules.
Seems that make is broken in some way.

Rudolf

On 12/15/06, Howard Lowndes <[EMAIL PROTECTED]> wrote:

Have you done yum install zaptel.  It's part of Fedora 6 Extras along
with openpbx, a fork of Asterisk.

Yuan LIU wrote:
>> From: "Rudolf Ladyzhenskii" <[EMAIL PROTECTED]>
>> Now I am trying to install the drivers.
>>
>> # modprobe zaptel
>> FATAL: Module zaptel not found.
>>
>> Fair enough, no zaptel driver is found on the system.
>>
>> Is there are any known problems with FC6? I did not have much 
trouble

>> running on FC3 before.
>
> I'm not running any Fedora, but I suspect that the installation 
layout

> no longer symblink under /lib/modules/ from [full-version] to
> [major-version].  Such is the case with Ubuntu I'm using.  For 
example,
> if your full kernel path is 2.6.15-27-386, you'll find zaptel 
modules in

> /lib/modules/2.6.15/misc/; in the meanwhile, Linux is looking under
> /lib/modules/2.6.15-27-386/ for any loadable kernel modules.  Of 
course

> module not found.
>
> If this is the case, there are two ways to get around.
>
> . Remove physical /lib/modules/2.6.15/, symblink 
/lib/modules/2.6.15 to

> /lib/modules/2.6.15-27-386/, rerun make install; or, alternatively,
>
> . Move /lib/modules/2.6.15/misc/ to under 
/lib/modules/2.6.15-27-386/,

> run depmod.
>
> Both should lead to a happy ending.  I prefer the first one as it 
makes

> future zaptel upgrades happier; of course you can also make symblink
> after the second.
>
> Hope this helps.
>
> Yuan Liu
>
>> Thanks,
>> Rudolf
>
>
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--
Howard.
LANNet Computing Associates - Your Linux people 


When you want a computer system that works, just choose Linux;
When you want a computer system that works, just, choose Microsoft.
--
Flatter government, not fatter government; abolish the Australian 
states.


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Re: [asterisk-users] chan_sip.c:5267 sip_reg_timeout Error

[Dovid B]

Did you do a reload ? Also when you say you commented it out you mean that you 
commented out the register statement ?
  - Original Message - 
  From: [EMAIL PROTECTED] 
  To: asterisk-users@lists.digium.com 
  Sent: Sunday, December 10, 2006 5:02 PM
  Subject: [asterisk-users] chan_sip.c:5267 sip_reg_timeout Error


  I am receiving this message on my asterisk server and I have commented out 
5748150837 in my sip.conf file but it keeps showing this message on the server.

  Dec 10 07:59:31 NOTICE[30448]: chan_sip.c:5267 sip_reg_timeout:-- 
Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1546)


  any ideas?


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Re: [asterisk-users] (no subject)

[Dovid B]

I have been using the sangoma A200 with echo cancelation and I have been 
real happy.


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


Sent: Thursday, December 14, 2006 3:23 PM
Subject: [asterisk-users] (no subject)


Hello everyone! I'm planning on setting up a new system shortly and  can't 
pick the right card...  We will have 2 or 3 lines coming in and  7 
extensions (GXP2k's).  Should I just get 2 or 3 X100P cards?  Or do  I 
need the Sangoma A20200 or even the A20200D (Echo cancelation)...   I was 
thinking I'd use a Dell 2.0 GHz machine as the server...  If  anyone has 
suggestions as to the benifits/problems of each card  choice, I'd love to 
hear it.

 thanks
  Todd
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RE: [asterisk-users] Broadvoice registration problems

[Kevin Kiely]

Any ideas?
Did anyone experience something like that?

Thx

Yes, unfortunately, all the time.  There answer is if it works with a sip
softphone client than it's not their problem.  It does work with the
softphone client.



-Original Message-
From: Bartosz Wegrzyn - maillists [mailto:[EMAIL PROTECTED] 
Sent: Thursday, December 14, 2006 3:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Broadvoice registration problems

Hello,

I have two broadvoice accounts.
Lately, very often my broadvoice accounts are in unregistered state.
When I log into asterisk I see:

voip*CLI> sip show registry
Host Username Refresh State
sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent
sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent

xxx and yyy are my ohone numbers,

when I do tcpdump on my router, I see only packets comming out my router
to broadvoice but nothing comming back on port 5060

23:06:41.398952 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411
23:06:42.306975 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411
23:06:42.398951 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411
23:06:44.306932 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411
23:06:44.398921 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411

the only way to get registered is to change the proxy in the host file,
after some time same story happens, I am in register sent state and
nothing is coming back from the broadvoice, changing proxy in the host
file again solves problem,

For me it sounds like broadvoice servers does not want talk with my
servers for some unknown reason,

I called them but their support is unqualified, and I could not get any
answer,

Any ideas?
Did anyone experience something like that?

thx


Bartosz Wegrzyn
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-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.432 / Virus Database: 268.15.18/586 - Release Date: 12/13/2006
6:13 PM
 

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[asterisk-users] Re: Zaptel under FC6

[Axel Thimm]

On Fri, Dec 15, 2006 at 12:05:13AM +1100, Rudolf Ladyzhenskii wrote:
> Hi, all
> 
> I am building a new server. Have installed FC 6 and put in TDM400 card.
> 
> Checked out latest asteriusk code, run make install in zaptel directory.
> So far all is fine.
> 
> Now I am trying to install the drivers.
> 
> # modprobe zaptel
> FATAL: Module zaptel not found.
> 
> Fair enough, no zaptel driver is found on the system.
> 
> Is there are any known problems with FC6? I did not have much trouble
> running on FC3 before.

You'll find zaptel, asterisk and all the glue packaged at ATrpms:

http://www.voip-info.org/wiki/view/ATrpms

-- 
Axel.Thimm at ATrpms.net


pgpxPocq6g1Sg.pgp
Description: PGP signature
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Re: [asterisk-users] Zaptel under FC6

[simon elliston ball]

The Fedora Extras rpm is tiny because it has nothing really of help  
in it. It's missing the modules.


I've had some success on Fedora Core 6 using the ATrpms repository,  
which has the zaptel-kmdl package for most variations of kernels  
included in FC6.


Simon


On 14 Dec 2006, at 22:31, Yuan LIU wrote:


From: "Rudolf Ladyzhenskii" <[EMAIL PROTECTED]>
I guess, modules are mot there. Running find / -name zaptel* did not
find any modules.


Be careful here - wildcard expansion takes place locally unless you  
quote the string:


$ find / -name 'zaptel*'

Of course search from / is suboptimal as you are going to find your  
source as well, besides a looong search.  I suggest starting from / 
lib/modules.  Or do a simple ls.



Seems that make is broken in some way.

Rudolf



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RE: [asterisk-users] Diva Server V-BRI-2 and internal numbers

[Gregory Duchatelet]

> It looks like 107 is busy ;-)
> Please increase verbosity, like
>   set verbose 5
>   capi debug
> to see what is happening.

Hi Armin,

Verbose was at 30 :)
107 is not busy since i can call it from 102, which is another internal
phone. All internal phones are busy for Asterisk...

Here is the log with verbose at 100 and capi debug enabled :


-- Executing Dial("SIP/Greg-081f5a10", "CAPI/ISDN1/b:107||rtT") in new
stack
   > data = ISDN1/b:107
   > parsed dialstring: 'ISDN1' 'b' '107' ''
   > capi request for interface 'ISDN1'
   > parsed dialstring: 'ISDN1' 'b' '107' ''
  == ISDN1: Call CAPI/ISDN1/107-1e   (pres=0x00, ton=0x00)
CONNECT_REQ ID=001 #0x1002 LEN=0047
  Controller/PLCI/NCCI= 0x1
  CIPValue= 0x1
  CalledPartyNumber   = <80>107
  CallingPartyNumber  = <00 80>b
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BProtocol
   B1protocol = 0x1
   B2protocol = 0x1
   B3protocol = 0x0
   B1configuration= default
   B2configuration= default
   B3configuration= default
   GlobalConfiguration= default
  BC  = default
  LLC = default
  HLC = default
  AdditionalInfo
   BChannelinformation= <00 00>
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default
   SendingComplete= default

-- Called ISDN1/b:107
   > CAPI devicestate requested for ISDN1/107
   > CAPI devicestate requested for ISDN1/107
CONNECT_CONF ID=001 #0x1002 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

-- ISDN1: received CONNECT_CONF PLCI = 0x101
INFO_IND ID=001 #0x11a8 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = <8a>

INFO_RESP ID=001 #0x11a8 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1: info element CHANNEL IDENTIFICATION 8a
INFO_IND ID=001 #0x11a9 LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x800d
  InfoElement = default

INFO_RESP ID=001 #0x11a9 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1: info element SETUP ACK
INFO_IND ID=001 #0x11ab LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8002
  InfoElement = default

INFO_RESP ID=001 #0x11ab LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1: info element CALL PROCEEDING
-- CAPI/ISDN1/107-1e is proceeding passing it to SIP/Greg-081f5a10
INFO_IND ID=001 #0x11ad LEN=0037
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x1c
  InfoElement = <91 a1 13 02 02 8f> <02 01 22>0<0a a1
05>0<03 02 01 00 82 01 01>

INFO_RESP ID=001 #0x11ad LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1: info element FACILITY
INFO_IND ID=001 #0x11ae LEN=0017
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8
  InfoElement = <81 d8>

INFO_RESP ID=001 #0x11ae LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1: info element CAUSE 81 d8
INFO_IND ID=001 #0x11af LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8045
  InfoElement = default

INFO_RESP ID=001 #0x11af LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1: info element DISCONNECT
-- ISDN1: Disconnect case 1
-- CAPI/ISDN1/107-1e is busy
  == ISDN1: CAPI Hangingup
-- ISDN1: activehangingup (cause=88)
DISCONNECT_REQ ID=001 #0x1003 LEN=0018
  Controller/PLCI/NCCI= 0x101
  AdditionalInfo
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default
   SendingComplete= default

  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing Hangup("SIP/Greg-081f5a10", "") in new stack
  == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on
'SIP/Greg-081f5a10' in macro 'appel_sortant'
  == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on
'SIP/Greg-081f5a10'
   > CAPI devicestate requested for ISDN1/107
   > CAPI devicestate requested for ISDN1/107
DISCONNECT_CONF ID=001 #0x1003 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

INFO_IND ID=001 #0x11b1 LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x805a
  InfoElement = default

INFO_RESP ID=001 #0x11b1 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1: info element RELEASE COM

[asterisk-users] VoipTalk unable to accept calls at present?

[Charlie Grosvenor]

I am trying to get asterisks to work with http://www.voiptalk.org 's IAX
service. I have configured asterisks as per their instructions and am
using the x-lite soft phone. When I get an incoming call the softphone
rings but the caller (from pstn) gets a recorded message saying the
number is unable to accept calls at present. Does anybody know what
might be causing this?

 

Thanks

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Re: [asterisk-users] Console latency

[Eric \"ManxPower\" Wieling]

Yuan LIU wrote:

From: "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>

Yuan LIU wrote:
Another bizarry: If I run the Echo application from the console, I 
can hear a very long delay (upward to 1,000 ms).  I can run the same 
application from a GrandStream phone (on the same LAN) and hear 
little delay.  What could possibly be wrong?  If it were interrupt 
overload, I'd hear lots of cracks in my echo, right?  I'm not hearing 
that.  Besides, a telephony card is not involved.


esd and artsd do buffering of audio.  Could that be the problem?


I'm totally ignorant about sound processing.  My system runs esd.  Any 
suggestions about what to tweak?  Or any test scenario that can 
definitely determine whether it is esd?  Thanks.


You could "killall esd" and see if anything blows up.
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Re: [asterisk-users] Zaptel under FC6

[Yuan LIU]

From: "Rudolf Ladyzhenskii" <[EMAIL PROTECTED]>
I guess, modules are mot there. Running find / -name zaptel* did not
find any modules.


Be careful here - wildcard expansion takes place locally unless you quote 
the string:


$ find / -name 'zaptel*'

Of course search from / is suboptimal as you are going to find your source 
as well, besides a looong search.  I suggest starting from /lib/modules.  Or 
do a simple ls.



Seems that make is broken in some way.

Rudolf



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Re: [asterisk-users] StripXXX apps missing fromasterisk-1.2.13?

[Yuan LIU]

From: "john beaman" <[EMAIL PROTECTED]>

StripLSD is obsolete: 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+StripLSD

StripMSD is being phased out: http://bugs.digium.com/view.php?id=5673

John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331


Thanks a lot.

Yuan Liu


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Re: [asterisk-users] Console latency

[Yuan LIU]

From: "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>

Yuan LIU wrote:
Another bizarry: If I run the Echo application from the console, I can 
hear a very long delay (upward to 1,000 ms).  I can run the same 
application from a GrandStream phone (on the same LAN) and hear little 
delay.  What could possibly be wrong?  If it were interrupt overload, I'd 
hear lots of cracks in my echo, right?  I'm not hearing that.  Besides, a 
telephony card is not involved.


esd and artsd do buffering of audio.  Could that be the problem?


I'm totally ignorant about sound processing.  My system runs esd.  Any 
suggestions about what to tweak?  Or any test scenario that can definitely 
determine whether it is esd?  Thanks.


Yuan Liu


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Re: [asterisk-users] StripXXX apps missing from asterisk-1.2.13?

[Eric \"ManxPower\" Wieling]

Yuan LIU wrote:
All of StripMSD, StripLSD, etc., are missing when I downloaded 
asterisk-1.2-current.tar.gz, which explodes into 1.2.13.  Are the strip 
club deprecated?  What replacement functions should I use?


See README.variables in the Asterisk source.
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Re: [asterisk-users] Fast Busy

[Eric \"ManxPower\" Wieling]

Rob Schall wrote:

We currently have a pri coming into our asterisk system. Most of the
time, the did numbers that we call into it work great. However,
occationally, we get fast busies, but we noticed those busies were not
due to anyone being on the line, etc...

Any ideas what could cause this? Is this a congestion thing? Is there
something I should add to the dial plan or configuration of the card to
fix this?


The telco will provide a Cause Code. This should be in HANGUPCAUSE.
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Re: [asterisk-users] Ssh access over a zap channel...

[Yuan LIU]

From: "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>

[Goodies skipped]

  -= Info about application 'ZapRAS' =-

[Synopsis]
Executes Zaptel ISDN RAS application

[Description]
  ZapRAS(args): Executes a RAS server using pppd on the given channel.
The channel must be a clear channel (i.e. PRI source) and a Zaptel
channel to be able to use this function (No modem emulation is included).


A side question: what's preventing Asterisk from emulating MODEM?

From what I see, an X100P card is but a DSPless MODEM, and Asterisk includes 
a DSP engine. (Effectively, a PRI interface can also be viewed as a DSPless 
MODEM pool.)  So theoretically, combining these two functions should give us 
MODEM capability, right?  Or maybe there is already such an application? (An 
Asterisk dialup BBS, for old time's sake...)  In fact, Motorola does provide 
a Linux driver for its SM56 soft MODEM's - well known X100P clones.  Can 
Asterisk make use of Motorola's driver?


Yuan Liu


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Re: [asterisk-users] Zaptel under FC6

[Rudolf Ladyzhenskii]

Thanks for suggestion.

Just tried that Was surprised with download size of only 72k. Anyway,
command work, but I still have same ptoblem.
I tried to run modprobe -- it failed.
Tried to run service zaptel start and get:

service zaptel start
No functioning zap hardware found in /proc/zaptel, loading ztdummy
Loading ztdummy: FATAL: Module ztdummy not found.
  [FAILED]
Running ztcfg:  Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

  [FAILED]
.

I guess, modules are mot there. Running find / -name zaptel* did not
find any modules.
Seems that make is broken in some way.

Rudolf

On 12/15/06, Howard Lowndes <[EMAIL PROTECTED]> wrote:

Have you done yum install zaptel.  It's part of Fedora 6 Extras along
with openpbx, a fork of Asterisk.

Yuan LIU wrote:
>> From: "Rudolf Ladyzhenskii" <[EMAIL PROTECTED]>
>> Now I am trying to install the drivers.
>>
>> # modprobe zaptel
>> FATAL: Module zaptel not found.
>>
>> Fair enough, no zaptel driver is found on the system.
>>
>> Is there are any known problems with FC6? I did not have much trouble
>> running on FC3 before.
>
> I'm not running any Fedora, but I suspect that the installation layout
> no longer symblink under /lib/modules/ from [full-version] to
> [major-version].  Such is the case with Ubuntu I'm using.  For example,
> if your full kernel path is 2.6.15-27-386, you'll find zaptel modules in
> /lib/modules/2.6.15/misc/; in the meanwhile, Linux is looking under
> /lib/modules/2.6.15-27-386/ for any loadable kernel modules.  Of course
> module not found.
>
> If this is the case, there are two ways to get around.
>
> . Remove physical /lib/modules/2.6.15/, symblink /lib/modules/2.6.15 to
> /lib/modules/2.6.15-27-386/, rerun make install; or, alternatively,
>
> . Move /lib/modules/2.6.15/misc/ to under /lib/modules/2.6.15-27-386/,
> run depmod.
>
> Both should lead to a happy ending.  I prefer the first one as it makes
> future zaptel upgrades happier; of course you can also make symblink
> after the second.
>
> Hope this helps.
>
> Yuan Liu
>
>> Thanks,
>> Rudolf
>
>
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--
Howard.
LANNet Computing Associates - Your Linux people 
When you want a computer system that works, just choose Linux;
When you want a computer system that works, just, choose Microsoft.
--
Flatter government, not fatter government; abolish the Australian states.

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Re: [asterisk-users] StripXXX apps missing from asterisk-1.2.13?

[john beaman]

StripLSD is obsolete: 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+StripLSD
StripMSD is being phased out: http://bugs.digium.com/view.php?id=5673



John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331

>>> [EMAIL PROTECTED] 12/14/2006 3:55:59 PM >>>
All of StripMSD, StripLSD, etc., are missing when I downloaded 
asterisk-1.2-current.tar.gz, which explodes into 1.2.13.  Are the strip club 
deprecated?  What replacement functions should I use?

Yuan Liu


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Re: [asterisk-users] Console latency

[Eric \"ManxPower\" Wieling]

Yuan LIU wrote:
Another bizarry: If I run the Echo application from the console, I can 
hear a very long delay (upward to 1,000 ms).  I can run the same 
application from a GrandStream phone (on the same LAN) and hear little 
delay.  What could possibly be wrong?  If it were interrupt overload, 
I'd hear lots of cracks in my echo, right?  I'm not hearing that.  
Besides, a telephony card is not involved.


esd and artsd do buffering of audio.  Could that be the problem?

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[asterisk-users] StripXXX apps missing from asterisk-1.2.13?

[Yuan LIU]

All of StripMSD, StripLSD, etc., are missing when I downloaded 
asterisk-1.2-current.tar.gz, which explodes into 1.2.13.  Are the strip club 
deprecated?  What replacement functions should I use?


Yuan Liu


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[asterisk-users] Fast Busy

[Rob Schall]

We currently have a pri coming into our asterisk system. Most of the
time, the did numbers that we call into it work great. However,
occationally, we get fast busies, but we noticed those busies were not
due to anyone being on the line, etc...

Any ideas what could cause this? Is this a congestion thing? Is there
something I should add to the dial plan or configuration of the card to
fix this?

Thanks,
Rob

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[asterisk-users] Console latency

[Yuan LIU]

Another bizarry: If I run the Echo application from the console, I can hear 
a very long delay (upward to 1,000 ms).  I can run the same application from 
a GrandStream phone (on the same LAN) and hear little delay.  What could 
possibly be wrong?  If it were interrupt overload, I'd hear lots of cracks 
in my echo, right?  I'm not hearing that.  Besides, a telephony card is not 
involved.


I'm running asterisk-1.2.13  and  zaptel-1.2.10 on Linux 2.6.15-27-386 
(Ubuntu 6 distribution without X).  Hardware includes a P III 600 MHz, 386 
MB RAM, an X100P card that's not part of this test (also used an X100P clone 
card to same result), and a CS4239 sound card (ISA) with ALSA driver (also 
tried with OSS to similar result but OSS had a harder time getting volume 
up).  ALSA needed a bit of tweak to work properly with CS4239, but afer 
carefully setting alsamixer, I don't hear much echo when making calls from 
the console.


Yuan Liu


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Re: [asterisk-users] Ssh access over a zap channel...

[Eric \"ManxPower\" Wieling]

Jordan Novak wrote:
My need to do this through asterisk is simply the ability to provide me 
access with no additional cost to my customer. It seems like a nice 
thing to include as long as authentication is done well. I have 
worked on a dozen or more types of switches and all of them have 
supported this or had the capabilty through hardware or licensing. I am 
trying to get around opening and closing the firewall, which at some 
locations is simply not accessible to me.


pbx-1*CLI> show application ZapRAS
pbx-1*CLI>
  -= Info about application 'ZapRAS' =-

[Synopsis]
Executes Zaptel ISDN RAS application

[Description]
  ZapRAS(args): Executes a RAS server using pppd on the given channel.
The channel must be a clear channel (i.e. PRI source) and a Zaptel
channel to be able to use this function (No modem emulation is included).
Your pppd must be patched to be zaptel aware. Arguments should be
separated by | characters.

pbx-1*CLI>
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Re: [asterisk-users] IBM Server / USB Ports

[Eric \"ManxPower\" Wieling]

Matt wrote:

Hi,
I have an IBM xSeries server... and the digium card is sharing IRQ
with USB and giving me crackling audio.


cat /proc/interrupts

It brings up these results:
   0:   10566547IO-APIC-edge  timer
   1:  9IO-APIC-edge  i8042
   2:  0  XT-PIC  cascade
   8:  1IO-APIC-edge  rtc
  12: 93IO-APIC-edge  i8042
  14: 23IO-APIC-edge  ide0
121: 481478   IO-APIC-level  uhci_hcd:usb2, eth0
137:  0   IO-APIC-level  uhci_hcd:usb3
153:  0   IO-APIC-level  ehci_hcd:usb1
161:  28479   IO-APIC-level  ips
169:   10526429   IO-APIC-level  wctdm24xxp


IBM is saying that I can not disable the USB ports!Can anyone
offer any insight.. either as to how to disable the USB ports, or how
to make this work on an IBM xSeries?


Move the card to a different slot.
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[asterisk-users] Voicemail Live

[Fernando BERRETTA]

Hi,

Philipp von Klitzing posted this solution in Dec. 2005
Answering machine mimic: Listen while caller is leaving voicemail for
you; with pick-up option

Is there any other way to listen while caller is leaving a voicemail for
you?

Thanks
Fernando
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Re: [asterisk-users] matching the beginning of an EXTEN

[Eric \"ManxPower\" Wieling]

Joao Pereira wrote:

Hello
how can I distinguish all the calls that arrive to my Asterisk starting 
with: 351217588XXX ?
I want match the first 9 digits does Asterisk has any function for 
this?


exten => _51217588XXX,1,Whatever
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Re: [asterisk-users] Hardware TDM Switching

[Eric \"ManxPower\" Wieling]

[EMAIL PROTECTED] wrote:

Do anybody know, if there is a way to connect 2 zap-channels with 
Hardware TDM Switching?


It's called DACS.  See the /etc/zapata.conf config file sample.
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[asterisk-users] Re: Vonage SIP access via asterisk?

[Steven]

I can dial out via FWD because the login is in the session.

Vonage requires a register even for outbound.
My understanding is that they log the register and then any call from that IP 
is from that user.
This is why I can't dial out vonage.

The root cause is that sip is not registering at all.

I assume the firewall is OK, because I can make outbound SIP calls to FWD.

Is there a way to force a registry so that I can capture the packets?

-- 
-- 
Steven

http://www.glimasoutheast.org



"Steven" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> This may not be vonage related as it appears that I can not register with any 
> sip servers.
> I tried FWD and also get a black "sip show registry"
>
> Could it be a firewall issue?
> I am running IP tables on the computer which is on the internet with no NAT.
> Asterisk 1.2.13
>
> I have allow outbound all.
> Allow inbound 5060, IAX and RTP.
>
>
> -- 
> -- 
> Steven
>
> http://www.glimasoutheast.org
>
>
>
> "Steven" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
>> That and any other ref.s I have found give me a 404 error when dialing out.
>>
>> My Sip show registry is also empty.
>>
>> ref:
>> We're at 64.x.x.x port 12146
>> Adding codec 0x4 (ulaw) to SDP
>> Adding codec 0x8 (alaw) to SDP
>> Adding codec 0x1 (g723) to SDP
>> Adding codec 0x2 (gsm) to SDP
>> Adding codec 0x10 (g726) to SDP
>> Adding codec 0x20 (adpcm) to SDP
>> Adding codec 0x40 (slin) to SDP
>> Adding codec 0x80 (lpc10) to SDP
>> Adding codec 0x100 (g729) to SDP
>> Adding codec 0x200 (speex) to SDP
>> Adding codec 0x400 (ilbc) to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>> 13 headers, 21 lines
>> Reliably Transmitting (NAT) to 216.115.20.41:5061:
>> INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
>> Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport
>> From: "SteveB TEST" ;tag=as35e23a92
>> To: 
>> Contact: 
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX
>> Max-Forwards: 70
>> Date: Fri, 08 Dec 2006 17:15:22 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Type: application/sdp
>> Content-Length: 494
>>
>> v=0
>> o=root 9983 9983 IN IP4 64.118.155.160
>> s=session
>> c=IN IP4 64.118.155.160
>> t=0 0
>> m=audio 12146 RTP/AVP 0 8 4 3 111 5 10 7 18 110 97 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:111 G726-32/8000
>> a=rtpmap:5 DVI4/8000
>> a=rtpmap:10 L16/8000
>> a=rtpmap:7 LPC/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:110 speex/8000
>> a=rtpmap:97 iLBC/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>>
>> ---
>>-- Called [EMAIL PROTECTED]
>> tg05*CLI>
>> <-- SIP read from 216.115.20.41:5061:
>> SIP/2.0 404 Not Found
>> Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport
>> From: "SteveB TEST" ;tag=as35e23a92
>> To: 
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 102 INVITE
>> Max-Forwards: 15
>> Content-Length: 0
>>
>>
>> --- (8 headers 0 lines) ---
>> Transmitting (NAT) to 216.115.20.41:5061:
>> ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0
>> Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport
>> From: "SteveB TEST" ;tag=as35e23a92
>> To: 
>> Contact: 
>> Call-ID: [EMAIL PROTECTED]
>> CSeq: 102 ACK
>> User-Agent: Asterisk PBX
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> -- 
>> -- 
>> Steven
>>
>> http://www.glimasoutheast.org
>>
>>
>>
>> "Al Bochter" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
>>> http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#621VonageBusinessPlusandVonageSoftphoneb
>>>
>>> Best regards,
>>>
>>> Al Bochter
>>> Bochter Services
>>> http://www.BochterServices.com/?t=Email
>>>
>>> (VOIP PBX) 1-866-638-1254
>>>
>>> (Voip PBX) Free World DialUp: 780-217
>>> WebSite: http://www.freeworlddialup.com/
>>>
>>> We have Toll Free DID's instock
>>> * * * NO MONTHLY FEE - LIMITED TIME ONLY * * *
>>> http://www.bochterservices.com/?t=TF(NM)did
>>>
>>> BUY Coins, Silver and Gold
>>> http://www.bochterservices.com/?j=gold&t=email
>>>
>>> For new and used security items
>>> http://www.bochterservices.com/?j=store&t=email_security
>>>
>>>
>>>
>>> BerkHolz, Steven wrote:
>>>
Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA)

I just signed up to test their service and they sent me a Number, Proxy, 
port and password.

Every reference I have tried leaves me with a 404 error coming from Vonage.

If you have a working setup, please post some config references.


 Thank You,
Steven BerkHolz



Soon to be known as HIROTEC AMERICA
www.hirotecamerica.com
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>>

[asterisk-users] Re: Vonage SIP access via asterisk?

[Steven]

This may not be vonage related as it appears that I can not register with any 
sip servers.
I tried FWD and also get a black "sip show registry"

Could it be a firewall issue?
I am running IP tables on the computer which is on the internet with no NAT.
Asterisk 1.2.13

I have allow outbound all.
Allow inbound 5060, IAX and RTP.


-- 
-- 
Steven

http://www.glimasoutheast.org



"Steven" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> That and any other ref.s I have found give me a 404 error when dialing out.
>
> My Sip show registry is also empty.
>
> ref:
> We're at 64.x.x.x port 12146
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x1 (g723) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x10 (g726) to SDP
> Adding codec 0x20 (adpcm) to SDP
> Adding codec 0x40 (slin) to SDP
> Adding codec 0x80 (lpc10) to SDP
> Adding codec 0x100 (g729) to SDP
> Adding codec 0x200 (speex) to SDP
> Adding codec 0x400 (ilbc) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> 13 headers, 21 lines
> Reliably Transmitting (NAT) to 216.115.20.41:5061:
> INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
> Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport
> From: "SteveB TEST" ;tag=as35e23a92
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Fri, 08 Dec 2006 17:15:22 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 494
>
> v=0
> o=root 9983 9983 IN IP4 64.118.155.160
> s=session
> c=IN IP4 64.118.155.160
> t=0 0
> m=audio 12146 RTP/AVP 0 8 4 3 111 5 10 7 18 110 97 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:10 L16/8000
> a=rtpmap:7 LPC/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:110 speex/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
>-- Called [EMAIL PROTECTED]
> tg05*CLI>
> <-- SIP read from 216.115.20.41:5061:
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport
> From: "SteveB TEST" ;tag=as35e23a92
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> Max-Forwards: 15
> Content-Length: 0
>
>
> --- (8 headers 0 lines) ---
> Transmitting (NAT) to 216.115.20.41:5061:
> ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0
> Via: SIP/2.0/UDP 64.118.155.160:5060;branch=z9hG4bK455fdd1c;rport
> From: "SteveB TEST" ;tag=as35e23a92
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
> -- 
> -- 
> Steven
>
> http://www.glimasoutheast.org
>
>
>
> "Al Bochter" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
>> http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#621VonageBusinessPlusandVonageSoftphoneb
>>
>> Best regards,
>>
>> Al Bochter
>> Bochter Services
>> http://www.BochterServices.com/?t=Email
>>
>> (VOIP PBX) 1-866-638-1254
>>
>> (Voip PBX) Free World DialUp: 780-217
>> WebSite: http://www.freeworlddialup.com/
>>
>> We have Toll Free DID's instock
>> * * * NO MONTHLY FEE - LIMITED TIME ONLY * * *
>> http://www.bochterservices.com/?t=TF(NM)did
>>
>> BUY Coins, Silver and Gold
>> http://www.bochterservices.com/?j=gold&t=email
>>
>> For new and used security items
>> http://www.bochterservices.com/?j=store&t=email_security
>>
>>
>>
>> BerkHolz, Steven wrote:
>>
>>>Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA)
>>>
>>>I just signed up to test their service and they sent me a Number, Proxy, 
>>>port and password.
>>>
>>>Every reference I have tried leaves me with a 404 error coming from Vonage.
>>>
>>>If you have a working setup, please post some config references.
>>>
>>>
>>> Thank You,
>>>Steven BerkHolz
>>>
>>>
>>>
>>>Soon to be known as HIROTEC AMERICA
>>>www.hirotecamerica.com
>>>___
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>>>
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>>>To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>>
>>>
>>>Inbound (clean). Database: 0654-1, 12/07/2006 - 12/8/2006 11:10:08 AM
>>>
>>>
>>>
>>>
>>>
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>
>
>
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RE: [asterisk-users] Zaptel under FC6

[Yuan LIU]

From: "Rudolf Ladyzhenskii" <[EMAIL PROTECTED]>
Now I am trying to install the drivers.

# modprobe zaptel
FATAL: Module zaptel not found.

Fair enough, no zaptel driver is found on the system.

Is there are any known problems with FC6? I did not have much trouble
running on FC3 before.


I'm not running any Fedora, but I suspect that the installation layout no 
longer symblink under /lib/modules/ from [full-version] to [major-version].  
Such is the case with Ubuntu I'm using.  For example, if your full kernel 
path is 2.6.15-27-386, you'll find zaptel modules in 
/lib/modules/2.6.15/misc/; in the meanwhile, Linux is looking under 
/lib/modules/2.6.15-27-386/ for any loadable kernel modules.  Of course 
module not found.


If this is the case, there are two ways to get around.

. Remove physical /lib/modules/2.6.15/, symblink /lib/modules/2.6.15 to 
/lib/modules/2.6.15-27-386/, rerun make install; or, alternatively,


. Move /lib/modules/2.6.15/misc/ to under /lib/modules/2.6.15-27-386/, run 
depmod.


Both should lead to a happy ending.  I prefer the first one as it makes 
future zaptel upgrades happier; of course you can also make symblink after 
the second.


Hope this helps.

Yuan Liu


Thanks,
Rudolf



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Re: [asterisk-users] IBM Server / USB Ports

[Tzafrir Cohen]

On Thu, Dec 14, 2006 at 11:13:29AM -0500, Matt wrote:
> Hi,
> I have an IBM xSeries server... 

Which model, exactly? With which customizations? 

> and the digium card is sharing IRQ
> with USB and giving me crackling audio.

Do you actually get any interrupts from the USB? 

> 
> >>>cat /proc/interrupts
> >>>
> >>>It brings up these results:
> >>>   0:   10566547IO-APIC-edge  timer
> >>>   1:  9IO-APIC-edge  i8042
> >>>   2:  0  XT-PIC  cascade
> >>>   8:  1IO-APIC-edge  rtc
> >>>  12: 93IO-APIC-edge  i8042
> >>>  14: 23IO-APIC-edge  ide0
> >>>121: 481478   IO-APIC-level  uhci_hcd:usb2, eth0
> >>>137:  0   IO-APIC-level  uhci_hcd:usb3
> >>>153:  0   IO-APIC-level  ehci_hcd:usb1

Try connecting a USB device. I guess you'd suddenly get interrupts here.
I bet all the interrupts on 121 are from the network controller.

> >>>161:  28479   IO-APIC-level  ips
> >>>169:   10526429   IO-APIC-level  wctdm24xxp
> 
> IBM is saying that I can not disable the USB ports!Can anyone
> offer any insight.. either as to how to disable the USB ports, or how
> to make this work on an IBM xSeries?

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

[Bartosz Wegrzyn - maillists]

Hello,

When in conversation, how can I put somebody on hold?

thx


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[asterisk-users] Broadvoice registration problems

[Bartosz Wegrzyn - maillists]

Hello,

I have two broadvoice accounts.
Lately, very often my broadvoice accounts are in unregistered state.
When I log into asterisk I see:

voip*CLI> sip show registry
Host Username Refresh State
sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent
sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent

xxx and yyy are my ohone numbers,

when I do tcpdump on my router, I see only packets comming out my router
to broadvoice but nothing comming back on port 5060

23:06:41.398952 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411
23:06:42.306975 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411
23:06:42.398951 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411
23:06:44.306932 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411
23:06:44.398921 IP adsl-myip.dsl.chcgil.ameritech.net.5060 >
147.135.12.128.5060: UDP, length 411

the only way to get registered is to change the proxy in the host file,
after some time same story happens, I am in register sent state and
nothing is coming back from the broadvoice, changing proxy in the host
file again solves problem,

For me it sounds like broadvoice servers does not want talk with my
servers for some unknown reason,

I called them but their support is unqualified, and I could not get any
answer,

Any ideas?
Did anyone experience something like that?

thx


Bartosz Wegrzyn
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Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

[Yuan LIU]

From: Steve Prior <[EMAIL PROTECTED]>

The feature request # is 4542, but I don't know any associated bug number, 
nor with what phones other people had to tweak.  My phone is a GE 
27935GE3-B. (Don't know what possessed me to say GM:-)


Yuan Liu


Just gotta ask - you did plug in the power supply connection on the
board, right?


That's right - and how other phones get to ring properly.  Two surprises 
when a GE phone sold in U.S. won't work "out of the PBX": 1) Why this phone 
requires significantly higher voltage, and 2) why zaptel wants to set 
default lower than normal U.S. lines when configured for use in the U.S. 
(The Linksys ATA rings this GE without a problem.)


Yuan Liu


Steve
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Re: [asterisk-users] (no subject)

[Ira]

At 05:23 AM 12/14/2006, you wrote:

Should I just get 2 or 3 X100P cards?  Or do
I need the Sangoma A20200 or even the A20200D (Echo cancelation)...



When I started down this path I choose the TDM04 and have always had 
occasional echo issues, not bad and not often, but it annoys the wife 
and one of these days I'll sell the TDM04 and replace it with the 
A20002D so I have hardware echo cancellation.  Someone else a few 
months back said the same thing about all the small business 
installations he did because he just didn't want to have complaints 
and the extra $300 was a small price to pay for peace of mind.


All that said, I don't have the Sangoma card yet and have never seen 
one so I could be blowing smoke!


Ira 


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Re: [Asterisk-Users] Siemens Gigaset SL75

[Ira]

At 03:40 AM 12/14/2006, you wrote:
I tested ZyXel Prestige 2000 but the sound was bad and it doesnt 
support 802.1x :(


Wow, I've always been impressed with the sound from my Zyxel 2000W.

Ira 


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[asterisk-users] Show agent queue status on the phone?

[Kevin Trumbull]

Title: Message



Hi All, Is it 
possible to show an agent's queue status on the phone? For example, in 
our current non-asterisk PBX, if a 
member of a call queue does not answer the phone when a queue call is sent to 
them, they go to a 'not ready' status, and this is indicated on their phone. So 
when they return to their desk, they can see that they are not ready, so they 
hit a button to put themselves back into a ready status. I can 
accomplish the 'not ready' functionality by using the PauseQueueMember function, 
but now I need to somehow display the pause/unpause status on the phone so the 
staff member knows if they got paused. Does anybody know how to do this?  This might be phone specific, so I'll mention 
that we are using Cisco 7961G's.
 
Thanks
 
-Kevin 
Trumbull
 
 


smime.p7s
Description: S/MIME cryptographic signature
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RE: [asterisk-users] Web-MeetMe ready for prime time?

[Dan Austin]

Jeremy wrote:
> What kind of luck are people having with the Web-MeetMe control?  The
> condition of the page on the voip-info wiki makes me a bit nervous
about
> putting Web-MeetMe into a production environment.  Use of MeetMe has
> really taken off here since installation and I need a scheduling and
> provisioning system for PIN numbers etc.  Are there any other
solutions
> out there?

It is funny that you should ask now.  I've reworked quite a bit of the
code and have been waiting for Asterisk 1.4 to be released before I
announced the changes.  It is possible that I'll need to tweak the
code after 1.4 is ready (but I hope not too much)

After 1.4 is out there will be two 'supported' Web-MeetMe releases:
2.X.X for Asterisk 1.2
and 
3.X.X for Asterisk 1.4

The reason for maintaining two versions relates to changes in the
Asterisk CLI, and a single WMM release would have to have extra,
likely fragile, code to support both.

Changes and new features in the yet to be released updates:

*  No longer needs register_globals
*  Replaced the DB abstraction layer with PEAR:DB
This opens up the possibility of a much wider choice 
of backend databases. but the Asterisk schedular
application is still MySQL only (I might be able to
migrate it to ODBC later)
*  Some cosmetic re-work
*  Better installation instructions and release notes
*  The 3.X chain also has a very basic report generator.
It is a little crude as I am learning GD-Lib as I go.

The project is now hosted on SourceForge and while it has not
seen a lot of updates recently, is still active.

I have had the 2.1.0 release in production on three moderatly
loaded production servers for close to one year.  I am aware of
a number of much larger installations, which I can only hope will
share their experiences.

Dan

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Re: [asterisk-users] Web-MeetMe ready for prime time?

[Bruce Reeves]

You might check out http://sourceforge.net/projects/web-meetme and version
2.1.0, I had to tweak it a little, but it has worked well for people to
schedule their own meetme conferences.

On 12/14/06, Porier, Jeremy M. <[EMAIL PROTECTED]> wrote:


What kind of luck are people having with the Web-MeetMe control?  The
condition of the page on the voip-info wiki makes me a bit nervous about
putting Web-MeetMe into a production environment.  Use of MeetMe has
really taken off here since installation and I need a scheduling and
provisioning system for PIN numbers etc.  Are there any other solutions
out there?

Thanks,
Jeremy
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--
Bruce
Nortex Networks
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Re: [asterisk-users] asterisk billing software

[Andrew Joakimsen]

Sorry to bring back a post from the grave, but what do you feel is the worst
problem you are having with AgileVoice? Is their support easy to reach?

On 11/16/06, Chris Mazuc <[EMAIL PROTECTED]> wrote:


Andrew Joakimsen wrote:
> Chris:
>
> We were evaluating AgileVoice currently, could you please elaborte on
> your problems? Did they not do the instattion for you?

They did the installation.

I'm going to be very careful with my wording here, but if you are
currently evaluating their software, I recommend you ask to see *all* of
the documentation for AgileVoice (not just AgileBill).

-Chris
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RE: [asterisk-users] how to define a secure trunk

[turby]

right, but who have production and tested code of application level
encryption for SIP and IAX for SECURE(!) trunks? 
turby

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Thursday, December 14, 2006 6:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] how to define a secure trunk

tunneling small rtp packets through vpn has big overhead, better to use
application level encryption - encrypted iax or srtp.
PJ



[EMAIL PROTECTED] wrote:
> joao,
> you can use ssh tunel, pptp or vpn for any sip/iax trunks or users.
>
> turby
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Pavel 
> Jezek
> Sent: Thursday, December 14, 2006 4:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] how to define a secure trunk
>
> as I know, only preliminary support:
>
> 0005413: [patch] Secure RTP (SRTP)
> http://bugs.digium.com/view.php?id=5413
>
>
>
>
> Joao Pereira wrote:
>   
>> Can I do the encrypted trunk in SIP? Does Asterisk supports it?
>>
>> Thanks
>> Joao Pereira
>>
>> Pavel Jezek wrote:
>> 
>>> http://www.voip-info.org/wiki/view/IAX+encryption
>>>
>>>
>>>
>>> Joao Pereira wrote:
>>>   
 Hello
 I would like to define a trunk from my Asterisk to a VoIP provider, 
 but I want to make it secure, because its through the Internet.
 I want to be sure no one makes calls as being me, and that my calls 
 aren't intercepted.
 Is it possible to define encrypted trunks? And should I define the 
 trunk in SIP, IAX or something else?

 Thanks
 Joao Pereira


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Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

[Steve Prior]

Yuan LIU wrote:

The feature request # is 4542, but I don't know any associated bug 
number, nor with what phones other people had to tweak.  My phone is a 
GE 27935GE3-B. (Don't know what possessed me to say GM:-)


Yuan Liu


Just gotta ask - you did plug in the power supply connection on the
board, right?

Steve
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[asterisk-users] Web-MeetMe ready for prime time?

[Porier, Jeremy M.]

What kind of luck are people having with the Web-MeetMe control?  The
condition of the page on the voip-info wiki makes me a bit nervous about
putting Web-MeetMe into a production environment.  Use of MeetMe has
really taken off here since installation and I need a scheduling and
provisioning system for PIN numbers etc.  Are there any other solutions
out there?

Thanks,
Jeremy
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Re: [asterisk-users] IBM Server / USB Ports

[Matt]

I see that the digium card doesn't share the IRQ however Digium
has recommended diabled USB still... additionally the Digium card is
on 169 which isn't a valid IRQ.. how can I find out what it is sharing
with?

On 12/14/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:

compile kernel without usb support or unload usb modules
turby

ps
your tdm card don't share the irq, your network card share the irq...



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, December 14, 2006 5:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IBM Server / USB Ports

Hi,
I have an IBM xSeries server... and the digium card is sharing IRQ with USB
and giving me crackling audio.

>>> cat /proc/interrupts
>>>
>>> It brings up these results:
>>>0:   10566547IO-APIC-edge  timer
>>>1:  9IO-APIC-edge  i8042
>>>2:  0  XT-PIC  cascade
>>>8:  1IO-APIC-edge  rtc
>>>   12: 93IO-APIC-edge  i8042
>>>   14: 23IO-APIC-edge  ide0
>>> 121: 481478   IO-APIC-level  uhci_hcd:usb2, eth0
>>> 137:  0   IO-APIC-level  uhci_hcd:usb3
>>> 153:  0   IO-APIC-level  ehci_hcd:usb1
>>> 161:  28479   IO-APIC-level  ips
>>> 169:   10526429   IO-APIC-level  wctdm24xxp

IBM is saying that I can not disable the USB ports!Can anyone
offer any insight.. either as to how to disable the USB ports, or how to
make this work on an IBM xSeries?
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Re: [asterisk-users] how to define a secure trunk

[Pavel Jezek]

tunneling small rtp packets through vpn has big overhead,
better to use application level encryption - encrypted iax or srtp.
PJ



[EMAIL PROTECTED] wrote:

joao,
you can use ssh tunel, pptp or vpn for any sip/iax trunks or users.

turby  


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Thursday, December 14, 2006 4:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] how to define a secure trunk

as I know, only preliminary support:

0005413: [patch] Secure RTP (SRTP)
http://bugs.digium.com/view.php?id=5413




Joao Pereira wrote:
  

Can I do the encrypted trunk in SIP? Does Asterisk supports it?

Thanks
Joao Pereira

Pavel Jezek wrote:


http://www.voip-info.org/wiki/view/IAX+encryption



Joao Pereira wrote:
  

Hello
I would like to define a trunk from my Asterisk to a VoIP provider, 
but I want to make it secure, because its through the Internet.
I want to be sure no one makes calls as being me, and that my calls 
aren't intercepted.
Is it possible to define encrypted trunks? And should I define the 
trunk in SIP, IAX or something else?


Thanks
Joao Pereira


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Re: [asterisk-users] AOC-D or similar

[Mailinglisten]

Ale wrote:

hi all,

I'm trying to send text messages to Snom 300 to show the credit 
remaining during the call...


Sending a MESSAGE  directly to the phone via udp i'm able to update 
the text on the display... but not during the conversation.


I read about AOC, but i can't find any documentation about Asterisk + 
SIP + AOC


Have you any experience, docs or workaround to suggest?

Thx  Ale
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AFAIK the problem is that AOC is simply not implemented in Asterisk at 
the time. There are patches for Zap channels, but not everybody seems to 
like them.


- Fabian Foerster
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Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

[Yuan LIU]

From: Tzafrir Cohen <[EMAIL PROTECTED]>
> >Have you tried toe boost ring voltage option then recompile Zaptel?
> >It is normally set to a fairly low voltage
> >
> >John Novack
>
> Thank you so much!  I googled a bit about how to change ring voltage and
> only found an old and suspended feature request from last year that
> concerned wcfxs.c, which is now superceded by wctdm.c.  Yet the same 
method

> applies.  So I changed the value of RING_X from 0x0160 to 0x023A as
> suggested for European countries (peak of 85V according to that 
document)

> but left frequency (RING_OSC) unchanged, like the following:

To which phones does this apply? What bug number?


The feature request # is 4542, but I don't know any associated bug number, 
nor with what phones other people had to tweak.  My phone is a GE 
27935GE3-B. (Don't know what possessed me to say GM:-)


Yuan Liu


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



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RE: [asterisk-users] how to define a secure trunk

[turby]

joao,
you can use ssh tunel, pptp or vpn for any sip/iax trunks or users.

turby  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Thursday, December 14, 2006 4:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] how to define a secure trunk

as I know, only preliminary support:

0005413: [patch] Secure RTP (SRTP)
http://bugs.digium.com/view.php?id=5413




Joao Pereira wrote:
> Can I do the encrypted trunk in SIP? Does Asterisk supports it?
>
> Thanks
> Joao Pereira
>
> Pavel Jezek wrote:
>>
>> http://www.voip-info.org/wiki/view/IAX+encryption
>>
>>
>>
>> Joao Pereira wrote:
>>> Hello
>>> I would like to define a trunk from my Asterisk to a VoIP provider, 
>>> but I want to make it secure, because its through the Internet.
>>> I want to be sure no one makes calls as being me, and that my calls 
>>> aren't intercepted.
>>> Is it possible to define encrypted trunks? And should I define the 
>>> trunk in SIP, IAX or something else?
>>>
>>> Thanks
>>> Joao Pereira
>>>
>>>
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RE: [asterisk-users] IBM Server / USB Ports

[turby]

compile kernel without usb support or unload usb modules
turby 

ps
your tdm card don't share the irq, your network card share the irq...



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, December 14, 2006 5:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IBM Server / USB Ports

Hi,
I have an IBM xSeries server... and the digium card is sharing IRQ with USB
and giving me crackling audio.

>>> cat /proc/interrupts
>>>
>>> It brings up these results:
>>>0:   10566547IO-APIC-edge  timer
>>>1:  9IO-APIC-edge  i8042
>>>2:  0  XT-PIC  cascade
>>>8:  1IO-APIC-edge  rtc
>>>   12: 93IO-APIC-edge  i8042
>>>   14: 23IO-APIC-edge  ide0
>>> 121: 481478   IO-APIC-level  uhci_hcd:usb2, eth0
>>> 137:  0   IO-APIC-level  uhci_hcd:usb3
>>> 153:  0   IO-APIC-level  ehci_hcd:usb1
>>> 161:  28479   IO-APIC-level  ips
>>> 169:   10526429   IO-APIC-level  wctdm24xxp

IBM is saying that I can not disable the USB ports!Can anyone
offer any insight.. either as to how to disable the USB ports, or how to
make this work on an IBM xSeries?
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[asterisk-users] IBM Server / USB Ports

[Matt]

Hi,
I have an IBM xSeries server... and the digium card is sharing IRQ
with USB and giving me crackling audio.


cat /proc/interrupts

It brings up these results:
   0:   10566547IO-APIC-edge  timer
   1:  9IO-APIC-edge  i8042
   2:  0  XT-PIC  cascade
   8:  1IO-APIC-edge  rtc
  12: 93IO-APIC-edge  i8042
  14: 23IO-APIC-edge  ide0
121: 481478   IO-APIC-level  uhci_hcd:usb2, eth0
137:  0   IO-APIC-level  uhci_hcd:usb3
153:  0   IO-APIC-level  ehci_hcd:usb1
161:  28479   IO-APIC-level  ips
169:   10526429   IO-APIC-level  wctdm24xxp


IBM is saying that I can not disable the USB ports!Can anyone
offer any insight.. either as to how to disable the USB ports, or how
to make this work on an IBM xSeries?
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Re: [asterisk-users] how to define a secure trunk

[Pavel Jezek]

as I know, only preliminary support:

0005413: [patch] Secure RTP (SRTP)
http://bugs.digium.com/view.php?id=5413




Joao Pereira wrote:

Can I do the encrypted trunk in SIP? Does Asterisk supports it?

Thanks
Joao Pereira

Pavel Jezek wrote:


http://www.voip-info.org/wiki/view/IAX+encryption



Joao Pereira wrote:

Hello
I would like to define a trunk from my Asterisk to a VoIP provider, 
but I want to make it secure, because its through the Internet.
I want to be sure no one makes calls as being me, and that my calls 
aren't intercepted.
Is it possible to define encrypted trunks? And should I define the 
trunk in SIP, IAX or something else?


Thanks
Joao Pereira


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Re: [asterisk-users] how to define a secure trunk

[Joao Pereira]

Can I do the encrypted trunk in SIP? Does Asterisk supports it?

Thanks
Joao Pereira

Pavel Jezek wrote:


http://www.voip-info.org/wiki/view/IAX+encryption



Joao Pereira wrote:

Hello
I would like to define a trunk from my Asterisk to a VoIP provider, 
but I want to make it secure, because its through the Internet.
I want to be sure no one makes calls as being me, and that my calls 
aren't intercepted.
Is it possible to define encrypted trunks? And should I define the 
trunk in SIP, IAX or something else?


Thanks
Joao Pereira


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Re: [asterisk-users] Pickup application

[Lacy Moore - Aspendora]

On 12/13/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:


Does anyone have the pickup application working?  I'm attempting to get



I did have it working.

The problem I'm having is in the fact that my phones register with mac

addresses instead of extensions, so I'm unsure as to what to put in the
pickup app args.  I've tried mac, extension, sip device name, etc... no
luck.  Anyone have any ideas?



That's my problem as well.  My phones now register with their mac addresses,
so its been broke since I made this change.  I guess you have to pick up the
account that is ringing, not the extension.
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Re: [asterisk-users] matching the beginning of an EXTEN

[Joao Pereira]

perfect!!!
its now working this way:

exten => _.,4,GotoIf($[ "${EXTEN:0:9}" = "351217588"] ? 20:10)


Thanks a lot
Joao Pereira

Ove Aursand wrote:

Use ${EXTEN:0:9}

Regards,
Ove

Joao Pereira wrote:

Hello
how can I distinguish all the calls that arrive to my Asterisk 
starting with: 351217588XXX ?
I want match the first 9 digits does Asterisk has any function 
for this?


Thanks
Regards
Joao Pereira
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Re: [asterisk-users] (no subject)

[Dave Fullerton]

Todd- Asterisk wrote:
Hello everyone! I'm planning on setting up a new system shortly and 
can't pick the right card...  We will have 2 or 3 lines coming in and 7 
extensions (GXP2k's).  Should I just get 2 or 3 X100P cards?  Or do I 
need the Sangoma A20200 or even the A20200D (Echo cancelation)...  I was 
thinking I'd use a Dell 2.0 GHz machine as the server...  If anyone has 
suggestions as to the benifits/problems of each card choice, I'd love to 
hear it.

 thanks
  Todd


In my humble opinion, X100P's are only good for one line (and barely 
that). They don't work as well as the TDM400s do, and having more than 
one X100 card in a system is an unnecessary bombardment of interrupts. 
For a 2-3 line setup I would strongly suggest looking at a TDM400 or the 
Sangoma A200. I have used both and have been happy with both. I use a 
TDM400 at home and have managed to remove almost all echo with the use 
of fxotune and adjusting the gains. I'm using a Sangoma A200 with the 
on-board echo canceler for a phone system at work and have been very 
happy with it. The only complaint of echo on this system is on an 
occasional incoming call and only for the first second or two.


If money is tight and you are willing to tune echo out of your system by 
hand, use the TDM400. If you are willing to spend the cash and don't 
want to have to deal with constant tweaking to remove echo, get the 
A200d (and make sure you download the latest drivers from sangoma).


-Dave
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Re: [asterisk-users] PRI to SIP

[laurent schweizer]

did you use T38 with you patton smart node  2400 ?
why Patton are very good GW and fax must work.

you must also check that the clock source is the Primary and not the
internal clock...


2006/12/14, Jerry Jones <[EMAIL PROTECTED]>:

Or any of a number of gateways that do this. Off the top of my head
you can get one from CarrierAccess, Vega, Audiocodes, Mediatrix,
Adtran, and others.
Just try to be very careful as they all have their strengths and
weaknesses and you need to evaluate how they would fit your needs.
Best is to try to get an eval unit and test first - or buy with a 30
day return setup.



On Dec 14, 2006, at 6:14 AM, Joao Pereira wrote:

> For PRI you have 3 main solutions. This is the order of stability
> (and pricing):
>
> 1. Digium or Sangoma cards use the computer processor and that
> could be bad if you have huge traffic through the PRI
>
> 2. Eicon Diva cards have their own processor, which releases the PC
> processor and gives more stability
>
> 3. You can also use a dedicated router (ex: Cisco) to do that.Its
> more expensive, but more reliable.
>
> Regards
> Joao Pereira
>
>
> Patrick Fortin wrote:
>> Hi
>>
>> Can someone recommend a PRI to SIP Box that work well with asterisk
>>
>> We are presently testing with a Patton Smartnode 2400 but we are
>> unable to fax through it.
>>
>> We don't want to use digium card in a linux box for the PRI
>> connection.
>>
>> Which Cisco box would work.
>>
>> Thanks
>>
>> Patrick
>>
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Re: [asterisk-users] Ssh access over a zap channel...

[Tim Panton]

On 14 Dec 2006, at 13:32, Jordan Novak wrote:

My need to do this through asterisk is simply the ability to  
provide me access with no additional cost to my customer. It seems  
like a nice thing to include as long as authentication is done  
well. I have worked on a dozen or more types of switches and all of  
them have supported this or had the capabilty through hardware or  
licensing. I am trying to get around opening and closing the  
firewall, which at some locations is simply not accessible to me.


I'm setting a similar thing up for fall-back remote access,
I've decided to add ex-girlfriend mode to the
dialplan, so it won't get war-dialed, meaning I can only use it from  
specific phone numbers.


I've yet to test it, but the plan was:

PRI -> * -> dialplan checks callerId -> SIP-> spa 2000 -> 1200baud  
modem -> tty socket-> unix login


Main advantage is that I have all the bits in the junk box in the  
machine room,
but it also doesn't open a generic IP tunnel (unless I run PPP at  
1200 baud).


Tim.





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Re: [asterisk-users] PRI to SIP

[Jerry Jones]

Or any of a number of gateways that do this. Off the top of my head  
you can get one from CarrierAccess, Vega, Audiocodes, Mediatrix,  
Adtran, and others.
Just try to be very careful as they all have their strengths and  
weaknesses and you need to evaluate how they would fit your needs.  
Best is to try to get an eval unit and test first - or buy with a 30  
day return setup.




On Dec 14, 2006, at 6:14 AM, Joao Pereira wrote:

For PRI you have 3 main solutions. This is the order of stability  
(and pricing):


1. Digium or Sangoma cards use the computer processor and that  
could be bad if you have huge traffic through the PRI


2. Eicon Diva cards have their own processor, which releases the PC  
processor and gives more stability


3. You can also use a dedicated router (ex: Cisco) to do that.Its  
more expensive, but more reliable.


Regards
Joao Pereira


Patrick Fortin wrote:

Hi

Can someone recommend a PRI to SIP Box that work well with asterisk

We are presently testing with a Patton Smartnode 2400 but we are  
unable to fax through it.


We don't want to use digium card in a linux box for the PRI  
connection.


Which Cisco box would work.

Thanks

Patrick

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Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

[Tzafrir Cohen]

On Thu, Dec 14, 2006 at 07:15:30AM -0600, Russ Price wrote:
> Yuan LIU wrote:
> >A configuration string "boostringer" was mentioned in several messages, 
> >including one concerning TDM400P, all without indicating the applicable 
> >configuration file.  This has no apparent effect on TDM400P wherever I 
> >tried.
> 
> That would go in your /etc/modprobe.conf which controls module loading.
> 
> On my CentOS 4.4 system:
> 
> install wctdm /sbin/modprobe --ignore-install wctdm fxshonormode=1 
> boostringer=1 && /sbin/ztcfg

My usual comment:

options wctdm fxshonormode=1 boostringer=1

The above is pure voodoo which attempts working around the need for a
proper init.d script. Dump it.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Ssh access over a zap channel...

[Jordan Novak]

My need to do this through asterisk is simply the ability to provide me
access with no additional cost to my customer. It seems like a nice
thing to include as long as authentication is done well. I have worked
on a dozen or more types of switches and all of them have supported this
or had the capabilty through hardware or licensing. I am trying to get
around opening and closing the firewall, which at some locations is
simply not accessible to me.
 
Jordan Novak
 
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[asterisk-users] agi scripts running slowly

[Richard Smith]

Hi all,

I recently installed asterisk 1.2.4 on a HP DL140 G2 server and co-located it. 
My only problem with the box is that there
is a noticeable delay in the processing of agi scripts compared to any other 
install of asterisk I have.

Has anyone got any ideas why this is happening and any guide to tweaking the 
agi to run faster?


Thanks guy for your help in advance.
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[asterisk-users] (no subject)

[Todd- Asterisk]

Hello everyone! I'm planning on setting up a new system shortly and  
can't pick the right card...  We will have 2 or 3 lines coming in and  
7 extensions (GXP2k's).  Should I just get 2 or 3 X100P cards?  Or do  
I need the Sangoma A20200 or even the A20200D (Echo cancelation)...   
I was thinking I'd use a Dell 2.0 GHz machine as the server...  If  
anyone has suggestions as to the benifits/problems of each card  
choice, I'd love to hear it.

 thanks
  Todd
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Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

[Russ Price]

Yuan LIU wrote:
A configuration string "boostringer" was mentioned in several messages, 
including one concerning TDM400P, all without indicating the applicable 
configuration file.  This has no apparent effect on TDM400P wherever I 
tried.


That would go in your /etc/modprobe.conf which controls module loading.

On my CentOS 4.4 system:

install wctdm /sbin/modprobe --ignore-install wctdm fxshonormode=1 
boostringer=1 && /sbin/ztcfg


Russ
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[asterisk-users] Zaptel under FC6

[Rudolf Ladyzhenskii]

Hi, all

I am building a new server. Have installed FC 6 and put in TDM400 card.

Checked out latest asteriusk code, run make install in zaptel directory.
So far all is fine.

Now I am trying to install the drivers.

# modprobe zaptel
FATAL: Module zaptel not found.

Fair enough, no zaptel driver is found on the system.

Is there are any known problems with FC6? I did not have much trouble
running on FC3 before.

Thanks,
Rudolf
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Re: [asterisk-users] MeetMe Conferencing and Marked Mode

[RR]

On 12/14/06, Tobias Wolf <[EMAIL PROTECTED]> wrote:

Hmmm, there is really not much to share. Most of the code handles
Authentication or other stuff, like informing another server that a new
user has entered an conf-room, or updating databases.

Mostly I look an the CallerId to decide if this should be a "marked
user" (but there are not many scenarios ther the CallerID is known in
advance), or the Caller has to make the choice by himself, by touching
the right button.

I use Asterisk-Java as AGI-Implementation.

Everytime then an AGI-Script gets executed by an Extensions the
service-Method of my Java-Class will be executed:

   private String adminMeetme = "dMaXq"; // not the 'a' for the marked Mode

   private String userMeetme = "dMXq";

   public void service(AGIRequest req, AGIChannel channel) throws
AGIException {
   // Ask if we want to be marked
   char option = getOption(wantAdminPrompt, "12", 360 * 1000);

   if (option == '1') {
   // Yes

   // This sets a Channel-Variable with the
   // correct MeetMe-Parameters
   exec("SET", "MEETMEOPTS=" + adminMeetme);
   } else {
   // No
   // Parameter for a normal user
   exec("SET", "MEETMEOPTS=" + userMeetme);
   }
   }

My dialplan looks like this:

exten => 1000,1,AGI(agi://localhost/askformarked.agi)
exten => 1000,n,MeetMe(${EXTEN},${MEETMEOPTS})
exten => 1000,n,Hangup()

This is a minimalistic Example, i have erased a lot of logic that has
little to do with the actual MeetMe-Room. But it is the essence of
dealing with the correct Parameters, there a lot of other way to
accomplish this. It depends on what you have in mind with your
application which way works for you.

So, you see there is no ready-to-use-multi-purpose-AGI which you can
simply plug-in to your Asterisk, sorry for that ;)

But i think the effort as not that great, even if you solve it only with
Dialplan-Logic, or AEL.

I am sure, you will come up with an solution :D

Take care,

Tobias Wolf
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Thanks Tobias,

That does help a lot actually, not least of all being saving me heaps
of time in trying to mess around with my dialplan thinking I didn't
know how to differentiate between marked and unmarked user. Maybe I'm
not thinking about all scenarios here but the logical thing (as per my
logic of course) would be to auto-set the user entering the admin
password as the "marked user". The reason I say this is that let's say
you have a conference you've setup with a potential client, now you're
the host, but if he's the "marked" user & he's the one without whom
the conference is pointless. So until he arrives, no one gets to talk
(I think that's how it works?). But if using what I'm saying, then the
"host" is also the "marked" user and as long as he/she's there, other
people can simply talk to each other and just wait for the potential
client to arrive. But if the "host" isn't there, then there's no one
to control/manage the conference hence all non-admin users should
simply stay in a holding pattern listening to MOH.

But I guess this discussion is only useful if the "dev" people are
reading this and they agree. Maybe I'm missing something, I don't
know.

Thanks for the AGI structure though, I had implemented this via
dial-plan except then it only works for a few static conferences with
static PINs. Our conferences reset conference PINs to random digits
every night and unless I do SQL queries (since I read meetme.conf from
the DB) I have no way of knowing what those PINs are and so can't
create DialPlan rules to check for the marked or unmarked user based
on the PIN. If I use your method, then I'll have to prompt the user if
they want to become marked or not. I don't want to offer the option.

But like you said, I'll figure out a solution (although I think I
already have while typing this) but something tells me, it'll be
difficult and messy without an AGI :)

Cheers
\R
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RE: [asterisk-users] matching the beginning of an EXTEN

[sandeep kalra]

Try
Exten => _351217588XXX, 1, Dial ( ... )

Thanks and Regards
--Sandeep Kalra

Ph: +91-120-4342000-X-2966
: +91-120-4342966 (direct)
M- 9810683168

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent: Thursday, December 14, 2006 5:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] matching the beginning of an EXTEN

Hello
how can I distinguish all the calls that arrive to my Asterisk starting
with: 351217588XXX ?
I want match the first 9 digits does Asterisk has any function for this?

Thanks
Regards
Joao Pereira
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Re: [Asterisk-Users] Siemens Gigaset SL75

[Pavel Jezek]

I think, Nokia E60/61/70 currently supports 802.1x



Joao Pereira wrote:
Do you know if it has 802.1x authentication as it is defined in 
EDUroam (  http://www.eduroam.org/ )   ?
I never found a WiFi phone working with 802.1x  I tested ZyXel 
Prestige 2000 but the sound was bad and it doesnt support 802.1x :(


Thanks
Joao Pereira


[EMAIL PROTECTED] wrote:
No, the Gigaset is the only WLAN phone I tested so long, so I can not 
compare it to the other phones you mentioned.


-Original Message-
*From:* Olivier [mailto:[EMAIL PROTECTED]
*Sent:* Friday, November 24, 2006 10:19 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] Siemens Gigaset SL75

Have you ever compared it to Linksys WIP 330 or Zyxel 2000 ?
Those 2 seem to get average reviews from users (short range,
autonomy, ...).
On paper, it seems to me a decent WiFi phone is still lacking today.

Maybe this Gigaset SL75 could fill the void.



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Re: [asterisk-users] PRI to SIP

[Joao Pereira]

For PRI you have 3 main solutions. This is the order of stability (and 
pricing):


1. Digium or Sangoma cards use the computer processor and that could be 
bad if you have huge traffic through the PRI


2. Eicon Diva cards have their own processor, which releases the PC 
processor and gives more stability


3. You can also use a dedicated router (ex: Cisco) to do that.Its more 
expensive, but more reliable.


Regards
Joao Pereira


Patrick Fortin wrote:

Hi

Can someone recommend a PRI to SIP Box that work well with asterisk

We are presently testing with a Patton Smartnode 2400 but we are 
unable to fax through it.


We don't want to use digium card in a linux box for the PRI connection.

Which Cisco box would work.

Thanks

Patrick

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Re: [asterisk-users] matching the beginning of an EXTEN

[Ove Aursand]

Use ${EXTEN:0:9}

Regards,
Ove

Joao Pereira wrote:

Hello
how can I distinguish all the calls that arrive to my Asterisk 
starting with: 351217588XXX ?
I want match the first 9 digits does Asterisk has any function for 
this?


Thanks
Regards
Joao Pereira
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[asterisk-users] matching the beginning of an EXTEN

[Joao Pereira]

Hello
how can I distinguish all the calls that arrive to my Asterisk starting 
with: 351217588XXX ?

I want match the first 9 digits does Asterisk has any function for this?

Thanks
Regards
Joao Pereira
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Re: [Asterisk-Users] Siemens Gigaset SL75

[Joao Pereira]

Do you know if it has 802.1x authentication as it is defined in EDUroam 
(  http://www.eduroam.org/ )   ?
I never found a WiFi phone working with 802.1x  
I tested ZyXel Prestige 2000 but the sound was bad and it doesnt support 
802.1x :(


Thanks
Joao Pereira


[EMAIL PROTECTED] wrote:
No, the Gigaset is the only WLAN phone I tested so long, so I can not 
compare it to the other phones you mentioned.


-Original Message-
*From:* Olivier [mailto:[EMAIL PROTECTED]
*Sent:* Friday, November 24, 2006 10:19 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] Siemens Gigaset SL75

Have you ever compared it to Linksys WIP 330 or Zyxel 2000 ?
Those 2 seem to get average reviews from users (short range,
autonomy, ...).
On paper, it seems to me a decent WiFi phone is still lacking today.

Maybe this Gigaset SL75 could fill the void.



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Re: [asterisk-users] Measuring VoIP latency and packet loss

[Chris Mason (Lists)]

Mochamad Susantok wrote:

I have already use smokeping, and great for measure latency and packet
loss, but not voip packet especialy, or you has been modified smokeping ?
  
I have not modified it. I take it that if the network has considerable 
latency, so will VOIP. It has been my experience that smokeping gives a 
pretty good representation of how call quality will be.


--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 



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Re: [asterisk-users] send fax by Iaxmodem ?

[Marco Mouta]

Hi all,

First let me say thank you for Lee Howard, you definitely  found my problem
on sending faxes!

I'm using hy-email2fax to send faxes, and i notice that is there the problem
is starting, as the subject of my .eml file contains only the phone number
but then some how hy-email2fax is not detecting the ^M at the end of
subject.

I'm not an used with sed so it will take me some time to understand how it
works and correct this.

By the way i notice a small isse "bug" on hy-email2fax available for
download on
http://wpkg.org/email2fax/files/hy-email2fax

--

#
# Make a temporary file out of an email #
#

cat |  sed 's/
$//'>$EMAILFILE

--

I only got Hy-email2fax working fine after changing this to:

cat |  sed 's/$//'>$EMAILFILE


Any ways this is working fine only for tests, because as i mentioned above
"I'm in troubles with a carriage return on email subject", to test
Sucessfully this hy-email2fax i hardcoded the variable NEWNUMBER="" to
my dialed number.

After this Hands on I can sucessfully send faxes with Hy-email2fax -->
Hylafax--->asterisk Sucessfully.

But as i mentioned before i need to get ride of ^M on the subject line.

Any one can help me on this?

Best regards,
Marco Mouta


On 12/13/06, Lee Howard <[EMAIL PROTECTED]> wrote:


Marco Mouta wrote:

> Dec 13 11:28:07.51: [ 9242]: DIAL 2079^M
> Dec 13 11:28:07.51: [ 9242]: <-- [9:ATDT2079\r]
> Dec 13 11:28:16.70: [ 9242]: --> [4:BUSY]
> Dec 13 11:28:46.70: [ 9242]: MODEM TIMEOUT: reading line from modem
> Dec 13 11:28:46.71: [ 9242]: MODEM 
> Dec 13 11:28:46.71: [ 9242]: SEND FAILED: JOB 1 DEST 2079^M ERR
> Unknown problem


In your case BUSY means exactly that, and you should take a look at the
Asterisk CLI to get more information as to what "busy" really means.

However, your dialstring terminated by a carriage return ( or ^M) is
problematic, too, because it essentially instructs HylaFAX to ignore all
responses after ATDT2079 except for "OK" and then proceed from there.
Basically you just need to get rid of that terminating carriage return.

Lee.
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Re: [asterisk-users] MeetMe Conferencing and Marked Mode

[Tobias Wolf]

RR schrieb:

> And what would someone have to do to sweet-talk you into sharing this
> AGI ;)

Hmmm, there is really not much to share. Most of the code handles
Authentication or other stuff, like informing another server that a new
user has entered an conf-room, or updating databases.

Mostly I look an the CallerId to decide if this should be a "marked
user" (but there are not many scenarios ther the CallerID is known in
advance), or the Caller has to make the choice by himself, by touching
the right button.

I use Asterisk-Java as AGI-Implementation.

Everytime then an AGI-Script gets executed by an Extensions the
service-Method of my Java-Class will be executed:

private String adminMeetme = "dMaXq"; // not the 'a' for the marked Mode

private String userMeetme = "dMXq";

public void service(AGIRequest req, AGIChannel channel) throws
AGIException {
// Ask if we want to be marked
char option = getOption(wantAdminPrompt, "12", 360 * 1000);

if (option == '1') {
// Yes

// This sets a Channel-Variable with the
// correct MeetMe-Parameters
exec("SET", "MEETMEOPTS=" + adminMeetme);
} else {
// No
// Parameter for a normal user
exec("SET", "MEETMEOPTS=" + userMeetme);
}
}

My dialplan looks like this:

exten => 1000,1,AGI(agi://localhost/askformarked.agi)
exten => 1000,n,MeetMe(${EXTEN},${MEETMEOPTS})
exten => 1000,n,Hangup()

This is a minimalistic Example, i have erased a lot of logic that has
little to do with the actual MeetMe-Room. But it is the essence of
dealing with the correct Parameters, there a lot of other way to
accomplish this. It depends on what you have in mind with your
application which way works for you.

So, you see there is no ready-to-use-multi-purpose-AGI which you can
simply plug-in to your Asterisk, sorry for that ;)

But i think the effort as not that great, even if you solve it only with
Dialplan-Logic, or AEL.

I am sure, you will come up with an solution :D

Take care,

Tobias Wolf
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[asterisk-users] AOC-D or similar

[Ale]

hi all,

I'm trying to send text messages to Snom 300 to show the credit 
remaining during the call...


Sending a MESSAGE  directly to the phone via udp i'm able to update the 
text on the display... but not during the conversation.


I read about AOC, but i can't find any documentation about Asterisk + 
SIP + AOC


Have you any experience, docs or workaround to suggest?

Thx  Ale
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[asterisk-users] WRAP+astlinux g729

[Jon Schøpzinsky]

Hello

How many simultaneous conversations g.729a should one expect with a WRAP board 
running Asterisk?
Has anybody tried this?

Kind Regards

Jon Leren Schøpzinsky

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[asterisk-users] RE: Extending Avaya IP Office ISDN30e with Asterisk

[Russell Brown]

>Has anyone hooked up * as an extension/trunk of an Avaya system that has
>around 2 ISDN30e's.

I'm currently running an Asterisk box between my ISDN30 PRI and my
Argent Office (pre Avaya takeover of Network Alchemy but still the same
box as the Avaya IP Office).

All it took was a two PRI digium card and a PRI crossover cable between
the Asterisk box and the Argent.  You'd need a four port card of course.

All seems to work well.  I did have some issues with the Argent
rejecting data calls when the DDI wasn't set to receive them but relied
on the default 'route all DATA type calls to this service' in the
Argent.  I cured this by setting an explicit incoming call route for
these DATA calls on the Argent and it worked.

-- 
 Regards,
 Russell
 
| Russell Brown  | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com  |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk |
 
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Re: [asterisk-users] MeetMe Conferencing and Marked Mode

[RR]

On 12/14/06, Tobias Wolf <[EMAIL PROTECTED]> wrote:

Actually you don't need 2 different extension, but two different
parameter-sets for the meetme-App. So, you have to implement some logic
that detects, if the calling user has to be marked or not. It's your
choice if you do this by dialplan logic or by AGI, or something else.

The second PIN, which you can define in meetme.conf, is not for the
marked mode, but for the admin mode. This gives the user some control
over the conf.

example:

exten => s,1,playback("choose one for marked mode or two for normal mode")

exten => 1,1,meetme(100,a)
exten => 2,1,meetme(100,w)

(Please note that the above is not an working dialplan ;-) )

Personally, I use an AGI for my Conferencing-Apps and let it generate
the correct Parameters for the meetme App.

Cheers,

Tobias


And what would someone have to do to sweet-talk you into sharing this AGI ;)
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Re: [asterisk-users] How to temporarily unload modules.

[Angel Heart]

Thanks Tzafrir and Marco for the info.

If I want to unload modules during start-up, I have to edit my 
/etc/asterisk/mudules.conf and add something like;

noload => app_test.so

or I can unload them immediately at CLI using Mr. Cohen suggestion.

Regards.



> /etc/asterisk/modules.conf

>Marco


- Original Message 
From: Tzafrir Cohen <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Wednesday, December 13, 2006 8:54:04 PM
Subject: Re: [asterisk-users] How to temporarily unload modules.


On Wed, Dec 13, 2006 at 02:03:09AM -0800, Angel Heart wrote:
> Hi,
> 
> In what Asterisk file can I edit for me to temporarily unload such 
> modules. But later I woudl still be able to load them.

Works fine as long as the module is not in use.

  asterisk -rx 'unload app_test.so'

Later on:

  asterisk -rx 'load app_test.so'

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Do you Yahoo!?
Everyone is raving about the all-new Yahoo! Mail beta.
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[asterisk-users] IAX trunk problem

[Lee Archer]

> I wonder if anyone can help me with this.  I have 4 sites running
> Asterisk and these are linked via IAX trunks and ADSL lines.  Calls
> coming into any of these sites are received locally and forwarded to a
> central operator.  E.g.  Call comes in on site A and is forwarded to
> the operator on site B.  99 out of 100 times the operator will send
> the call back to someone at the site from where it came but site B's
> Asterisk server seems to be staying in the loop.  E.g. A > B > A.
> I've had a look and can't see anything obvious as I had assumed that
> Asterisk would pass the call off.  I've tried notransfer on the trunks
> but site B's Asterisk server doesn't seem to be joining the endpoints
> and staying in the loop and therefore the call is going over the
> trunks twice.
> 
> Thanks
> 
> Lee
###

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Re: [asterisk-users] MeetMe Conferencing and Marked Mode

[Tobias Wolf]

Savoy, Kevin - Williston, ND schrieb:
> I'll give this a try but seems silly to require 2 different extensions
> for one conference room. Thanks for the input.
> 
Actually you don't need 2 different extension, but two different
parameter-sets for the meetme-App. So, you have to implement some logic
that detects, if the calling user has to be marked or not. It's your
choice if you do this by dialplan logic or by AGI, or something else.

The second PIN, which you can define in meetme.conf, is not for the
marked mode, but for the admin mode. This gives the user some control
over the conf.

example:

exten => s,1,playback("choose one for marked mode or two for normal mode")

exten => 1,1,meetme(100,a)
exten => 2,1,meetme(100,w)

(Please note that the above is not an working dialplan ;-) )

Personally, I use an AGI for my Conferencing-Apps and let it generate
the correct Parameters for the meetme App.

Cheers,

Tobias
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Re: [asterisk-users] Stress test

[Andrew Joakimsen]

http://sipp.sourceforge.net/
SIPp is a free Open Source test tool / traffic generator for the SIP
protocol

On 12/13/06, Andre Luiz Martins Rodrigues <[EMAIL PROTECTED]> wrote:


Hello peoples,


I need to do a test of urgent stress.  It know as much as connections
simultaneous my equipment is going to do passing codec g729 and g723.
Someone knows say me as obtain does him?


Andre Luiz Martins
mailto:[EMAIL PROTECTED]

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[asterisk-users] Hardware TDM Switching

[asterisk]

Hello,

Do anybody know, if there is a way to connect 2 zap-channels with Hardware 
TDM Switching?


Thanks

Nico

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Re: [asterisk-users] Problem in making outbound calls in PRI

[Danny]

Hey everyone !

This config worked !

; zapata.conf
[channels]
language=en
context=from-pstn
switchtype=euroisdn
pridialplan=local
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=0
callgroup=1
pickupgroup=1
channel => 1-15,17-31


Thank you !

- Danny


Doug Lytle wrote:

Danny wrote:



Is there anybody who can help me out on this ?

I am pretty much lost in forums and docs, and I m getting nowhere.


Danny,

You've got a lot of stuff in there that isn't used for a PRI/ISDN.  
Mine setup attached.  This if for a T1, and in the US.  Please make 
adjustments for your area:


[zaptel.conf]

span=1,1,0,esf,b8zs
defaultzone=us
loadzone=us
bchan=1-23
dchan=24

span=2,0,0,esf,b8zs
fxsks=25-32
fxoks=33-48
defaultzone=us
loadzone=us

[zapata.conf]

[channels]

musiconhold=tape
switchtype=national
context=pri
signalling=pri_cpe
group=1
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
rxgain=-1.0
txgain=-4.0
busydetect=no
callprogress=no
pridialplan=unknown
usercallerid=yes
callerid=asreceived
channel => 1-23




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