Outgoing call

kandy

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I have an Fritz!box ATA, I am able to receive the call throw my telephone providere by internet but I do not know how to make the outgoing calls.
I need to specify some rules?
I do not know german.

Thank you

p.s.
I went under
Telefonie -> Wahlregeln
and I put
Rufnummernbereich Verbindung
9 myprovider

and I dial 9numeber
but does not work
 
Hi kandy,

I expect you already registered your VoIP-numbers at the menu "Telefonie" => Internettelefonie.

You can set the default dial to VoIP at the menu "Telefonie" => "Nebenstellen"
( Festnetz => land-based numbers )
( Internet => Internet numbers )

The attached picture shows the setting for "VoIP as default".

You can choose e.g. the second line "Festnetz analog" ( so land based line ) by dialing
"*2#" first. "Internet 052546*" my second VoIP provider can be chosen by "*3#"

You can chose dialing rules for number ranges by :
Telefonie => Wahlregeln ( dial rules ) => Rufnummernbereich ( number range ) => Verbindung ( connection ) + selection of " Festnetz" ( land based line ), "Internet", sperren ( bared).

I hope this information helps.

Haveaniceday
 

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The problem was the provider I can receive call using messagenet.it but for some reason I am not able to do. Probably because messagenet use a port 5061 instead the 5060.
I tried an other provider and everything is working.

Thank you for your help
 
Hi Kandy,

I am nit sure if you can change the outgoing port.
May be somebody else knows ?

Best regards,

Haveaniceday

PS: I can translate the answers. If somebody just likes to write in german.
 
Hi.
Vielleicht so:
Code:
~ # voipd -?
usage: voipd voipd [options]
options:
  -?                 - print this help
  -f                 - run in forground. (NOTSET)
  -s                 - stop daemon. (NOTSET)
  -p STRING          - Pidfile. ("/var/run/voipd.pid")
  -v INTEGER         - osip2 debug. (0)
  -i INTEGER         - deprecated. (1)
  -M STRING          - log memory debug to .... (NULL)
  -C                 - capi trace. (NOTSET)
  -N                 - no higher priority. (NOTSET)
  -P INTEGER         - SIP source port. (5060)
  -R                 - tell voipd to register now. (NOTSET)
  -w INTEGER         - set WLAN bridge limit while RTP (packets/s; 0:no-limit).(300)
  -1                 - 1 connection per useragent. (NOTSET)
  -2                 - not longer used. (NOTSET)
  -D STRING          - switch debug logs on. (NULL)
~ #

MfG Oliver
 
Here is part of voipd log

Code:
voipd: startup (AVM FRITZ!Box Fon ata 11.03.37 AVM SIP v1.05.20 Jan 20 2005 14:50:36)
voipd: using capi controller 4
voipd: brutto speed 1120/160
config: VAD off/10000, PLC off, HandleCAPIUnderrun on, Jitter (auto on, 50 ms, 0 packets), tx packet size 0 ms
voipd: 0: [email][email protected][/email] configured
voipd: 1: [email][email protected][/email] configured
voipd: 2 useragents configured
voipd: INFO led: off (value=0)
voipd: priority is -20
voipd: connstatus 0 -> 5
voipd: dns: _sip._udp.sip.messagenet.it: query
voipd: dns: _sip._udp.sip.mytcom.it: query
voipd: dns; _sip._udp.sip.messagenet.it: not found
voipd: >>> Request: REGISTER sip:sip.messagenet.it
voipd: dns; _sip._udp.sip.mytcom.it: not found
voipd: dns: sip.messagenet.it: query
voipd: >>> Request: REGISTER sip:sip.mytcom.it
voipd: dns: sip.mytcom.it: query
voipd: dns: sip.messagenet.it: 212.97.59.76 ttl=25985 from 62.211.69.150.
voipd: dns: sip.mytcom.it: 217.72.105.31 ttl=900 from 62.211.69.150.
voipd: <<< Status: 407 Proxy Authentication Required
voipd: ip address changed 192.168.254.150 -> 80.11X.XXX.XXX
voipd: 530XXXX: my address 80.11X.XXX.XXX:5060
voipd: >>> Request: REGISTER sip:sip.messagenet.it
voipd: <<< Status: 100 Trying
voipd: ip address changed 192.168.254.150 -> 80.11X.XXX.XXX
voipd: 458XXX: my address 80.11X.XXX.XXX:5060
voipd: <<< Status: 401 Unauthorized
voipd: >>> Request: REGISTER sip:sip.mytcom.it
voipd: <<< Status: 200 OK
voipd: sip:[email protected]: REGISTER complete (next in 1620 seconds)
voipd: <<< Status: 100 Trying
voipd: <<< Status: 200 OK
voipd: sip:[email protected]: REGISTER complete (next in 1620 seconds)

I belive that the problem is voipd since the fritz!box tries to register the sip with messagenet on the port 5060 instead the 5061
Probabily it is a limitation of voipd?
 
Hi.
To change the SIP-Listen-Port you have to do the following:
voipd -s (stops the voipd)
voipd -P 5061 (restarts the voipd)

MfG Oliver
 
But I have two sip providers one use the standard port 5060 and the other the port 5061
If I set the port 5061 I have "the same situation" only one provider!
 
Hi,

...
voipd: 530XXXX: my address 80.11X.XXX.XXX:5060
...
voipd: >>> Request: REGISTER sip:sip.mytcom.it
voipd: <<< Status: 200 OK
voipd: sip:[email protected]: REGISTER complete (next in 1620 seconds)
voipd: <<< Status: 100 Trying
voipd: <<< Status: 200 OK
voipd: sip:[email protected]: REGISTER complete (next in 1620 seconds)

It looks like the registration is OK.
The first line says that the Port of the Fritz!Box ATA is 5060 and not the port of your VoIP-provider.

The Fritz!Box uses the first Provider in the "Nebenstellen" list as default line.
If you want to use the other VoIP-provider for some special numbers you need the dial rules.

p.E.: you want to dial the number 01234/56789012 and you want that all numbers which start with 0123 have to go through provider2
the dial rule should be: "Rufnummernbereich":0123 and "Verbindung:" provider2

Now you have to dial the number 0123456789012 and it should go through the second Provider.

I hope i could help.

Robert
 
When you change the sip port to 5061, is it then possible to do a call over messagenet?
Someone here on the board has two FritzBoxes in one network. So he changed the port to 5061 for the second one.
No Problem, he even is able to do an internal call from one Box to the other...

I have actually found a thread opened by you.
Was the solution not right for you? Or why are you asking again?
Are you able to post a log of a failling call?

MfG Oliver
 
Here you can find my rules and I think are right
 

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The problem is that I can receive the call (Registration is ok) but I cannot do.
I have a Grandstream 101 in my network and it is able to go in internet to the same provider.
With the grandstream I am able to make and receive calls.
I am wondering if I am able to manage the two providers with the same fritz.
 
Your rules aren't correct.

p.E. you want to dial the numer 123456 with provider 4580.
you dial in your phone 458123456, right?

the FritzBox now use the provider 4580, but dials the NUMBER 458123456. but you wanted the NUMBER 123456
That's the problem.

As prefix you could use
*11# for land-based-numbers
*12# for the first VoIP-Provider in the "Internettelefonie" list
*13# for the second VoIP-Provider in the "Internettelefonie" list
...

or for each Nebenstelle (Fon1 / Fon2)
*1# for the first entry in the "Nebenstellen" list
*2# for the second entry in the "Nebenstellen" list
...

If your 4580 provider is the second VoIP Provider in the Internettelefonie"-list then you should diel the following number: *13#123456 to dial the number 123456 over the 4580 Provider

Greets,
Robert
 

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I was not clear in my post.
if I want digit a number 458123 this number must be dial thorw the provider 4580.
If I want digit a number 530123 this number must be dial throw the providere 530.
Since are number related to the provider.

Here you have the call log for the outgoing call

Code:
Mar 14 11:32:50 voipd[304]: dns: _sip._udp.sip.messagenet.it: query
Mar 14 11:32:50 voipd[304]: dns; _sip._udp.sip.messagenet.it: not found
Mar 14 11:32:50 voipd[304]: >>> Request: INVITE sip:[email protected]
Mar 14 11:32:51 voipd[304]: >>> Request: INVITE sip:[email protected]
Mar 14 11:32:53 voipd[304]: >>> Request: INVITE sip:[email protected]
Mar 14 11:32:56 voipd[304]: >>> Request: INVITE sip:[email protected]
Mar 14 11:33:00 voipd[304]: disconnected(appl=2 plci=0x204 ncci=0x0 incoming): remote: 0x3490 (0x0000) - Normal call clearing
Mar 14 11:33:00 voipd[304]: >>> Request: INVITE sip:[email protected]
Mar 14 11:33:00 voipd[304]: >>> Request: CANCEL sip:[email protected]
Mar 14 11:33:01 voipd[304]: >>> Request: CANCEL sip:[email protected]
Mar 14 11:33:03 voipd[304]: >>> Request: CANCEL sip:[email protected]
Mar 14 11:33:06 voipd[304]: >>> Request: CANCEL sip:[email protected]
Mar 14 11:33:09 voipd[304]: >>> Request: INVITE sip:[email protected]
Mar 14 11:33:11 voipd[304]: >>> Request: CANCEL sip:[email protected]
Mar 14 11:33:16 voipd[304]: >>> Request: CANCEL sip:[email protected]
Mar 14 11:33:21 voipd[304]: >>> Request: CANCEL sip:[email protected]
Mar 14 11:33:23 voipd[304]: call to sip:[email protected] terminated (0)
Mar 14 11:33:26 voipd[304]: >>> Request: CANCEL sip:[email protected]
Mar 14 11:33:31 voipd[304]: >>> Request: CANCEL sip:[email protected]
where the 530YYYY is the called.
 
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