from:"Daniel Heckl"

Re: [asterisk-users] Voice recognition IVR Is it possible?

2016-02-22 Thread Daniel Heckl

Read README, check the requirements and get the google speech api key.

Then add a custom destination in FreePBX and edit your extensions_custom.conf.

> Am 22.02.2016 um 21:03 schrieb Daniel Chavez :
> 
> Ok. Where I am now is, I copied the speech-recog.agi to the agi-bin directory.
> Where do I go from here?
> 
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice recognition IVR Is it possible?

2016-02-22 Thread Daniel Heckl

I use FreePBX as well. There is no module for speech recognition. You have too 
create a custom destination.

> Am 22.02.2016 um 20:53 schrieb Daniel Chavez :
> 
> Thanks, this looks promising. I was wondering if there's an easier way to get 
> this to work inside FreePBX?
> I have all of the dependencies installed for it, but now I want to know if 
> there's a mod I can use in FreePBX to get it setup?
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice recognition IVR Is it possible?

2016-02-22 Thread Daniel Heckl

Daniel,

try this http://zaf.github.io/asterisk-speech-recog/.

I have tested it myself, it works very well.

Daniel

> Am 22.02.2016 um 19:34 schrieb Daniel Chavez :
> 
> Thanks for the link.
> Are there no free alternatives for speech recognition?
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PJSIP Stun/ICE

2016-01-26 Thread Daniel Heckl

Bryant,

I have the same problem with dynamic public IPs and PJSIP. What is your idea to 
solve the problem?

My suggestion would be to write a script that monitors the change, 
pjsip.transports.conf updated and Asterisk restarts?

Daniel

> Am 26.01.2016 um 14:21 schrieb Joshua Colp :
> 
> Bryant Zimmerman wrote:
>> Joshua
>> So once a transport is pulled from the transports table in realtime
>> during asterisk startup it can't get any updates?
>> Can a new transport be added to the table and the associated endpoints
>> be updated to use the new transport, or are transport types only read at
>> startup across the board?
> 
> Transports can only be loaded at startup. This stems from PJSIP not being 
> dynamic with transports (it doesn't like its environment changed to that 
> degree while in use). I'm afraid if your IP changes you'd have to restart 
> Asterisk when you are using PJSIP.
> 
> -- 
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
> 
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>  http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PJSIP Stun/ICE

2016-01-26 Thread Daniel Heckl

Bryant,

that sounds interesting. I am searching for a script which monitors and updates 
the ip address. Does this your script? Can you share your script with us?

Thanks
Daniel

> Am 26.01.2016 um 16:39 schrieb Bryant Zimmerman <brya...@zktech.com>:
> 
> Daniel
>  
> Thank you for your response. I was considering this as well. I have a script 
> that monitors the IP Address now. I was hoping to use the real-time 
> transports table now that alembic creates. I am trying to figure out which 
> pjsip module is responsible for the transports contexts as I need to now 
> configure it in the sorcery.conf file. I thought it would be under the 
> [res_pjsip] context, but it is not even trying to pull from my transports 
> table when it is there.  I am hoping someone will know what module it is in 
> so I can move my configuration under the correct context.
>  
> Thanks
> 
> Bryant
>  
> From: "Daniel Heckl" <daniel.he...@gmail.com>
> Sent: Tuesday, January 26, 2016 10:15 AM
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] PJSIP Stun/ICE
>  
> Bryant,
> 
> I have the same problem with dynamic public IPs and PJSIP. What is your idea 
> to solve the problem?
> 
> My suggestion would be to write a script that monitors the change, 
> pjsip.transports.conf updated and Asterisk restarts?
> 
> Daniel
> 
> > Am 26.01.2016 um 14:21 schrieb Joshua Colp <jc...@digium.com>:
> >
> > Bryant Zimmerman wrote:
> >> Joshua
> >> So once a transport is pulled from the transports table in realtime
> >> during asterisk startup it can't get any updates?
> >> Can a new transport be added to the table and the associated endpoints
> >> be updated to use the new transport, or are transport types only read at
> >> startup across the board?
> >
> > Transports can only be loaded at startup. This stems from PJSIP not being 
> > dynamic with transports (it doesn't like its environment changed to that 
> > degree while in use). I'm afraid if your IP changes you'd have to restart 
> > Asterisk when you are using PJSIP.
> >
> > --
> > Joshua Colp
> > Digium, Inc. | Senior Software Developer
> > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> > Check us out at: www.digium.com & www.asterisk.org
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Transfer calls "on demand"

2015-12-29 Thread Daniel Heckl

You are searching for „Call Pickup“. It is implemented in Asterisk by default.

https://wiki.asterisk.org/wiki/display/AST/Call+Pickup 

Take a look under section „Configuration Options“.

Daniel

> Am 29.12.2015 um 07:53 schrieb Luca Bertoncello :
> 
> Hi list!
> 
> Right now I configured my Asterisk to forward the calls for the number X to
> both phones (mine and the phone of my wife).
> It works, of course, but I'm not enthusiast...
> 
> I see what we have at office: if one phone rings, other phones in the same
> group can "catch the call", so that if a colleague is not present, another
> colleague can catch the call.
> 
> I'd like to have the same procedure at home. I think, Asterisk can do that,
> but I have no idea how to implement this.
> 
> Shortly: what I want is that every phone rings only on calls for the own
> number, and I can catch the call from the other phone, if for example my wife
> is not at home, for example pressing "*5#" or other key combination.
> 
> Thanks a lot for your suggestion!
> 
> Luca Bertoncello
> (lucab...@lucabert.de)
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Transfer calls "on demand"

2015-12-29 Thread Daniel Heckl

On top of the page: "Call pickup support added in Asterisk 11“

I think that is the problem. I do not know a solution for 1.8, but maybe 
someone other.

> Am 29.12.2015 um 10:20 schrieb Luca Bertoncello <lucab...@lucabert.de>:
> 
> Daniel Heckl <daniel.he...@gmail.com> schrieb:
> 
>> You are searching for „Call Pickup“. It is implemented in Asterisk by
>> default.
>> 
>> https://wiki.asterisk.org/wiki/display/AST/Call+Pickup
>> <https://wiki.asterisk.org/wiki/display/AST/Call+Pickup> Take a look under
>> section „Configuration Options“.
> 
> Hi, Daniel!
> 
> Thanks for your answer...
> I'm using Asterisk 1.8.30.0 on an OpenWRT-Router.
> I found the configuration for call pickup in the sip.conf and features.conf,
> so I tried to activate it...
> Unfortunately, unsuccessfully...
> 
> So, my sip.conf:
> 
> callgroup=1,3-4 ; We are in caller groups 1,3,4
> pickupgroup=1,3-5   ; We can do call pick-p for call group 1,3,4,5
> 
> my features.conf:
> 
> ; Pickup Options
> ;
> pickupexten = *8   ; Configure the pickup extension. (default is 
> *8)
> ;pickupsound = beep ; to indicate a successful pickup (default: 
> no sound)
> ;pickupfailsound = beeperr  ; to indicate that the pickup failed 
> (default: no sound)
> 
> my users.conf:
> 
> [general]
> callgroup = 1
> pickupgroup = 1
> 
> my extensions.conf:
> 
> [anika_incoming]
> exten => _0049351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0049351222,Set(CHANNEL(pickupgroup)=1)
> exten => _0049351222,n,Dial(local/222@anika_incoming)
> exten => _0351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0351222,n,Dial(local/222@anika_incoming)
> exten => _222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _222,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" = "+49" 
> ]?0${CALLERID(num):3}:${CALLERID(num)})})  ; Damit das "+49" mit "0" ersetzt 
> wird
> exten => _222,n,Set(CHANNEL(musicclass)=default)
> ;;;exten => 
> _222,n,Dial(SIP/0049351222/1@luca_for_anika_voip_mobile,19,RcxX)
> exten => _222,n,Dial(SIP/0049351222,19,RcxX)
> exten => _222,n,Verbose(2,Voicemail for Anika)
> exten => _222,n,Set(CALLERID(name)=)  
>  ; Damit in der E-Mail der AB nicht den Namen steht
> exten => _222,n,VoiceMail(0049351222,us)
> exten => _222,n,Hangup
> 
> Then I called the 222 with my mobile phone and I tried to get the call
> from the other phone, calling the *8.
> Unfortunately I get an error (invalid number) on the display of the phone,
> and the phone 222 continue to ring.
> No error on the log of Asterisk...
> 
> Any suggestion?
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



smime.p7s
Description: S/MIME cryptographic signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Update peer IP address

2015-09-16 Thread Daniel Heckl

Sebastian,

If I have understood you correctly, the SIP communication is now via NAT 
instead forwarded ports. For safety, it is much better.

I think it is not because of a UDP timeout, but rather because of a NAT 
timeout. For this is "qualify" exactly the right thing to let the NAT port 
opened. 

Daniel

> Am 14.09.2015 um 21:51 schrieb Marie Fischer :
> 
> 
> On 14.09.2015, at 21:58, Sebastian Kemper  wrote:
> 
>> So I got rid of the firewall rule that opened the RTP ports. And then it
>> dawned on me that I don't even need to open the 5060 port. The REGISTER
>> requests established a UDP connection that the kernel's conntrack module
>> was tracking anyway. The only issue was that the REGISTERs occurred only
>> every 480s and the UDP connections were removed after 180s already.
>> 
>> So at first I raised net.netfilter.nf_conntrack_udp_timeout_stream to
>> 500. That worked. But I didn't really want to raise the default. So
>> instead I added "qualify=yes" to the dtag_inbound peer. Now asterisk is
>> sending an OPTIONS request to Telekom every 120s (I raised the frequency
>> from 60 to 120 by setting "qualifyfreq=120" under [general]), which
>> keeps the connection open.
> 
> As far as I understand, raising the UDP session timeout (or lowering the 
> REGISTER timeout, if possible) is actually the better solution. Most Telcos I 
> know don't answer the OPTIONS request anyway and some might object to the 
> traffic overhead.
> 
> -- 
> marie
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Update peer IP address

2015-04-02 Thread Daniel Heckl

Okay, Scott, I think we are on the wrong path. Maybe I'm wrong though.

I will summarize again briefly the problems together:
The peer ip address could be another than the ip address of incoming invites
After an re-register the REGISTER is send to the new SIP server, answered with 
OK. But the peer ip address is still the old one (sip show peers).
If now is a INVITE, the request is answered with 401 Unauthorized.

That’s why I would say, the problem is not the port or a needed authentication. 
My Asterisk works behind a NAT without port forwarding and nat=no, I have 
qualify=yes that it does not come to a NAT timeout.

Here is an example. The peer ip address was at this time 217.0.23.100, the 
INVITE came from 217.0.23.68 an was rejected with 401 Unauthorized:

INVITE sip:06123456789@80.000.111.222:45061 SIP/2.0
Max-Forwards: 58
Via: SIP/2.0/UDP 
217.0.23.68:5060;branch=z9hG4bKg3Zqkv7ib7h2smv8whryjnos88srot1i7
To: sip:6123456...@telekom.de
From: sip:+49123456...@tel.t-online.de;user=phone;tag=h7g4Esbg_44c62525
Call-ID: af71bbfbf269b895@62.155.0.75
CSeq: 3950540 INVITE
Contact: sip:sgc_c@217.0.23.68;transport=udp
Record-Route: sip:217.0.23.68;transport=udp;lr
Min-Se: 900
P-Asserted-Identity: sip:+49123456...@tel.t-online.de;user=phone
Session-Expires: 3600
Supported: histinfo
Supported: timer
Supported: norefersub
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 204
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE

v=0
o=- 0 0 IN IP4 217.0.23.68
s=-
c=IN IP4 217.0.4.134
t=0 0
m=audio 36480 RTP/AVP 9 8 102
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:102 telephone-event/8000
a=maxptime:20
a=ptime:20

 Am 02.04.2015 um 22:00 schrieb Scott Griepentrog sgriepent...@digium.com:
 
 Actually, the IP address is still used to identify the incoming invite.  With 
 the insecure=port option set, Asterisk will presume the invite to still match 
 the trunk account even if the NAT router has mangled (changed) the port 
 number.  My suspicion is that when the new register goes out, it's creating a 
 new state in the firewall, resulting in a new port number, which is why you 
 would have to allow anonymous calls to then accept it without insecure=port.  
 The other possibility is that you have a port forward in the router set, 
 which is similarly mangling the port number.  With a valid registration being 
 held, and assuming the router does not drop UDP states faster than 30 
 minutes, and also assuming that the provider is sending you invites on the 
 registered port rather than always on 5060, there should not be a need for an 
 inbound port forward to Asterisk, and you should not need insecure=port.
 
 The invite option disables authentication - which means only that Asterisk 
 will not force a check of the password on the other end.  Where the IP 
 address is well known and trusted, the extra overhead and delay of 
 authenticating incoming INVITEs is not needed.
 
 
 
 On Thu, Apr 2, 2015 at 2:28 PM, Daniel Heckl daniel.he...@gmail.com 
 mailto:daniel.he...@gmail.com wrote:
 Scott, I have changed the configuration as said it and will test it. I’m 
 curious.
 
 Can you briefly explain what insecure=invite,port does?
 
 ;insecure=port  ; Allow matching of peer by IP address without
 ; matching port number
 ;insecure=invite; Do not require authentication of incoming INVITEs
 ;insecure=port,invite   ; (both)
 
 Do I understand correctly that in this mode the IP address is not checked and 
 no authentication is required? 
 
 Am 02.04.2015 um 20:11 schrieb Scott Griepentrog sgriepent...@digium.com 
 mailto:sgriepent...@digium.com:
 
 I'd be curious if setting
 
 insecure=invite,port
 
 makes any difference either (without alllowguest on).
 
 
 On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl daniel.he...@gmail.com 
 mailto:daniel.he...@gmail.com wrote:
 Ok, I have tested dnsmgr. This is not a solution, the situation has not 
 changed. With dnsmgr I can not place outbound calls. I do not know why and 
 what dnsmgr really do.
 
 My current solution is as follows:
 
 Say allowguest=yes, configure the default context that there can not be 
 placed outbound calls. Use iptables to DROP all at your SIP port and allow 
 only your local phones and the sip trunk ip range. I think srvlookup must be 
 set to yes to place outbound calls if there is an ip address change.
 
 I think with the restriction of the firewall that should be a secure 
 solution.
 
  Am 01.04.2015 um 19:23 schrieb Sebastian Kemper sebastian...@gmx.net 
  mailto:sebastian...@gmx.net:
 
  On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote:
  On 4/1/15 10:48 AM, Daniel Heckl wrote:
  John,
 
  thank you four your answer. I think you have misunderstood the
  problem. It’s about a ip address change of the sip trunk, not of my
  asterisk server.
  You would probably benefit by enabling the DNS Manager to allow for
  dynamic IP changes:
 
  # cat dnsmgr.conf [general

Re: [asterisk-users] Update peer IP address

2015-04-02 Thread Daniel Heckl

I do not want set allowguest=yes. The problem is, there is no official list 
with ip addresses of Telekom Germany. But I think all ip addresses comes from 
the ip range 217.0.0.0/13.

I have now the following addition to sip.conf. I think it is the only safe 
option. Or what would you say?

[telekom](!)
context=from-trunk
type=peer
defaultuser=
authuser=
remotesecret=
fromdomain=tel.t-online.de
qualify=no
dtmfmode=rfc2833
directmedia=no
sendrpid=pai
trustrpid=no
insecure=port,invite
disallow=all
allow=g722
allow=alaw
allow=gsm
deny=0.0.0.0/0
permit=217.0.0.0/13

[DTAG-IP_IN18_016](telekom)
host=217.0.18.16

[DTAG-IP_IN18_036](telekom)
host=217.0.18.36

etc.


 Am 02.04.2015 um 23:21 schrieb Scott Griepentrog sgriepent...@digium.com:
 
 That sounds like asterisk was working 100% correctly.  If you receive an 
 INVITE from an unknown IP address, then it should fail.  Unless you want to 
 allow anonymous, which is genearlly a very bad idea.
 
 If you are registering to IP X, but the provider may be transmitting invites 
 from any number of other IP addresses, then you need a list of IP addresses, 
 and have a trunk configuration set up for each one so that they are all 
 recognized (with insecure=port,invite).
 
 If the provider is requiring you to accept invites from random IP addresses, 
 get a new provider.
 
 
 On Thu, Apr 2, 2015 at 3:23 PM, Daniel Heckl daniel.he...@gmail.com 
 mailto:daniel.he...@gmail.com wrote:
 Okay, Scott, I think we are on the wrong path. Maybe I'm wrong though.
 
 I will summarize again briefly the problems together:
 The peer ip address could be another than the ip address of incoming invites
 After an re-register the REGISTER is send to the new SIP server, answered 
 with OK. But the peer ip address is still the old one (sip show peers).
 If now is a INVITE, the request is answered with 401 Unauthorized.
 
 That’s why I would say, the problem is not the port or a needed 
 authentication. My Asterisk works behind a NAT without port forwarding and 
 nat=no, I have qualify=yes that it does not come to a NAT timeout.
 
 Here is an example. The peer ip address was at this time 217.0.23.100, the 
 INVITE came from 217.0.23.68 an was rejected with 401 Unauthorized:
 
 INVITE sip:06123456789@80.000.111.222:45061  SIP/2.0
 Max-Forwards: 58
 Via: SIP/2.0/UDP 
 217.0.23.68:5060;branch=z9hG4bKg3Zqkv7ib7h2smv8whryjnos88srot1i7
 To: sip:6123456...@telekom.de 
 From: sip:+49123456...@tel.t-online.de;user=phone ;tag=h7g4Esbg_44c62525
 Call-ID: af71bbfbf269b895@62.155.0.75 mailto:af71bbfbf269b895@62.155.0.75
 CSeq: 3950540 INVITE
 Contact: sip:sgc_c@217.0.23.68;transport=udp 
 Record-Route: sip:217.0.23.68;transport=udp;lr 
 Min-Se: 900
 P-Asserted-Identity: sip:+49123456...@tel.t-online.de;user=phone 
 Session-Expires: 3600
 Supported: histinfo
 Supported: timer
 Supported: norefersub
 Content-Type: application/sdp
 Content-Disposition: session
 Content-Length: 204
 Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
 
 v=0
 o=- 0 0 IN IP4 217.0.23.68
 s=-
 c=IN IP4 217.0.4.134
 t=0 0
 m=audio 36480 RTP/AVP 9 8 102
 a=rtpmap:9 G722/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:102 telephone-event/8000
 a=maxptime:20
 a=ptime:20
 
 Am 02.04.2015 um 22:00 schrieb Scott Griepentrog sgriepent...@digium.com 
 mailto:sgriepent...@digium.com:
 
 Actually, the IP address is still used to identify the incoming invite.  
 With the insecure=port option set, Asterisk will presume the invite to still 
 match the trunk account even if the NAT router has mangled (changed) the 
 port number.  My suspicion is that when the new register goes out, it's 
 creating a new state in the firewall, resulting in a new port number, which 
 is why you would have to allow anonymous calls to then accept it without 
 insecure=port.  The other possibility is that you have a port forward in the 
 router set, which is similarly mangling the port number.  With a valid 
 registration being held, and assuming the router does not drop UDP states 
 faster than 30 minutes, and also assuming that the provider is sending you 
 invites on the registered port rather than always on 5060, there should not 
 be a need for an inbound port forward to Asterisk, and you should not need 
 insecure=port.
 
 The invite option disables authentication - which means only that Asterisk 
 will not force a check of the password on the other end.  Where the IP 
 address is well known and trusted, the extra overhead and delay of 
 authenticating incoming INVITEs is not needed.
 
 
 
 On Thu, Apr 2, 2015 at 2:28 PM, Daniel Heckl daniel.he...@gmail.com 
 mailto:daniel.he...@gmail.com wrote:
 Scott, I have changed the configuration as said it and will test it. I’m 
 curious.
 
 Can you briefly explain what insecure=invite,port does?
 
 ;insecure=port  ; Allow matching of peer by IP address without
 ; matching port number
 ;insecure=invite; Do not require authentication of incoming INVITEs
 ;insecure

Re: [asterisk-users] Update peer IP address

2015-04-02 Thread Daniel Heckl

Ok, I have tested dnsmgr. This is not a solution, the situation has not 
changed. With dnsmgr I can not place outbound calls. I do not know why and what 
dnsmgr really do. 

My current solution is as follows:

Say allowguest=yes, configure the default context that there can not be placed 
outbound calls. Use iptables to DROP all at your SIP port and allow only your 
local phones and the sip trunk ip range. I think srvlookup must be set to yes 
to place outbound calls if there is an ip address change.

I think with the restriction of the firewall that should be a secure solution.

 Am 01.04.2015 um 19:23 schrieb Sebastian Kemper sebastian...@gmx.net:
 
 On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote:
 On 4/1/15 10:48 AM, Daniel Heckl wrote:
 John,
 
 thank you four your answer. I think you have misunderstood the
 problem. It’s about a ip address change of the sip trunk, not of my
 asterisk server.
 You would probably benefit by enabling the DNS Manager to allow for
 dynamic IP changes:
 
 # cat dnsmgr.conf [general] enable=yes ; enable creation
 of managed DNS lookups ;   default is 'no' refreshinterval=180   ;
 refresh managed DNS lookups every n seconds ;   default is 300 (5
 minutes)
 
 Hello Andres,
 
 I read that same suggestion elsewhere in connection with Deutsche
 Telekom, so it seems there's some benefit in it.
 
 Daniel, did you try it out already?
 
 Kind regards,
 Sebastian
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Update peer IP address

2015-04-02 Thread Daniel Heckl

Scott, I have changed the configuration as said it and will test it. I’m 
curious.

Can you briefly explain what insecure=invite,port does?

;insecure=port  ; Allow matching of peer by IP address without
; matching port number
;insecure=invite; Do not require authentication of incoming INVITEs
;insecure=port,invite   ; (both)

Do I understand correctly that in this mode the IP address is not checked and 
no authentication is required? 

 Am 02.04.2015 um 20:11 schrieb Scott Griepentrog sgriepent...@digium.com:
 
 I'd be curious if setting
 
 insecure=invite,port
 
 makes any difference either (without alllowguest on).
 
 
 On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl daniel.he...@gmail.com 
 mailto:daniel.he...@gmail.com wrote:
 Ok, I have tested dnsmgr. This is not a solution, the situation has not 
 changed. With dnsmgr I can not place outbound calls. I do not know why and 
 what dnsmgr really do.
 
 My current solution is as follows:
 
 Say allowguest=yes, configure the default context that there can not be 
 placed outbound calls. Use iptables to DROP all at your SIP port and allow 
 only your local phones and the sip trunk ip range. I think srvlookup must be 
 set to yes to place outbound calls if there is an ip address change.
 
 I think with the restriction of the firewall that should be a secure solution.
 
  Am 01.04.2015 um 19:23 schrieb Sebastian Kemper sebastian...@gmx.net 
  mailto:sebastian...@gmx.net:
 
  On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote:
  On 4/1/15 10:48 AM, Daniel Heckl wrote:
  John,
 
  thank you four your answer. I think you have misunderstood the
  problem. It’s about a ip address change of the sip trunk, not of my
  asterisk server.
  You would probably benefit by enabling the DNS Manager to allow for
  dynamic IP changes:
 
  # cat dnsmgr.conf [general] enable=yes ; enable creation
  of managed DNS lookups ;   default is 'no' refreshinterval=180   ;
  refresh managed DNS lookups every n seconds ;   default is 300 (5
  minutes)
 
  Hello Andres,
 
  I read that same suggestion elsewhere in connection with Deutsche
  Telekom, so it seems there's some benefit in it.
 
  Daniel, did you try it out already?
 
  Kind regards,
  Sebastian
 
  --
  _
  -- Bandwidth and Colocation Provided by http://www.api-digital.com 
  http://www.api-digital.com/ --
  New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users 
  http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com 
 http://www.api-digital.com/ --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users 
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 -- 
 
 Scott Griepentrog
 Digium, Inc · Software Developer
 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
 direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
 Check us out at: http://digium.com http://digium.com/ · http://asterisk.org 
 http://asterisk.org/
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Update peer IP address

2015-04-01 Thread Daniel Heckl


Scott, thank you four your reply.

I had already though about both options, but the problem is, that after an ip 
change AND a new registration the ip address of the peer is not updated 
automatically. INVITES are answered with 401.

Only after a sip reload the peer works again.

That can't be normal...

Daniel

 Am 31.03.2015 um 22:45 schrieb Scott Griepentrog sgriepent...@digium.com:
 
 You have two options for dealing with an IP change during the registration 
 period:
 
 1) set the registration time to shorter period of time to minimize the 
 downtime
 
 2) detect that the IP address has changed via whatever method available, and 
 then issue a sip reload CLI command to asterisk, which will cause it to 
 resend registrations immediately.
 
 On Tue, Mar 31, 2015 at 1:36 PM, Daniel Heckl daniel.he...@gmail.com wrote:
 Maybe someone could elaborate on my first question again.
 
 If the ip address changes while a REGISTER period, the ip address of the 
 peer isn't been updated. How can asterisk update the ip address of the peer?
 
 Am 31.03.2015 um 12:36 schrieb Daniel Heckl daniel.he...@gmail.com:
 
 Hello Sebastian,
 
 I had already seen this list of the hosts, but it is not active. All 
 servers with which my Asterisk has been communicated are not listed.
 
 A port scan, to eventually update the list, found hundreds of servers 
 provided in the address range 217.0.0.0/13 with open port 5060, some were 
 even not found. I think there must be another solution.
 
 If I change insecure to insecure=port,invite - could that be a solution?
 
 Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no 
 problem)? Has there anyone experience with dynamic ip addresses of Asterisk?
 
 Daniel
 
 Am 30.03.2015 um 20:09 schrieb Sebastian Kemper sebastian...@gmx.net:
 
 On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote:
 Hello
 
 I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom
 Germany. We have sometimes problems with incoming and outgoing calls.
 I hope I can explain it understandable.
 
 Hello Daniel,
 
 I'll find myself in the same situation a few weeks from now :-)
 
 
 For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de
 http://tel.t-online.de/), the message is answered with OK and the
 peer is registered.
 
 Usually INVITES comes now from this ip address. All works fine. But
 sometimes INVITES comes from an other IP address, for example
 217.0.23.100. This request Asterisk responds with 401 Unauthorized.
 
 In the next register procedure REGISTER are sent to the new ip address
 and answered also with OK. But qualify OPTIONS are continue be sent to
 the old ip address. Incoming and outgoing calls are canceled. Outgoing
 calls are answered with Forbidden.
 
 Even if the REGISTER procedure works with the new ip address, the
 peers are connected with the old address.
 
 Waiting doesn’t help, only a „sip reload“ update the ip address of the
 peer. 
 
 What is the solution for this problem? How can asterisk update the
 peer?
 
 I think the solution - for the inbound issue at least - could be to add
 more hosts as a peer. Have a looks at this forum post:
 
 http://www.ip-phone-forum.de/showthread.php?t=268787p=1999371viewfull=1#post1999371
 
 The user used a template and than he added peers, each with its own IP
 address. The provided list was last updated in 2014, though, so I assume
 the provider in the meantime has added to that list.
 
 It looks pretty tedious, though, I mean there could be dozens of IPs
 you'd have to add. But I guess this is the way to go with Asterisk 11
 and chan_sip.
 
 The future looks brighter :-) I read that with pjsip, which I understand
 is the replacement for chan_sip, you can have one peer entry and match
 an IP range instead of a single host. That should tidy up the dialplan.
 
 What I'm a little afraid of is the SIP provider using IPs out of a range
 that they also use for other services. Maybe out of the same range they
 hand out IPs to their customers. I guess we got to be careful :-)
 
 Kind regards,
 Sebastian
 
 The Asterisk is local behind a NAT with a firewall, following settings
 are used:
 
 externhost with DynDNS stun with stun.t-online.de
 http://stun.t-online.de/ nat=yes srvlookup=yes allowguest=no
 trustrpid=no insecure=invite qualify=yes
 
 Thank you!  Daniel
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

Re: [asterisk-users] Update peer IP address

2015-04-01 Thread Daniel Heckl

John,

thank you four your answer. I think you have misunderstood the problem. It’s 
about a ip address change of the sip trunk, not of my asterisk server.

Kind regards,
Daniel

 Am 01.04.2015 um 16:40 schrieb Tech Support aster...@voipbusiness.us 
 mailto:aster...@voipbusiness.us:
 
 If I correctly understand what the problem is, what I did was write a 
 script that runs out of CRON every 15 minutes. It checks the outside IP 
 address by querying http://checkip.dyndns.org http://checkip.dyndns.org/ 
 and compares it to the IP address stored in the parameter “externip” in the 
 [general] section of sip.conf. If the two values are the same, the script 
 exits quietly. If they are different, the script updates “externip” with the 
 new address, does a sip reload, and shoots me an email saying there was an 
 update. It's a fairly simple and straightforward process and does the job. I 
 use this script for all PBX’s that are behind a NAT. I hope this helps.
 Regards;
 John

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Update peer IP address

2015-03-31 Thread Daniel Heckl

Hello Sebastian,

I had already seen this list of the hosts, but it is not active. All servers 
with which my Asterisk has been communicated are not listed.

A port scan, to eventually update the list, found hundreds of servers provided 
in the address range 217.0.0.0/13 with open port 5060, some were even not 
found. I think there must be another solution.

If I change insecure to insecure=port,invite - could that be a solution?

Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? 
Has there anyone experience with dynamic ip addresses of Asterisk?

Daniel

 Am 30.03.2015 um 20:09 schrieb Sebastian Kemper sebastian...@gmx.net:
 
 On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote:
 Hello
 
 I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom
 Germany. We have sometimes problems with incoming and outgoing calls.
 I hope I can explain it understandable.
 
 Hello Daniel,
 
 I'll find myself in the same situation a few weeks from now :-)
 
 
 For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de
 http://tel.t-online.de/), the message is answered with OK and the
 peer is registered.
 
 Usually INVITES comes now from this ip address. All works fine. But
 sometimes INVITES comes from an other IP address, for example
 217.0.23.100. This request Asterisk responds with 401 Unauthorized.
 
 In the next register procedure REGISTER are sent to the new ip address
 and answered also with OK. But qualify OPTIONS are continue be sent to
 the old ip address. Incoming and outgoing calls are canceled. Outgoing
 calls are answered with Forbidden.
 
 Even if the REGISTER procedure works with the new ip address, the
 peers are connected with the old address.
 
 Waiting doesn’t help, only a „sip reload“ update the ip address of the
 peer. 
 
 What is the solution for this problem? How can asterisk update the
 peer?
 
 I think the solution - for the inbound issue at least - could be to add
 more hosts as a peer. Have a looks at this forum post:
 
 http://www.ip-phone-forum.de/showthread.php?t=268787p=1999371viewfull=1#post1999371
 
 The user used a template and than he added peers, each with its own IP
 address. The provided list was last updated in 2014, though, so I assume
 the provider in the meantime has added to that list.
 
 It looks pretty tedious, though, I mean there could be dozens of IPs
 you'd have to add. But I guess this is the way to go with Asterisk 11
 and chan_sip.
 
 The future looks brighter :-) I read that with pjsip, which I understand
 is the replacement for chan_sip, you can have one peer entry and match
 an IP range instead of a single host. That should tidy up the dialplan.
 
 What I'm a little afraid of is the SIP provider using IPs out of a range
 that they also use for other services. Maybe out of the same range they
 hand out IPs to their customers. I guess we got to be careful :-)
 
 Kind regards,
 Sebastian
 
 The Asterisk is local behind a NAT with a firewall, following settings
 are used:
 
 externhost with DynDNS stun with stun.t-online.de
 http://stun.t-online.de/ nat=yes srvlookup=yes allowguest=no
 trustrpid=no insecure=invite qualify=yes
 
 Thank you!  Daniel
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Update peer IP address

2015-03-31 Thread Daniel Heckl

Maybe someone could elaborate on my first question again.

If the ip address changes while a REGISTER period, the ip address of the peer 
isn't been updated. How can asterisk update the ip address of the peer?

 Am 31.03.2015 um 12:36 schrieb Daniel Heckl daniel.he...@gmail.com:
 
 Hello Sebastian,
 
 I had already seen this list of the hosts, but it is not active. All servers 
 with which my Asterisk has been communicated are not listed.
 
 A port scan, to eventually update the list, found hundreds of servers 
 provided in the address range 217.0.0.0/13 with open port 5060, some were 
 even not found. I think there must be another solution.
 
 If I change insecure to insecure=port,invite - could that be a solution?
 
 Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no 
 problem)? Has there anyone experience with dynamic ip addresses of Asterisk?
 
 Daniel
 
 Am 30.03.2015 um 20:09 schrieb Sebastian Kemper sebastian...@gmx.net:
 
 On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote:
 Hello
 
 I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom
 Germany. We have sometimes problems with incoming and outgoing calls.
 I hope I can explain it understandable.
 
 Hello Daniel,
 
 I'll find myself in the same situation a few weeks from now :-)
 
 
 For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de
 http://tel.t-online.de/), the message is answered with OK and the
 peer is registered.
 
 Usually INVITES comes now from this ip address. All works fine. But
 sometimes INVITES comes from an other IP address, for example
 217.0.23.100. This request Asterisk responds with 401 Unauthorized.
 
 In the next register procedure REGISTER are sent to the new ip address
 and answered also with OK. But qualify OPTIONS are continue be sent to
 the old ip address. Incoming and outgoing calls are canceled. Outgoing
 calls are answered with Forbidden.
 
 Even if the REGISTER procedure works with the new ip address, the
 peers are connected with the old address.
 
 Waiting doesn’t help, only a „sip reload“ update the ip address of the
 peer. 
 
 What is the solution for this problem? How can asterisk update the
 peer?
 
 I think the solution - for the inbound issue at least - could be to add
 more hosts as a peer. Have a looks at this forum post:
 
 http://www.ip-phone-forum.de/showthread.php?t=268787p=1999371viewfull=1#post1999371
 
 The user used a template and than he added peers, each with its own IP
 address. The provided list was last updated in 2014, though, so I assume
 the provider in the meantime has added to that list.
 
 It looks pretty tedious, though, I mean there could be dozens of IPs
 you'd have to add. But I guess this is the way to go with Asterisk 11
 and chan_sip.
 
 The future looks brighter :-) I read that with pjsip, which I understand
 is the replacement for chan_sip, you can have one peer entry and match
 an IP range instead of a single host. That should tidy up the dialplan.
 
 What I'm a little afraid of is the SIP provider using IPs out of a range
 that they also use for other services. Maybe out of the same range they
 hand out IPs to their customers. I guess we got to be careful :-)
 
 Kind regards,
 Sebastian
 
 The Asterisk is local behind a NAT with a firewall, following settings
 are used:
 
 externhost with DynDNS stun with stun.t-online.de
 http://stun.t-online.de/ nat=yes srvlookup=yes allowguest=no
 trustrpid=no insecure=invite qualify=yes
 
 Thank you!  Daniel
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Update peer IP address

2015-03-30 Thread Daniel Heckl

Hello

I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have 
sometimes problems with incoming and outgoing calls. I hope I can explain it 
understandable.

For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de 
http://tel.t-online.de/), the message is answered with OK and the peer is 
registered.

Usually INVITES comes now from this ip address. All works fine. But sometimes 
INVITES comes from an other IP address, for example 217.0.23.100. This request 
Asterisk responds with 401 Unauthorized.

In the next register procedure REGISTER are sent to the new ip address and 
answered also with OK. But qualify OPTIONS are continue be sent to the old ip 
address. Incoming and outgoing calls are canceled. Outgoing calls are answered 
with Forbidden.

Even if the REGISTER procedure works with the new ip address, the peers are 
connected with the old address.

Waiting doesn’t help, only a „sip reload“ update the ip address of the peer. 

What is the solution for this problem? How can asterisk update the peer?

The Asterisk is local behind a NAT with a firewall, following settings are used:

externhost with DynDNS
stun with stun.t-online.de http://stun.t-online.de/
nat=yes
srvlookup=yes
allowguest=no
trustrpid=no
insecure=invite
qualify=yes

Thank you!
Daniel-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice recognition IVR Is it possible?

Re: [asterisk-users] Voice recognition IVR Is it possible?

Re: [asterisk-users] Voice recognition IVR Is it possible?

Re: [asterisk-users] PJSIP Stun/ICE

Re: [asterisk-users] PJSIP Stun/ICE

Re: [asterisk-users] Transfer calls "on demand"

Re: [asterisk-users] Transfer calls "on demand"

Re: [asterisk-users] Update peer IP address

Re: [asterisk-users] Update peer IP address

Re: [asterisk-users] Update peer IP address

Re: [asterisk-users] Update peer IP address

Re: [asterisk-users] Update peer IP address

Re: [asterisk-users] Update peer IP address

Re: [asterisk-users] Update peer IP address

Re: [asterisk-users] Update peer IP address

Re: [asterisk-users] Update peer IP address

[asterisk-users] Update peer IP address

17 matches

Site Navigation

Mail list logo

Footer information