Error SIP/2.0 488 Not acceptable here

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Hello, a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider
The user calling me is also using Sipgate and is calling my landline phone
number from Sipgate (not [my sip id]@sipgate.de). My sip.conf including the codec restrictions looks like this…

Asterisk Users 3.3 years ago 3 Answers

No progress tones on transferred call

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Asterisk 1.4
We are experiencing an issue on transfers where no progress tones are heard
by the caller:
1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF).
1595 answers
2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears
MoH.
3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595
hears no ringing
When xfer is pressed and the extension is dialled:
U 203.89.001.001:5060 -> 121.98.001.001:1034
INVITE sip:1CDF0F4AFFFF@192.168.1.72:5060 SIP/2.0..Via: SIP/2.0/UDP
203.89.001.001:5060;branch=z9hG4bK5286810e;rport..From: "C Allerid"
;tag=as72616c50..To:
..Contact:
..Call-ID:

Asterisk Users 3.3 years ago 0 Answers

Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

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Hi list, we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
version and face the following problem: one of our peer
(voicetrading.com) doesn't accept our calls anymore, we receive a
timeout error "Packet timed out after 32000ms with no response". Switching back to 1.6 make things working again! In sip.conf we have nat=no, peer conf is: [myPeerDef]
type=peer
host=111.111.1.111
context=honeypot
insecure=invite
directmedia=no
disallow=all
allow=ulaw,alaw
dtmfmode=inband We aren't registered, our IP is authorized by their system. Debug of sessions (222.222.22.22 is our server…

Asterisk Users 3.3 years ago 7 Answers

Can't make Asterisk send authentication to remote peer on INVITE

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This is a really simple problem that I just can't get to work. There
are two Asterisk servers with the following sip user and peer. When a
call is attempted, Asterisk is not sending authentication details in
response to the 401. Note, if the secret is blank on 172.16.0.2 test,
the INVITE succeeds. on 172.16.0.2: [test]
type=friend
secret=abcde
host=dynamic
context=demo on 172.16.0.1 : [natty]
type=peer
host=172.16.0.2
fromuser=test
secret=abcde originate SIP/natty/1234568 extension 200
== Using SIP RTP CoS mark 5
Audio is at 172.16.0.1 port 19486

Asterisk Users 3.4 years ago 2 Answers

SIP and NAT best practices in Asterisk

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What should I do in order to to be as secure as possible and with “clean” logs?

Well, for an article about Asterisk security best practices, consider reading this article. About SIP and NAT best practices, in short, the simplest answer is to always use ‘nat=yes’ (or at least ‘nat=force_rport’ in recent versions of Asterisk that support it), until you come across a SIP endpoint that fails to work properly with that setting. If you do come across such an endpoint, try hard to get it to work with that setting; if you…

Asterisk Tips 3.6 years ago 0 Answers

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 3.6 years ago 0 Answers

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 3.7 years ago 0 Answers

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 3.8 years ago 3 Answers

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 4.2 years ago 2 Answers

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 4.4 years ago 0 Answers