asterisk users - Sep 2004 - RES: Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual.

Gonzalo, 

I have an APA III-4FXO and I tried using your configurations, I received the
message below:  

-- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix")
in new stack
Sep  6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted
    -- Called 2217008@Mediatrix
Sep  6 16:54:54 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814409c (len 360) to 192.168.199.5 returned -1: Operation not permitted
  == Spawn extension (from-sip, 92217008, 1) exited non-zero on
'SIP/2010-edfc'
Sep  6 16:54:56 WARNING[1125350192]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814409c (len 360) to 192.168.199.5 returned -1: Operation not permitted
Sep  6 16:54:57 WARNING[1125350192]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call 3445c94e1f7d9b532912c69977067137@192.168.199.4 for
seqno 102 (Critical Request)
Sep  6 16:54:59 WARNING[1125350192]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x814409c (len 360) to 192.168.199.5 returned -1: Operation not permitted
Sep  6 16:55:00 WARNING[1125350192]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call 3445c94e1f7d9b532912c69977067137@192.168.199.4 for
seqno 102 (Non-critical Request)

My configurations are:

SIP.Mib

sipUAServerStaticRegistrarHost = 192.168.199.4
sipUAServerStaticRegistrarPort = 5060
sipUAServerStaticProxyHost = 192.168.199.4
sipUAServerStaticProxyPort = 5060
sipUAServerStaticOutboundProxyHost = 192.168.199.4
sipUAServerStaticOutboundProxyPort = 5060

sipUA1PrefixCCAndAC = 0
sipUA1MainAlias = 1111
sipUA1FriendlyName = 1111
sipUA1OtherAliases = 
sipUA1MustUseSessionTimers = 0
sipUA1MaximumSessionExpirationDelay = 60
sipUA1AuthUsrPwd = 
sipUA1AuthValid = 1

My sip.conf and extension.conf are the same that you send.

Any help will be appreciated.

Kind regards,

Miguel 
>Date: Sat, 4 Sep 2004 15:38:42 -0700 (PDT)
>From: Gonzalo Gasca Meza <xomeboy@yahoo.com>
>Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help.>
>	Anyone with	user manual
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>	<asterisk-users@lists.digium.com>
>Message-ID: <20040904223842.16049.qmail@web60701.mail.yahoo.com>
>Content-Type: text/plain; charset="us-ascii"
>
>
>Here is my configuration for MEdiatrix 1204, by default the 1204 strips one
>digit, so it is not necessary to use:
>
>To dial OUTSIDE
>
>EXTENSIONS.CONF
>
>[locales]
>;ignorepat => 9
exten => _9XXXXXXXX,1,Dial(SIP/${EXTEN-1}@Mediatrix)
exten => _9XXXXXXXX,2,Congestion
exten => _9XXXXXXXX,102,Congestion

To receive calls

[from-pstn]
;Incoming calls from Mediatrix 1204, the 1204, sends an invite to
1111@110.10.200.2


exten => 1111,1,Dial(SIP/100,20)
exten => 1111,2,Voicemail(u100)
exten => 1111,102,Voicemail(b100)
exten => 1111,103,Hangup


****************************************************************************
***************************

SIP.CONF

;Mediatrix Telecomm 1204
[Mediatrix]
type=peer
host=110.10.200.10
mask=255.255.255.255
context=from-sip
qualify=yes
canreinvite=yes
disallow=g729
nat = yes

In MEdiatrix 1204 use a program called Unit Manager Network a Configure the
first port as extension 1111 for port 1, in option SIP. as user agent. also
edit registar an dproxy SIP as the IP address of Asterisk.

Works VERY GOOD with one line, although i have seen some scenarios with more
than 1 line which experince problems.

Hi there,

we have a strange problem with a E100P Digium card at a Deutsche 
Telekom S2M port. (Asterisk 1.0rc2)

We are calling from a number in the same local area code and there 
seems to be only the 6 most significant numbers of the target adress 
arrive in Asterisk.

For example, we are calling 99999123 and the CLI shows only the 999991 
and tries to match that with a extension rule.

(We've set up a rule for _99999XXX with StripMSD(5) and a rule for 
extension 123.)

The strange thing is that if I dial the number with the area code (i.e. 
089/99999123) or from a mobile fone it works and properly connects to 
the correct extension.

We had other pbxs on the same S2M line that work correctly.

Anyone else noticed this problem?

Regards,

Oliver.

Hi there,

there is an option in zaptel.conf where you can configure a 
"prilocaldialplan" to national, local, international and more.

What does that do?

I've tried to make sense of the source code, but I can't figure it out.

Regards,

Oliver.

Hi there,

I noticed, that in my other PBX, I can (and have to) set my local area 
code and the "master number" (in our case a local 5 digit number that 
prefixes the extensions) of our PRI EuroISDN line. I imagine that this 
has some sort of use to those softwares. Is there a place where I have 
to put this information in asterisk config files?

Regards,

Oliver.

On Mon, 6 Sep 2004, Oliver Breidenbach wrote:
> there is an option in zaptel.conf where you can configure a 
> "prilocaldialplan" to national, local, international and more.
> 
> What does that do?
That is what the "a subscriber number" (more or less caller id) is 
marked as when it is sent in the setup message. The "pridialplan" is
what
the called number is marked as.

Peter

On Mon, 2004-09-06 at 16:34, Oliver Breidenbach wrote:
> We are calling from a number in the same local area code and there 
> seems to be only the 6 most significant numbers of the target adress 
> arrive in Asterisk.
Try pridialplan=unknown

-- 
          Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

On Mon, 6 Sep 2004, Eric Wieling wrote:
> On Mon, 2004-09-06 at 16:34, Oliver Breidenbach wrote:
> > We are calling from a number in the same local area code and there 
> > seems to be only the 6 most significant numbers of the target adress 
> > arrive in Asterisk.
> 
> Try pridialplan=unknown
I thought the pridialplan only affected the outgoing messages, not the 
incoming?

Peter