Displaying 20 results from an estimated 2000 matches similar to: "Anyone using sipcall.co.uk ?"
2003 Nov 03
1
Asterisk compliance with RFC 2617 (qop, nc and cnonce) - in relation to sipcall.co.uk
Hi All,
I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex
kit to provide the SIP service. Unfortunately Asterisk cannot seem to
authenticate against Intertex. Having provided SIP debug info the
provider has informed me that Asterisk does not appear to support 'qop',
'nc' and 'cnonce' which are used to stop replay attacks.
So, does Asterisk support
2004 Mar 30
3
Sipcall.co.uk & [*]
Hello all.
Has anyone managed to get SIPCALL.co.uk's service working with the [*] box?
I've managed to register with other SIP providers but not SIPcall.
The debug just show's [*] attempting to register.
But receiving a 401 error everytime.
Cheers
Matt
2003 Sep 30
1
SIP Registration Difficulties
I have SIP registrations working correctly for FWD and Sipphone, but it
is impossible to connect to Sipcall or Nikotel, I saw that someone on
the list has problems with ICH.
To try and sort out the problem I tried to register to Sipcall with
Linphone and sent the dialogs to tech support of the equipment provider.
Here is their answer:-
The reason the registration fails is because not
2003 Nov 04
1
Does anyone provide inbound UK numbers using IAX ?
Hi All,
Is there anyone providing UK geographic numbers that can be terminated
on Asterisk using IAX ? It must be a geographic number (eg. Start 01 or
02, not 08xx). I've tried the sipcall.co.uk service and it looks very
good when using X-Lite but it will not work with Asterisk. Switching to
IAX should also resolve issues around NAT - hurray!
-Nathan
2003 Aug 06
4
New SIP Phone
Michael Robertson, founder of both MP3.com and Lindows, has launched a
new company to supply inexpensive SIP phones ($129 for two) and related
services. See today's press release at
http://www.sipphone.com/tiki-index.php?page=SIPphone%20Inc
My question for the list is who will be the first to report on the
compatibility and usability of the SIPphone with Asterisk? The
functionality
2004 Apr 29
0
SIPCALL and [*]
Sorry to bug the entire list with this as this is really a question for
those who have been sucessful in configuring [*] to place and receive a
SIPCALL call.
Everying looks right in my config, I can see it registered etc but when I
try to place the call I get:
-- Executing Dial("SIP/100-2371", "SIP/8703409095@sipcall/04") in new stack
Apr 29 22:50:34 WARNING[27089840]:
2003 Aug 12
1
Working with FWD, IPTel, SIPPhone?
I'll admit it. I'm a asterisk newbie (but no stranger to telephony).
The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone
specials) on a private segment calling to a Linux box acting as the
segment's firewall with a leg on our public network. The phones are
setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks
to the Asterisk HOWTO).
Getting IAX
2004 Apr 06
1
SIP phone registering problem
I am clearly doing something ridiculously wrong.
Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are
unable to register. They keep trying and then time out.
With the sip debug on in Asterisk nothing is logged.
Here is the trace from one of the phones (kphone):
(192.168.100.13 is kphone, 192.168.100.3 is Asterisk)
sipclient: sending: 21:47:45.454
2008 Nov 18
1
Asterisk 1.4.21.2 and gtalk2voip
Hi,
Ii try to connect an Asterisk server running 1.4.21.2 version with
gtalk2voip services. Everything is fine till the call for DTMF test:
there is no audio and Asterisk shows
[Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
SIPCALL-435578583-1984100284 at 72.20.112.114 for seqno 1 (Critical Response)
[Nov 18 14:51:47] WARNING[20502]:
2009 Mar 30
2
no ringtone - just silence/bridging of external calls
Hi Community!
If this issue was already topic, please excuse or delete my request...
Topic 1 "no ringtone":
I configured a SIP registration with my SIP provider (SIPCall).
Everything works fine except the ring tone for the caller. The caller
hears silence until the called party takes up the phone.
I used the DIAL command with the r and R option but no luck... :(
Has anybody the same
2014 Feb 03
1
Incoming Fax Issue with Asterisk 11.7 and Digium Fax
Hi, im using a Asterisk Server which is not behind NAT.
First i had problems with the fax detection. But this is now solved
after adding a wait(2) at the correct place. But i'm still unable to
receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too
short after the Fax session has started.
My sip.conf includes
[general]
allowguest=no
alwaysauthreject=yes
sendrpid=rpid
2003 Oct 06
5
Help with questions for initial Asterisk wizard (GUI)
Hey all,
I am in the middle of creating a new user wizard which will generate all
the .conf's the new Asterisk user will require to get themselves up and
running in Asterisk without having to touch a single configuration file.
This is what I have come up with as a rough draft. It is far from
complete, so I'm asking people to submit things that should be added,
changed, removed
2003 Aug 20
1
IAX <> IAX trunking... DP cache?
I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one
another using IAX/IAX2 trunks.
I've managed to get a semi-functional NAT Firewall working as a PBX
(with Asterisk running directly on the firewall itself), but there are
issues with bind()ing to various interfaces which is causing outbound
SIP issues.
To get around these issues, the idea is to do something like
2004 Aug 23
4
Asterisk WITH Swyx... Any Idea?
Hi,
I'm a student and my thesis work consist in testing
Asterisk with Swyx(SwyxWare).
My approach is to declare asterisk as h323 gateway for
the Swyxserver using oh323 Plugin.
Is there any possibility to connect Asterisk with
Swyx? how?
the outgoing call must pass from Swyxit->to
Swyxserver-> Asterisk->to PTSN
Thanks
2009 Nov 09
3
E1 Extensions.conf
Hi,
I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports
I want to make a loop test between digium card E1 to test the
configuration of dahdi
What I want to do scenario is
I connect port 1 and port4 in the digium card with E1 cable
SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local.
kindly can any can help me to
2006 Mar 24
3
Call terminated after 60 seconds
Hello,
I switched from my PSTN provider to a voip provider. (Voicedata in
the Netherlands)
>From the moment i switched all inbound calls are terminated after
aproximatly 1 minute.
The provider tells me it's not their issue since I have no other
configuration than all their other users.
What can I do.
I removed all asterisk functionality by forwarding the inboud call
directly to a local
2003 Nov 23
1
SIP Asterisk -> Nikotel disconnects after 1 Minute
Hello list!
I'm using "Asterisk CVS-11/22/03-04:28:51" and try to route my normal
(classic) phone calls via nikotel (www.nikotel.com). I can talk about 1
minute and get then disconnected. Here my current configuration parts
which affect nikotel:
register => chabrol:PASSWORD_REMOVED@nikotel/500
[nikotel]
type=friend
secret=PASSWORD_REMOVED
username=chabrol
fromuser=chabrol
2003 Oct 03
9
No Ringback on Iconnect
When I place a call using Iconnecthere as my sip provider, I hear no
ringback when making a call. Does anyone else have this problem or
offer any suggestions? Thanks, Kevin
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2005 Jan 10
2
Some questions (maybe Nikotel related)
Hi list,
I have some nontrivial questions. I am no telecommunication guru and I
will explain it with my simple words. I hope someone can help me with
these issues (with Asterisk 1.0.3):
- If I call outside (with Nikotel to German Telekom) there is a remote
hangup after 2 minutes. I've seen other people posting this but nothing
helped. I luckily managed to get around this issue with the
2020 Sep 30
4
some domains resolving issues
Hello.
I have two records in dialplan:
exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org)
exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org)
Calling testA works fine while testB fails with "CONGESTION".
Adding debug for console shows that pjsip_resolver.c does
`New queries added, performing parallel resolution again`
for linphone after