Re: [asterisk-users] common/shared voicemail box

[Rob Hillis]

The only possible way I can think of achieving this would be to mangle
the incoming caller ID to include the extension that the call came
from.  Given that Asterisk's voicemail boxes are separate to extensions,
I can't see another solution.

Benjamin Jacob wrote:
> Hello All,
>
> I am using ODBC storage for voicemail on my asterisk box. I want to have 
> a common voicemail box for different extensions.
> I know how to do that, but the question troubling me is how and where do 
> I store the the extension name for which a particular voicemail was left.
> e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 5.
> Now, when someone calls 1000, and leaves a voicemail, I want to store 
> the fact that this voicemail was meant for extension 1000.
> Similarly for 1001 and so on.
>
> Any ideas anyone?
>
> TiA
> - Benjamin Jacob.
>
>
>
>
>
>
>
>
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[asterisk-users] Phones Not Registering

[Edwin Kariuki]

Hi,

I have a voip platform that has a SIP server where about 450 sipura phones & 
adaptors register. On two occassions some phones (which were previously 
working) have refused to register with certain IPs but when I change the IP the 
phones register. The failing IP can the work after two days.

A trace from the server shows that the phone is sending a registration signal 
to the server & that the server is also sending back the same but its not 
getting to the phone.

What could be the cause of this?

Thanks,

Edwin

   
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Re: [asterisk-users] Phones Not Registering

[Benjamin Jacob]

The reason could be bad routing, IPs used by multiple devices.. n so on...



Edwin Kariuki wrote:

> Hi,
>
> I have a voip platform that has a SIP server where about 450 sipura 
> phones & adaptors register. On two occassions some phones (which were 
> previously working) have refused to register with certain IPs but when 
> I change the IP the phones register. The failing IP can the work after 
> two days.
>
> A trace from the server shows that the phone is sending a registration 
> signal to the server & that the server is also sending back the same 
> but its not getting to the phone.
>
> What could be the cause of this?
>
> Thanks,
>
> Edwin
>
> 
> Get easy, one-click access to your favorites. Make Yahoo! your 
> homepage. 
>
>
>
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information in this message that do not relate to official business of Mascon 
shall be understood to be neither given nor endorsed by Mascon. Any information 
contained in this email, when addressed to Mascon clients is subject to the 
terms and conditions in governing client contract.

Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we 
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Re: [asterisk-users] Phones Not Registering

[Brett Crapser]

On Thu, 22 Nov 2007, Edwin Kariuki wrote:
> Hi,
>
> I have a voip platform that has a SIP server where about 450 sipura 
> phones & adaptors register. On two occassions some phones (which were 
> previously working) have refused to register with certain IPs but when I 
> change the IP the phones register. The failing IP can the work after two 
> days.
>
> A trace from the server shows that the phone is sending a registration 
> signal to the server & that the server is also sending back the same but 
> its not getting to the phone.
>
> What could be the cause of this?

What kind of voip platform?
What SIP server?
Out over the Internet or local lan or a VPN?

Going to need a lot more information...

Brett

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Re: [asterisk-users] ACD functionality , Skills for agents

[Kyriakos]

What about tagging calls with skills and then putting them all in one queue?
Skill for agents would be declared in queues.conf just like penalties?
member => Agent/1001,Sales,10
where Sales=skilland 10 = weight of skill for agent.

Is that feasible?







-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Wednesday, November 21, 2007 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ACD functionality , Skills for agents

Kyriakos wrote:
> It would be nice to add an option of choosing to answer the call with the
> longest waiting time, or answer randomly, or round robin, etc...
>
>
>   
  Agreed, but, understand that each queue defined in app_queue is separate.
The way the weights work is only by instructing a thread to go into another
queue's data space (while holding a mutex lock to make sure  multiple
threads aren't walking on the same space) and make sure there aren't calls
waiting where that queue has a higher weight than the one currently
processing before it decides whether or not it can serve up calls to an
available member. There is not one large, consolidated, pool of calls
waiting for consideration when you are dealing with multiple queues in the
current design of app_queue. As a result, true skills based routing with the
existing app_queue is, difficult, at best. 

  The queue application does a fairly good job for what most people need for
it to do, but when you start getting into these more complex call/queue
routing scenarios, you're defining a scope of requirements that the original
app_queue just wasn't designed for. Features like queue weight were/are band
aids to try to get you closer to the end run goal, but that band aid and
others like it has come with its own costs as well (mutex deadlocks,etc)
that many people here have complained about in the past.

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




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Re: [asterisk-users] Phones Not Registering

[Edwin Kariuki]

Brett Crapser <[EMAIL PROTECTED]> wrote: 
On Thu, 22 Nov 2007, Edwin Kariuki wrote:
> Hi,
>
> I have a voip platform that has a SIP server where about 450 sipura 
> phones & adaptors register. On two occassions some phones (which were 
> previously working) have refused to register with certain IPs but when I 
> change the IP the phones register. The failing IP can the work after two 
> days.
>
> A trace from the server shows that the phone is sending a registration 
> signal to the server & that the server is also sending back the same but 
> its not getting to the phone.
>
> What could be the cause of this?

What kind of voip platform?
What SIP server?
Out over the Internet or local lan or a VPN?

Going to need a lot more information...

Brett

The platform run on Linux & asterisk.
Devices register over the internet.


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Re: [asterisk-users] ACD functionality , Skills for agents

[Jan-Hendrik Palic]

Hi,

Örn Arnarson schrieb:
> I have often wondered the same thing.
> 
> It seems to me to be random, or it works it out some way I am not
> familiar with. I have seen calls with wait time of 30 seconds get
> answered before calls with 30 minutes wait time from queues with equal
> weight.

I can confirm it with asterisk 1.4.11 on debian/lenny. We tested it like
that:

- two queues with the same weight are configures
- 2 agents are logged in both queues with strategy "last recent"
- 5 test callers are in the queue
- 1st caller gets an agent, the phone is ringing and we let time it out
- the callers after him in the queue get agent but the first caller
  never

But, as said, it happens randomly and is not easy to reproduce.

But we do not have any solutions for that beside to use only one queue.

Best Regards.

Jan

-- 
Jan-Hendrik Palic
TNG - NETWORK MANAGEMENT GmbH
Projensdorfer Str. 324, D-24106 Kiel, Germany
mailto:[EMAIL PROTECTED] http://www.tng.de


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Re: [asterisk-users] spandsp as T.38 termination?

[Robert Moskowitz]

[EMAIL PROTECTED] wrote:
> You need a T38 gateway of sorts, sort of like the app_t38gateway of 
> CallWeaver.
>   

the app_rxfax and app_txfax applications in 0.0.4 of spandsp won't provide it?  
It seems, functionally, that with spandsp supporting T.38, simple dialplans 
where you call a SIP extension, and it kicks off rxfax (I can provide mine 
here) should work just fine with T.38 as they do with T.30 over RTP.

> However digium refuses to include such a program with Asterisk.
>
> On Nov 21, 2007 6:13 PM, Robert Moskowitz <[EMAIL PROTECTED]> wrote:
>   
>> It seems that Spandsp has everything in it (when you include rxfax and
>> txfax) to be a T.38 termination when used with Asterisk 1.4?
>>
>> And if so, what version of Spandsp?
>>
>> What version of IAXModem (so I don't have to also deal with T38Modem)?
>>
>>
>>
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[asterisk-users] Digium and Asterisk

[bilal ghayyad]

Hi List;

Is Digium the best telephony cards to be used with
Asterisk? The prices are some how high, any
suggestion?

Regards
Bilal


  

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Re: [asterisk-users] Queue Drops to Voicemail

[Lenz]

I believe you should define the agents in agents.conf as well! :-)
l.

On Wed, 21 Nov 2007 19:39:46 +0100, Gregory Malsack <[EMAIL PROTECTED]>  
wrote:

> Hello All,
>
>
> I am hoping someone out there can enlighten me on this issue. I am using
> asterisk 1.4.11. We have a call queue setup, and our agents log into the
> queue. As long as no one is on the phone the queue works properly.
> However, when there are agents on the phone, the queue will erratically
> drop calls to the queue.
>
>
> Any help will be extremely appreciated, and I will provide any conf
> files you may require. I have included excerpts of the config files I
> think you may need.
>
>
> Sincerely,
>
> Gregory Malsack
>
>
> Incoming line in extensions.conf
>
> exten => 8582294,1,answer()
>
> exten => 8582294,n,goto(csr|s|1)
>
>
> CSR context in extensions.conf
>
> [csr]
>
> include => default
>
> exten => s,1,answer()
>
> exten => s,n,Set(CDR(accountcode)=800)
>
> exten => s,n,Queue(802|n|||30)
>
> exten => s,n,background(csr)
>
> exten => s,n,queue(800)
>
>
> agents listed in users.conf (agent  logs into extension 111
> (normally))
>
> []
>
> callwaiting = no
>
> fullname = Agent 1
>
> hasagent = yes
>
> hasdirectory = no
>
> hasiax = no
>
> hasmanager = no
>
> hassip = no
>
> hasvoicemail = no
>
> host = dynamic
>
> mailbox = 
>
> secret = 1234
>
> threewaycalling = no
>
> registeriax = no
>
> registersip = no
>
> canreinvite = no
>
> nat = no
>
> dtmfmode = rfc2833
>
> disallow = all
>
> allow = all
>
>
> [111]
>
> callwaiting = no
>
> fullname = Conference Room
>
> hasagent = no
>
> hasdirectory = no
>
> hasiax = no
>
> hasmanager = no
>
> hassip = yes
>
> hasvoicemail = yes
>
> host = dynamic
>
> mailbox = 111
>
> secret = 111
>
> threewaycalling = no
>
> vmsecret = 111
>
> registeriax = no
>
> registersip = yes
>
> canreinvite = no
>
> nat = no
>
> dtmfmode = rfc2833
>
> disallow = all
>
> allow = all
>
>
> All directives in agents.conf are remarked out.
>
>
> queues.conf
>
> [800]
>
> fullname = CSR Agent Queue
>
> strategy = rrmemory
>
> timeout = 8
>
> wrapuptime = 20
>
> autofill = yes
>
> autopause = no
>
> maxlen =
>
> joinempty = yes
>
> leavewhenempty = no
>
> reportholdtime = no
>
> musicclass = csr
>
> member = Agent/
>
> member = Agent/1112
>
> member = Agent/1113
>
> member = Agent/1114
>
> member = Agent/1110
>
> member = Agent/1107
>
> member = Agent/1138
>
> member = Agent/1118
>
> member = Agent/1149
>



-- 
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http://queuemetrics.com

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Re: [asterisk-users] [asterisk-dev] trunk working under windows!

[Max McGraw]

 Drew Gibson wrote:

> but ... why?

 so  windows  lawyers can sneak a few patents thru the patent office
 and sue Digium for patent infringement.

 I am not criticizing Zoa or Luigi here, just reflecting on what ends up
 happening eventually.

 Think  BSD code into windows, think file & receive a few patents for
 stolen ideas, think sue linux & open source for patent infringement.

 In my humble opinion.

 I applaud the technical merit of the effort to port things to windows,
 but please remember that you are aiding and abetting the enemy.

 Obviously, it it your time and your dime to do with as you please...
 but you may end up biting the hand that feeds you.


> > Zoa wrote:
> >
> >> Cool, i'll help out a bit with the windows port,  i will start right
> >> away with a new project on asteriskguru making nightly executable builds
> >> and installers - will post the links in -users when i'm done.
> >>
> >> Well done luigi, this will make it a lot easier for a lot of non linux
> >> guys to make their first steps in the asterisk world
> >>
> >> Crossposted to -users.
> >>
> >> Zoa
> >>
> >> Luigi Rizzo wrote:
> >>
> >>
> >>> As a result of the commit below, now trunk can be built and run under
> >>> Windows/cygwin, including the building of modules.
> >>>
> >>>
==

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[asterisk-users] mailbox name length

[Tomasz Zieleniewski]

Hi,

is there a way to set the length of the mailbox name - now it is 4

Cheers

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Re: [asterisk-users] Phones Not Registering

[Gordon Henderson]

On Thu, 22 Nov 2007, Edwin Kariuki wrote:

>
>
> Brett Crapser <[EMAIL PROTECTED]> wrote:
> On Thu, 22 Nov 2007, Edwin Kariuki wrote:
>> Hi,
>>
>> I have a voip platform that has a SIP server where about 450 sipura
>> phones & adaptors register. On two occassions some phones (which were
>> previously working) have refused to register with certain IPs but when I
>> change the IP the phones register. The failing IP can the work after two
>> days.
>>
>> A trace from the server shows that the phone is sending a registration
>> signal to the server & that the server is also sending back the same but
>> its not getting to the phone.
>>
>> What could be the cause of this?
>
> What kind of voip platform?
> What SIP server?
> Out over the Internet or local lan or a VPN?
>
> Going to need a lot more information...
>
> Brett
>
> The platform run on Linux & asterisk.
> Devices register over the internet.

Some random thoughts:

Too many NAT sessions for the router to track?
Running out of ARP table space on the Linux box if not going via NAT?

Gordon

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[asterisk-users] Toll fraud detection/password script

[J. Oquendo]

So I was bored yesterday and tried solving a few
problems with one stone:

1) Notify me of potential brute forcers (multiple attempts
to register multiple numbers from one address)
2) Notify me of (l)users who are having password issues

So I whipped up a simple script to run in cron and
notify me that UserX from X_IP_Space had X amout of
password issues. I'm currently running this from
cron and it works fine. My personal version is
modified to block (l)users after 10 failures on
2 separate accounts or 50 failures on one account.

Methodology is, if someone hasn't complained within
two minutes of something happening to their phones
that they can't log in, then they won't need to
use that phone right now. Let them call in and
complain...

http://www.infiltrated.net/scripts/astrap

-- 
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA #579 (FW+VPN v4.1)
SGFE #574 (FW+VPN v4.1)

echo c2lsQGluZmlsdHJhdGVkLm5ldAo=|\
python -c "import sys; print sys.stdin.read().decode('base64')"

http://pgp.mit.edu:11371/pks/lookup?op=get&search=0xF684C42E


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Re: [asterisk-users] Toll fraud detection/password script

[Rony Ron]

Thanks for your contrib

On Nov 22, 2007 2:56 PM, J. Oquendo <[EMAIL PROTECTED]> wrote:
>
> So I was bored yesterday and tried solving a few
> problems with one stone:
>
> 1) Notify me of potential brute forcers (multiple attempts
> to register multiple numbers from one address)
> 2) Notify me of (l)users who are having password issues
>
> So I whipped up a simple script to run in cron and
> notify me that UserX from X_IP_Space had X amout of
> password issues. I'm currently running this from
> cron and it works fine. My personal version is
> modified to block (l)users after 10 failures on
> 2 separate accounts or 50 failures on one account.
>
> Methodology is, if someone hasn't complained within
> two minutes of something happening to their phones
> that they can't log in, then they won't need to
> use that phone right now. Let them call in and
> complain...
>
> http://www.infiltrated.net/scripts/astrap
>
> --
> =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
> J. Oquendo
> SGFA #579 (FW+VPN v4.1)
> SGFE #574 (FW+VPN v4.1)
>
> echo c2lsQGluZmlsdHJhdGVkLm5ldAo=|\
> python -c "import sys; print sys.stdin.read().decode('base64')"
>
> http://pgp.mit.edu:11371/pks/lookup?op=get&search=0xF684C42E
>
>
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[asterisk-users] Dial problem

[Rilawich Ango]

HI,
  I have 2 TDM400s plugged in a PC.  I failed to use same channels to
make a call to PSTN.  It shows it can't establish connection after
dial command issued.  Below is the log.  Actually, the call is
established as I can hear voice from the called party but the
softphone is still showing ringing.  It seems the TDM card can't get
an answered signal from PSTN.  After 15 seconds, the call dropped
because there is no answered signal.  I want to know how to handle the
problem? Is it related to settng?  Can anyone tell me?

[Nov 23 01:23:11] VERBOSE[5722] logger.c: -- Executing
[EMAIL PROTECTED]:1] Dial("SIP/2001-0a0240c0",
"Zap/2/1872800|15") in new stack
[Nov 23 01:23:11] DEBUG[5722] dsp.c: dsp busy pattern set to 0,0
[Nov 23 01:23:11] DEBUG[5722] chan_zap.c: Dialing '1872800'
[Nov 23 01:23:11] DEBUG[5722] chan_zap.c: Deferring dialing...
[Nov 23 01:23:11] VERBOSE[5722] logger.c: -- Called 2/1872800
[Nov 23 01:23:14] DEBUG[5722] chan_zap.c: Done dialing, but waiting
for progress detection before doing more...
[Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Nobody picked up in 15000 ms
[Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Hungup 'Zap/2-1'
[Nov 23 01:23:27] NOTICE[5722] cdr.c: CDR on channel 'Zap/2-1' not posted
[Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Executing
[EMAIL PROTECTED]:2] Hangup("SIP/2001-0a0240c0", "") in new
stack
[Nov 23 01:23:27] VERBOSE[5722] logger.c:   == Spawn extension
(internal-admin, 91872800, 2) exited non-zero on 'SIP/2001-0a0240c0'
[Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Executing
[EMAIL PROTECTED]:1] Hangup("SIP/2001-0a0240c0", "") in new stack
[Nov 23 01:23:27] VERBOSE[5722] logger.c:   == Spawn extension
(internal-admin, h, 1) exited non-zero on 'SIP/2001-0a0240c0'

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Re: [asterisk-users] quality after call transfer

[F6HQZ]

Hi,

I suspect that you are "transcoding", meaning that the call is comming in a
specific codec format, and the second phone uses another codec. So, when you
do your tranfert, Asterisk is in the middle and is coding from the original
to your phone with two different codecs. If you are passing from G711 to
GSM, for example, it's normal that the audio quality is worst.

Best Regards,
Francois BERGERET


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Rilawich Ango
Envoyé : mercredi 21 novembre 2007 03:25
A : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] quality after call transfer


Hi,
  We are using attended call transfer to transfer the call.  In the
direct call, the quality of the voice and dtmf are acceptable.  After
transfer, the quality becomes worst.  Voice can't be heard clearly and
dtmf wrong detection will occur sometime.  I wonder call transfer will
affect he quality of the call.  Anyone has same experience?  Anything
to do in asterisk level can get a better quality after call transfer?


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Re: [asterisk-users] Problem installing Asterisk

[Ugo Bellavance]

Matt wrote:
> On Nov 21, 2007 11:45 AM, Tilghman Lesher 
> <[EMAIL PROTECTED] 
> > wrote:
> 
> On Wednesday 21 November 2007 09:09:13 Matt wrote:
>  > I have installed Asterisk with FreeTDS many times before (this same
>  > Asterisk and same TDS version)... but today when I did the make
> it gave me
>  > this error:
>  >
>  > ake[1]: Entering directory
> `/home/matth/asterisk126/asterisk-1.2.6/cdr'
> 
> We don't support version 1.2.6 anymore.  That is a VERY old version.
> 
> 
> Sadly, it is one of the only stable versions. Is 1.4 even out of 
> beta yet?   I'm not aware that it is, yet it's being forced down 
> people's throats.

1.4 is out of beta, yes, but if you want to use 1.2, the latest version 
is 1.2.24.

Ugo


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[asterisk-users] Calling with hidden callerid

[Mike]

Hi,
 
I have a wholesale provider that allows me to put any caller id I want when
dialing out.  In some cases, I`d like the outgoing callerid to be hidden.
How do I do this?
 
I`ve set callerid name to "unknown", that works well, but when I put an
empty number it goes out with the name "asterisk".  Which is NOT what I
want.
 
Is there a standard way to say "hid my number"?
 
 
Mike
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[asterisk-users] Odd bug in Siemens C460IP ?

[Robert Lister]

Hello,

I think I have encountered an odd bug in Siemens C460 IP/dect handsets, 
which is a bit annoying, and I'm not (yet) sure how to get round it without 
lots of hacks.

Basically, on all external incoming calls, we set:

exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2)

This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a 
different ring cadence so to differentiate between external and internal 
calls.

Other handsets that do not support Alert-Info: just ignore the presence 
of this header.

When this header is set in a call to the C460 IP, it does not alert, in fact 
it does not respond to any INVITE requests; asterisk just retries the 
requests a few times and then gives up.

Anyone able to reproduce?  I have firmware version 0107 / 041.00

I suppose as a workaround I could add an astDB entry for these extensions, 
and a bit of logic in the dialplan to tell asterisk not to add the header 
for extensions that have that flag set.


Regards,



Rob


-- 
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510

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Re: [asterisk-users] Dial problem

[Eric "ManxPower" Wieling]

Remove callprogress=yes from /etc/asterisk/zapata.conf  There is a 
REASON it is listed as EXPERIMENTAL.  It simply does not work well.

Rilawich Ango wrote:
> HI,
>   I have 2 TDM400s plugged in a PC.  I failed to use same channels to
> make a call to PSTN.  It shows it can't establish connection after
> dial command issued.  Below is the log.  Actually, the call is
> established as I can hear voice from the called party but the
> softphone is still showing ringing.  It seems the TDM card can't get
> an answered signal from PSTN.  After 15 seconds, the call dropped
> because there is no answered signal.  I want to know how to handle the
> problem? Is it related to settng?  Can anyone tell me?
> 
> [Nov 23 01:23:11] VERBOSE[5722] logger.c: -- Executing
> [EMAIL PROTECTED]:1] Dial("SIP/2001-0a0240c0",
> "Zap/2/1872800|15") in new stack
> [Nov 23 01:23:11] DEBUG[5722] dsp.c: dsp busy pattern set to 0,0
> [Nov 23 01:23:11] DEBUG[5722] chan_zap.c: Dialing '1872800'
> [Nov 23 01:23:11] DEBUG[5722] chan_zap.c: Deferring dialing...
> [Nov 23 01:23:11] VERBOSE[5722] logger.c: -- Called 2/1872800
> [Nov 23 01:23:14] DEBUG[5722] chan_zap.c: Done dialing, but waiting
> for progress detection before doing more...
> [Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Nobody picked up in 15000 ms
> [Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Hungup 'Zap/2-1'
> [Nov 23 01:23:27] NOTICE[5722] cdr.c: CDR on channel 'Zap/2-1' not posted
> [Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Executing
> [EMAIL PROTECTED]:2] Hangup("SIP/2001-0a0240c0", "") in new
> stack
> [Nov 23 01:23:27] VERBOSE[5722] logger.c:   == Spawn extension
> (internal-admin, 91872800, 2) exited non-zero on 'SIP/2001-0a0240c0'
> [Nov 23 01:23:27] VERBOSE[5722] logger.c: -- Executing
> [EMAIL PROTECTED]:1] Hangup("SIP/2001-0a0240c0", "") in new stack
> [Nov 23 01:23:27] VERBOSE[5722] logger.c:   == Spawn extension
> (internal-admin, h, 1) exited non-zero on 'SIP/2001-0a0240c0'
> 
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[asterisk-users] NAT keep-alive

[Ugo Bellavance]

Hi,

 On my linksys/sipura phones/ATA, there is a setting called "NAT 
Mapping Enable" and another called "NAT Keep Alive Enable"

These settings must be on in my setup so that my phones/ATA remain 
connected to my * server.  My setup is:

Home LAN - Pfsense (NAT, Dynamic Public IP)- Internet - PFsense (1-to-1 
NAT, Static public IP) - Asterisk server.

I was wondering:

What are doing those parameters?

I looked on my Polycom 330 and I haven't found anything similar... Is 
Linksys the only Mfg that has a similar setting?  The polycom doesn't 
have STUN settings either.  I'm looking to buy some SNOM phones (M3 and 
a wired one), does SNOM phones have something similar?

BTW, are there public STUN servers, or must I have my one to use it?

Regards,

Ugo


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[asterisk-users] Work

[Paul Hales]

Hey, we are looking for someone to work to the end of january , and
maybe even stay on after that. 

_Immediate start_.

Low to Mid level asterisk work (phone support and onsite install work)

You MUST be living in Melbourne, Australia.

Email me off list for more details.

PaulH



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Re: [asterisk-users] Calling with hidden callerid

[Nick Brown]

You can set callerid within the [general] section of your sip.conf. This
should work for you.


On 23/11/07 8:02 AM, "Mike" <[EMAIL PROTECTED]> wrote:

> Hi,
>  
> I have a wholesale provider that allows me to put any caller id I want when
> dialing out.  In some cases, I`d like the outgoing callerid to be hidden.  How
> do I do this?
>  
> I`ve set callerid name to "unknown", that works well, but when I put an empty
> number it goes out with the name "asterisk".  Which is NOT what I want.
>  
> Is there a standard way to say "hid my number"?
>  
>  
> Mike
> 
> 
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Regards,
Nick Brown

Ipera Communications Pty Ltd
Level 1, 9 Denison Street, 
Newcastle West NSW 2302
PO Box 2115, Dangar NSW 2309   

Ü P: +61 2 4910 1000
Ü F: +61 2 4910 1099
Ü ABN: 31 090 964 104

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Re: [asterisk-users] NAT keep-alive

[Ugo Bellavance]

Ugo Bellavance wrote:
> Hi,
> 
>  On my linksys/sipura phones/ATA, there is a setting called "NAT 
> Mapping Enable" and another called "NAT Keep Alive Enable"
> 
> These settings must be on in my setup so that my phones/ATA remain 
> connected to my * server.  My setup is:
> 
> Home LAN - Pfsense (NAT, Dynamic Public IP)- Internet - PFsense (1-to-1 
> NAT, Static public IP) - Asterisk server.
> 
> I was wondering:
> 
> What are doing those parameters?
> 
> I looked on my Polycom 330 and I haven't found anything similar... Is 
> Linksys the only Mfg that has a similar setting?  The polycom doesn't 
> have STUN settings either.  I'm looking to buy some SNOM phones (M3 and 
> a wired one), does SNOM phones have something similar?
> 
> BTW, are there public STUN servers, or must I have my one to use it?
> 
> Regards,
> 
> Ugo

I found a part of the answer here:

http://www.sipura.com/Documents/SPA941AdminGuide.pdf, page 41, but I'm 
still wondering how to get many polycoms working in a setup like mine... 
Or Aastra, or maybe Snom.

Regards,

Ugo


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Re: [asterisk-users] problem with tdm2400p configuration

[Mark Quitoriano]

On Nov 19, 2007 2:31 PM, Mark Quitoriano <[EMAIL PROTECTED]> wrote:
> On Nov 19, 2007 12:10 PM, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
> > Mark Quitoriano wrote:
> > >>> that's the same question i got(regarding question 1). Is it possible
> > >>> for PCI compatibility issue? i need to check for the motherboard specs
> > >>> to post later :)
> > >> Hopefully someone will have someone smarter to say. This specific ioctl,
> > >> if it actually gets to zaptel, should never be answered by -ENOTTY
> > >> ("Inappropriate ioctl for device"). So:
> > >>
> > >>   ls -l /dev/zap
> > >>
> >
> > Sounds to me like a mismatch between Asterisk version and Zaptel version.
>
>
> hmmm what do you mean mismatch? like asterisk 1.2 and zaptel 1.4?
>
> anyway the motherboard is an Intel S300E motherboard. Do you think
> there's a hardware problem?
>
> thanks!
>

anyone can help?

thanks!

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Re: [asterisk-users] Calling with hidden callerid

[Ira]

At 01:02 PM 11/22/2007, you wrote:
I`ve set callerid name to "unknown", that works well, but when I put 
an empty number it goes out with the name "asterisk".  Which is NOT 
what I want.


Is there a standard way to say "hid my number"?


I set it to 1234567890 if that would work for you.

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[asterisk-users] Enough Turkey? Voip Users Conference today at 12:30 EST to help digest it all

[randulo]

Friday at 12:30 PM Eastern, 9:30 AM Pacific, 17:30 GMT join us (or
listen to) what is really the
Asterisk Users Conference that dare not speak its name, except when
called the VOIP Users Conference.

http://www.VoipUsersConference.org

IRC on freenode.net #voip-users-conference


In a nutshell

At 12:30 PM EST,

* PSTN in the US, Call (724) 444-7444
* SIP sip:[EMAIL PROTECTED]

After the call connects, enter the conference id: 22622# PIN#
if your PIN == your CallerID, enter 2# instead of PIN.
if you have no PIN, enter 1#

The PIN just helps us see who you are to call on you when you want to
talk. No personal info is kept.

http://groups.google.com/group/Voip-Users-Conference

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Re: [asterisk-users] Calling with hidden callerid

[Paul Hales]

The dialplan command 'setcallerpres' is also good.

PaulH


On Fri, 2007-11-23 at 12:44 +1100, Nick Brown wrote:
> You can set callerid within the [general] section of your sip.conf.
> This should work for you.
> 
> 
> On 23/11/07 8:02 AM, "Mike" <[EMAIL PROTECTED]> wrote:
> 
> Hi,
> 
> I have a wholesale provider that allows me to put any caller
> id I want when dialing out.  In some cases, I`d like the
> outgoing callerid to be hidden.  How do I do this?
> 
> I`ve set callerid name to "unknown", that works well, but when
> I put an empty number it goes out with the name "asterisk".
>  Which is NOT what I want.
> 
> Is there a standard way to say "hid my number"?
> 
>  
> Mike
> 
> 
> __
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> 
> 
> Regards,
> Nick Brown
> 
> Ipera Communications Pty Ltd
> Level 1, 9 Denison Street,  
> Newcastle West NSW 2302
> PO Box 2115, Dangar NSW 2309
> 
> Ü P: +61 2 4910 1000
> Ü F: +61 2 4910 1099
> Ü ABN: 31 090 964 104
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[asterisk-users] How to bridge two connected calls

[Alberto Pastore]

Hi everybody.

I am in the following scenario:

1 Customer "A" calls an asterisk box over a Zap channel on
   a toll free number during night time

2 The incoming call enters an AGI script on the dialplan

3 The AGI script plays back a welcome message, then
   starts the music-on-hold stream

4 The AGI script originates a calls to a
   stand-by operator's cell phone (operator "B")

5 When the operator "B" answers the call, he is prompted
   (via another AGI script in the dialplan)
   to dial "1" to be recognized as "human" (the AMD()
   function is too random to be useful)

6 After being recognized as human, Customer "A" must
   be bridged to Operator "B"

Everything is ok from 1 to 5, but I cannot really figure out
how to accomplish task #6
I've tried with MeetMe or call parking but with no success.

Can anyone point me in the right direction?
Thanks


-- 
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it

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