Sipgate Outgoing Calls



i am trying to setup sipgate gateway

i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line

-- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" ;tag=as30eb9dd1' -- SIP/sipgate-0000014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)

here is my sip.conf file

[general] port = 5060 bindaddr = context

Asterisk Users 2 years ago 11 Answers

Sip Trunk Failing To Register Causes Sip Phones To Become Unreachable



I have only seen this problem when using SIPgate SIP trunks which actually "register". If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or 1.8 version of asterisk I've tried. Is there a work around that doesn't involve putting an opensips server between the asterisk server and the sip trunk?


Regards, John

Asterisk Users 3.1 years ago 4 Answers

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 3.2 years ago 3 Answers

Missing voicemail prompt beginning


Hello, I am using the voicemail module of asterisk. When I did some test calls
from my mobile phone, sometimes the beginning of the prompt was missing,
e.g. instead of something like "number 12345 not available" I was only
hearing "345 not available". Verbose level 5 on the asterisk console didn't
give me any hint on this, it only shows that playback of the prompt started
correctly in every test case. Any hints on how I can debug this? I think
it's some problem on my local configuration, I doubt it's a problem with…

Asterisk Users 3.3 years ago 9 Answers

asterisk not connecting to sipgate / NAT related issue?


I configured asterisk in sip.conf like that: ===== register => [sipgate-out]
canreinvite=no ===== But all I get on CLI is: [Jan 19 15:59:55] NOTICE[1928]: chan_sip.c:12104 sip_reg_timeout: --
Registration for '' timed out, trying again (Attempt #1) My first impression was that there must be a NAT related issue, so I
captured some SIP debug packets and got: ===== Retransmitting #6 (NAT) to
Via: SIP/2.0/UDP;branch=z9hG4bK4c5e9a7f;rport

Asterisk Users 3.7 years ago 0 Answers

rtp.conf and Asterisk as a sip agent/client


Hello list, I use Asterisk with one trunk. Asterisk connects to as a sip agent/client (with "register =>" statement in
sip.conf). If I restrict the number of ports used in rtp.conf (to 10000-10005 for
example) - will that affect the sip sessions to as well -
or only those sessions where Asterisk acts as a sip proxy/server? Many thanks, Sebastian

Asterisk Users 4 years ago 1 Answer

Sipgate, Yate and Astricon


Hi All, Today on the VUC, we'll be welcoming Sipgate CEO Thilo Salmon back to
tell about their choice of partners in their latest services. I will be announcing the first #VUC VoIP & Tell discount code for
Astricon in Denver, October 25-27. Join us on using g722 or g711 codec or see for the other ways to connect. Hear you there! /r

Asterisk Users 4.3 years ago 0 Answers