Can someone tell me what is this issue ?

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Your Server Voipon isn't responding- See if internet is working fine, or
your Voipon sip trunk/peer is registered OK? On Fri, Feb 3, 2012 at 5:53 PM, virendra bhati wrote: > Call is not routing from server to destination.
>
>
> app8*CLI> console dial 00918885268942
>
> [Feb 3 06:01:15] NOTICE[28124]: console_video.c:133 console_video_start:
> voice only, console video support not present
>
> -- Executing [00918885268942@default:1] Answer("Console/dsp", "") in
> new stack
>
> < < Console call has been answered >>
>

Asterisk Users 3.5 years ago 0 Answer

SIP trunk call initiated as Anonymous@anonymous.invalid

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I have a Grandstream HT-502 device connected to my Asterisk PBX. It is
configured not to place anonymous calls, and from my mostly layman
reading of the invitation that the device sends, it should not be
anonymous. However, the Asterisk PBX sends an anonymous invitation to
our SIP trunk provider. Can anyone explain why? The two INVITE packets follow. The devices sends the following INVITE: INVITE sip:2223334444@pbx.xxxxx.com SIP/2.0
Via: SIP/2.0/UDP 192.168.9.197:46538;branch=z9hG4bK526774101;rport
From: "222333555" ;tag=2072922124
To:
Call-ID: 1082640776-46538-3@BJC.BGI.J.BJH
CSeq: 21 INVITE
Contact: "222333555"
Authorization:…

Asterisk Users 3.5 years ago 0 Answer

Possible remote enumeration of SIP endpoints with differing NAT settings

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Asterisk Project Security Advisory - AST-2011-013 Product Asterisk
Summary Possible remote enumeration of SIP endpoints with
differing NAT settings
Nature of Advisory Unauthorized data disclosure
Susceptibility Remote unauthenticated sessions
Severity Minor
Exploits Known Yes
Reported On 2011-07-18
Reported By Ben Williams
Posted On
Last Updated On December 7, 2011
Advisory Contact Terry Wilson
CVE Name Description It is possible to enumerate SIP usernames when the general
and user/peer NAT settings differ in whether to respond to
the port a request is sent from or…

Asterisk Users 3.6 years ago 3 Answer

How can I Add my own Word in option packets in from field of SIP "From Asterisk??"

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Hello All,
Is there any one who can help me to change the From field parameters in
option packets, I have seen that in option packtes asterisk sends its own
information,If you see the below option packet i have highlighted the
asterisk word in from field and in from field tag how can i changed it
Please let me know same as in User Agent.
192.168.207.70:5060 -> 192.168.207.177:5065
OPTIONS sip:192.168.207.177 SIP/2.0..Via: SIP/2.0/UDP 192.168.207.70:5060
;branch=z9hG4bK57e5b165;rport.*.From: "asterisk" 192.168.207.70>;t
ag=as0977f8f5..To: ..Contact: <
sip:asterisk@192.168.207.70>..Call-ID:
272c85316b257dfa168c9d0155089b8a@192.168.207.70..CSeq: 102…

Asterisk Users 4 years ago 2 Answer

Issue with Asterisk & Aastra 57i at v3.2

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On 05/05/11 04:37, Richard Kenner wrote:
> I recently tried to update my Aastra 57i to version 3.2 and ran into
> a problem. It won't properly register and says "contact mismatch".
> I added "sip contact matching: 2" to aastra.cfg, but that didn't help.
>
> When I look at the SIP trace, but I see is the Aastra sending a
> REGISTER and Asterisk replying with the 401. The phone then sends
> the REGISTER again, this time with the hash. Asterisk now replies OK,
> but sends an OPTION…

Asterisk Users 4.2 years ago 0 Answer