Error SIP/2.0 488 Not acceptable here


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 - 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] My sip.conf including the codec restrictions looks like this…

Asterisk Users 3.4 years ago 3 Answers

No progress tones on transferred call


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
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 ->
INVITE sip:1CDF0F4AFFFF@ SIP/2.0..Via: SIP/2.0/UDP;branch=z9hG4bK5286810e;rport..From: "C Allerid"

Asterisk Users 3.4 years ago 0 Answers

Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error


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
( 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]
dtmfmode=inband We aren't registered, our IP is authorized by their system. Debug of sessions ( is our server…

Asterisk Users 3.4 years ago 7 Answers

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


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 test,
the INVITE succeeds. on [test]
context=demo on : [natty]
secret=abcde originate SIP/natty/1234568 extension 200
== Using SIP RTP CoS mark 5
Audio is at port 19486

Asterisk Users 3.5 years ago 2 Answers

SIP and NAT best practices in Asterisk


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 3.7 years ago 0 Answers