Displaying 20 results from an estimated 1000 matches similar to: "Silly incoming SIP failure"
2007 Mar 01
3
UK SIP Gateway
Hi,
Now that I have Asterisk up and running I would like to find a good SIP gateway in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I am looking for peoples recommendations.
Apologies if this is the incorrect forum for this type of request.
Regards,
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2006 Jan 24
8
UK Provider
Hi
Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account.
www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations.
Many Thanks
Scott
2004 Dec 26
2
Asterisk behind IX66
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2004 Apr 14
1
Most Reliable Proxy Server?
Hi all,
Do you know if there's any free public SIP proxy server that is more reliable that FWD and Iptel?
Thanks
Ron
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2005 Feb 09
1
Re: Newbie help/pointers required -configure xlite with asterisk
Unfortunately I seem to have another problem!
I am using sipgate for the incoming line - and it appears that you cannot
get DTMF to work in that configuration. Unless anyone knows anything
different of course!!
> > I just want one of my incoming numbers to go to an IVR service that
> > will allow me to select what I want.
> >
> > For example
> >
> >
2005 May 28
2
UK DID providers
Hi
Can anyone provide me with a Manchester (0161) UK DID number, preferably
IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume
will be low.
The critical thing is that DTMF must be correctly passed 100% of the time,
unlike Sipgate, my current (free) provider, whose DTMF detection/passing is
not at all reliable, making it useless for a virtual receptionist scenario.
I
2005 Feb 07
2
Record() cut off after 40 sec
Hi,
i am recording a message, but it is always cut off at 40 secs.
There are no time out configured.
Gabriel
--
The educated person is not the person who can answer the questions but
the person who can question the answer.
2004 May 28
5
SIP Changes???
Hi Everybody
Any significant changes to CVS HEAD over the last couple of days. I've got
two asterisk boxes - both on public IP but one is dynamic. The one on
dynamic IP registers at the other one - that part is fine.
Calls going from the one with dynamic to the static one goes fine.
Call the other way results now in:
Failed to authenticate user "1101"
2009 Jan 06
5
Queue
Hi,
I was asked to create a Queue which instead of playing MoH it generates
the ringing tone. I had a look around but could find anything, I would
welcome and help.
Regards
Mateusz
2004 Jun 23
1
Codecs and pauses
Hi all
My * implementation is working brilliantly with only one small fault left to
kill.
I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the
pstn network; if I set my codec to GSM everything works great - no pauses
but quality is a bit poor. If it set the codec to alaw (I think I'm using
the correct one - I'm in the UK) I get intermittent pauses on the call.
2005 May 13
1
broadvoice replacement
Does anyone know of a BYOD provider that terminates calls to NCFA
numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those
numbers, but this is getting silly now, it doesnt work and no answer if
after switching to a new provider it will ever work.
Can anyone suggest an alternative provider that serves NCFA?
--
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605
2005 May 16
4
Asterisk@home 1.0 + Sipgate UK/SIP Provider
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
"8|." to place all calls with a 8 prefix tot he sipgate account the
softphones dial the number, the Asterisk
2013 Sep 18
2
sipgate outgoing calls
Hello
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 at sipgate
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on INVITE to
'"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1'
--
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
ping times are fine as well:
[root at freepbx asterisk]# ping sipgate.de
PING sipgate.de (217.10.79.9) 56(84) bytes of data.
64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57
2006 Feb 25
2
sipgate.de question
Hi,
Anyone here using sipgate.de ?
It worked for months, but for a couple of days now I'm
unable to register with them.
My account is ok, because I can login to the website.
Asterisk keeps showing me:
Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n)
I looked at the sip debug stuff, and all I
2004 Jun 07
3
Voip-talk?
Hi everyone
I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI analog problem. (see [Asterisk-Users] Multiple DDI & Hunting on Analog Lines (UK) for details) Does anyone on the list have any recent comments on reliability etc? I would really appricated some positive and negative comments.
Cheers
Matt
2004 Sep 26
6
SIP Registration Timeout, No FW
Hi people,
My asterisk wont register with any sip providers, I have tried three
different but they all end up with:
Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout:
Registration for 'whatever@provider.tld' timed out, trying again
There is no firewall and my server has a public IP. Could this be a Asterisk
problem?
-Fredrik vK
2009 Oct 18
1
Call from 'sip-id' to extension 'sip-id' rejected because extension not found ?
I'm trying to setup sipgate on 1.6.1. Following the instructions on the
site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk,
I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf:
[sipgate]
type=friend
secret= ;;SIP_PASSWORD
insecure=port,invite
defaultuser= ;; SIP-ID
fromuser= ;;SIP-ID
context=sipgate_in
fromdomain=sipgate.com
host=sipgate.com
2006 Oct 25
2
Call is not coming through sipgate.co.uk+Asterisk
Hi,
I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx.
I configured my Asterisk server with 0207100xxxx. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
Hi all,
I have an issue with asterisk 13 and pjsip. I guess it is somehow
Firewall related, but I'm unsure.
A registration to Sipgate is established successfully:
<Registration/ServerURI..............................> <Auth..........>
<Status.......>
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