Re: [asterisk-users] ST2030 replacement

2016-01-07 Thread Frank 
On Thu, 2016-01-07 at 17:35 +0100, Sil wrote:

> Can you give me a return on the models you use ?

Yealink T26P




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Re: [asterisk-users] placing calls with linphone.org SIP account

2016-01-07 Thread Frank 
On Wed, 2016-01-06 at 11:27 +0100, Yves wrote:

>  how can I call other users 
> registered at other SIP-Providers?
> I tried all well-known SIP URI Syntaxes but none worked... does anyone 
> reliably know, if it is possible at all and if so, what is the
> dialstring looking like? 

It depends if the "other SIP-Provider" accepts calls from other
networks. 
Some providers accept calls from other networks. Unfortunately, most
providers do not. 



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[asterisk-users] ST2030 replacement

2016-01-07 Thread Sil 

Hello,
I am looking for a replacement for my Thomson ST2030SIP.
My specifications are as follows :
- 2 lines.
- 6 BLF keys.
- PoE.
Can you give me a return on the models you use ?
Thanks.

Sil

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Re: [asterisk-users] ST2030 replacement

2016-01-07 Thread Markos Vakondios 
Grandstream GXP-1628

On 8 January 2016 at 01:00, Frank  wrote:

> On Thu, 2016-01-07 at 17:35 +0100, Sil wrote:
>
> > Can you give me a return on the models you use ?
>
> Yealink T26P
>
>
>
>
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Re: [asterisk-users] The To header was truncated in call... Whats this means?

2016-01-07 Thread Yves 
I have seen these messages only on asterisk boxes that are open to 
public and I think this may have something to
do with sip-attacks... I´d recommend some wiresharking or at least sip 
debugging...


yves

Am 07.01.2016 um 21:23 schrieb Vitor Mazuco:

Hi everybody,

My Asterisk, all time appear this log

[Jan  7 15:37:04] ERROR[1174] chan_sip.c: The To header was truncated
in call '6c66e5b6058ae257003c0f7e778da0fe@191.x'. This call
setup will fail.
[Jan  7 15:37:18] ERROR[1174] chan_sip.c: The To header was truncated
in call '18e0a12e434364254b0cc2e52d20755b@191.x. This call setup
will fail.
...

Whats this massege means?

Thanks.




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[asterisk-users] The To header was truncated in call... Whats this means?

2016-01-07 Thread Vitor Mazuco 
Hi everybody,

My Asterisk, all time appear this log

[Jan  7 15:37:04] ERROR[1174] chan_sip.c: The To header was truncated
in call '6c66e5b6058ae257003c0f7e778da0fe@191.x'. This call
setup will fail.
[Jan  7 15:37:18] ERROR[1174] chan_sip.c: The To header was truncated
in call '18e0a12e434364254b0cc2e52d20755b@191.x. This call setup
will fail.
...

Whats this massege means?

Thanks.

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Re: [asterisk-users] Getting Asterisk to use the SIP Path header

2016-01-07 Thread Peter Baines 
If anyone else is having this issue, asterisk 1.8.32.3 uses the Path header
as expected, if you want to follow progress I've created a bug report:
https://issues.asterisk.org/jira/browse/ASTERISK-25666


On 6 January 2016 at 10:29, Peter Baines  wrote:

> Hi,
>
> How do I get asterisk to use the SIP Path header value from registrations
> when calling devices?
>
> I am trying to use opensips as a proxy for asterisk, when a client
> registers I am adding the Path header before forwarding the REGISTER onto
> asterisk. The problem is when asterisk recieves an INVITE it does not use
> the value from the Path header, it is sending directly to the device. Can
> anyone point me in the right direction as to why?
>
> I am using asterisk 13.6.0 with the default configuration, the changes I
> have made are:
>
> In sip.conf I have uncommented:
> supportpath=yes
> rtsavepath=yes
>
> In users.conf I have:
>
> [6000]
> secret =
> host=dynamic
> context = default
>
> [6001]
> secret =
> host=dynamic
> context = default
>
> [6002]
> secret =
> host=dynamic
> context = default
>
> In extensions.conf I have made default like:
> [default]
> ;include => demo
> exten => 6000,1,Dial(SIP/6000,18)
> exten => 6000,n,Hangup()
>
> exten => 6002,1,Dial(SIP/6002,18)
> exten => 6002,n,Hangup()
>
> exten => 6001,1,Dial(SIP/6001,18)
> exten => 6001,n,Hangup()
>
>
> Below is the 6000 user REGISTER going from opensips (10.15.20.137:5060)
> into asterisk (192.168.68.68:5070) with the Path header.
>
> U 2016/01/06 10:04:23.399170 10.15.20.137:5060 -> 192.168.68.68:5070
> REGISTER sip:10.15.20.137 SIP/2.0.
> Via: SIP/2.0/UDP 10.15.20.137:5060;branch=z9hG4bKcc2c.b40fb511.0.
> Via: SIP/2.0/UDP 10.15.20.53:52666
> ;received=10.15.20.53;branch=z9hG4bK-d8754z-91422161f08a7943-1---d8754z-;rport=52666.
> Max-Forwards: 69.
> Contact: .
> To: .
> From: ;tag=9e95da50.
> Call-ID: OTQ1ZTdmZmE3OTM1ZWVkYzMzYWZiMDMzMDgyODhmOTU.
> CSeq: 2 REGISTER.
> Expires: 3600.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO.
> User-Agent: Bria 3 release 3.5.5 stamp 71243.
> Content-Length: 0.
>
>
>
> *Path: .*
> Below is the INVITE going from opensips to asterisk for 6000
>
> U 2016/01/06 10:11:13.668929 10.15.20.137:5060 -> 192.168.68.68:5070
> INVITE sip:6000@10.15.20.137;transport=UDP SIP/2.0.
> Record-Route: .
> Via: SIP/2.0/UDP 10.15.20.137:5060;branch=z9hG4bK8f77.9d6ef7e7.0.
> Via: SIP/2.0/UDP 188.39.51.2:35631
> ;rport=35631;received=10.15.20.53;branch=z9hG4bK-d8754z-d46f3a0333dc5d49-1---d8754z-.
> Max-Forwards: 69.
> Contact: .
> To: .
> From: ;tag=870fdf72.
> Call-ID: YWVjN2VjMDZmYmZmNjg4MTE2MzJlZGU1ZDNjZGU2NDc..
> CSeq: 2 INVITE.
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE.
> Content-Type: application/sdp.
> Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri.
> User-Agent: Z 3.3.21933 r21903.
> Allow-Events: presence, kpml.
> Content-Length: 237.
> .
> v=0.
> o=Z 0 0 IN IP4 188.39.51.2.
> s=Z.
> c=IN IP4 188.39.51.2.
> t=0 0.
> m=audio 8000 RTP/AVP 3 110 8 0 98 101.
> a=rtpmap:110 speex/8000.
> a=rtpmap:98 iLBC/8000.
> a=fmtp:98 mode=20.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=sendrecv.
>
>
> I would now expect asterisk to send the INVITE to the value of the Path
> header in the registration (10.15.20.137:5060) however it is sending the
> INVITE directly to the device (10.15.20.53:52666):
>
> U 2016/01/06 10:11:13.671345 192.168.68.68:5070 -> *10.15.20.53:52666
> *
> INVITE sip:6000@10.15.20.53:52666;rinstance=d4284982f7c18786 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.68.68:5070;branch=z9hG4bK308a4ef5;rport.
> Max-Forwards: 70.
> Route: .
> From: "New User" ;tag=as3daea415.
> To: .
> Contact: .
> Call-ID: 55202bc71f9e684d0b82c7cb2e8684ab@192.168.68.68:5070.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX 13.6.0.
> Date: Wed, 06 Jan 2016 10:11:13 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces, timer, path.
> Content-Type: application/sdp.
> Content-Length: 286.
> .
> v=0.
> o=root 887525354 887525354 IN IP4 192.168.68.68.
> s=Asterisk PBX 13.6.0.
> c=IN IP4 192.168.68.68.
> t=0 0.
> m=audio 12356 RTP/AVP 0 8 3 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=maxptime:150.
> a=sendrecv.
>
>
> Thanks,
> Peter
>
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Re: [asterisk-users] Virtual domain redirects

2016-01-07 Thread D'Arcy J.M. Cain 
On Thu, 7 Jan 2016 00:04:05 -0500
"D'Arcy J.M. Cain"  wrote:
> I am trying to figure out how to allow da...@example.com to be
> translated to dc2...@vex.net (out ISP domain) but I am at a loss to

I think I see where I can hook this.

same => n,Verbose(0,To: ${SIP_HEADER(To)})

This shows me the actual address called.  e.g.
"To: ".  Now I just need to look for a
place to grab this and put in a GoTo.  Perhaps a new incoming context
that does the lookup and then jumps to the existing incoming context.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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