Displaying 20 results from an estimated 20 matches for "nexton".
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newton
2007 Nov 15
1
Help on strange problem...
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hey all,
I'm having problems with calls dropping after 15 - 20 seconds from a
particular provider. The are using a NexTone gateway. Here are the details:
Successful call:
INVITE cseq 1 From NexTone
100 Trying cseq 1 From Asterisk
100 Trying cseq 1 From Asterisk
200 OK (G711U) cseq 1 From Asterisk
ACK cseq 1 From NexTone
INVITE (G711U) cseq 2 From NexTone
100 Trying cseq 2 From Asterisk
200 OK cseq 2 From A...
2005 Jul 27
1
Question about Nextone softswitch
...our local DID's in Seattle
2. comes into our Asterisk server in Los Angeles or Denver
3. is routed by Asterisk for termination back to a different Seattle
PSTN
....and if our VOIP call termination provider requires (in order to get
their best rate) all calls to go through their Nextone softswitch in Dallas
before ultimately terminating at the desired Seattle PSTN line...
What is the resulting affect as it relates to any difference in "user
experience" for the caller in Seattle....and what, if any, is the cost difference
on our end due to the extra hop?
CHEE...
2008 Jun 25
0
Res: Asterisk with Nextone using H323
Thank`s all,
Chris Ziomkowski wrote:
> If you only want to use H.323 with Asterisk, you should configure it
as an H.323 gateway.
> Why are you trying to set "softswitch"?
I was asked by a costumer, because he could not use a asterisk as a
softswitch in the Nextone configuration, so I`m looking for the
difference in asterisk configs files.
Thank`s a lot for you help...
Everton Goularth
2009 Jan 10
3
Asterisk/GXW410x IP Analog Gateway
Hello All,
I am trying to setup a small system where Nextone Softswitch will send
traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog
Gateway but for some odd reasons the call are flashed back from
Grandstream to Asterisk and creating a Black loop...
I did follow the instructions provided by Grandstream support but it
doesn't se...
2007 Nov 30
3
How to setup redundant SIP peers
Hello list,
I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:
[peer1]
type=peer
host=10.10.10.1
[peer2]
type=peer
host=10.10.10.2
Now dialout is no problem. Extensions.conf says:
exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30)
But how can I setup a failure-route if the SIP-Proxy "peer1" ist not
2005 Mar 11
1
SIP signalling and RTP to different servers
Hello,
we're in process of testing an interconnection with a trans-european
carrier. But the carrier wants the SIP signalling to server 1 and the
RTP stream to server 2. How do I configure asterisk to work with that
type of installation. It seems they are using NexTone as SIP Signaling
and RTP servers. Can someone help me???
Regards,
Marc
--
CTO Marc Storck
MS Networks SA mstorck@msnetworks.lu
IT Service Provider http://www.msnetworks.lu
15, route d'Esch Phone: +352 2727 3030
L-4450 Belv...
2010 Mar 21
1
Asterisk Died - Ver-1.6.2.6.
...afe_asterisk" just sent me an email saying "Asterisk on bill exited on
signal 11. Might want to take a peek.". Looking at the
/var/log/asterisk/message doesn't show me anything...
This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and
it is routing calls from Nextone MSW Softswitch to VPS Softswitch...
Any reason why Asterisk died?
Thanking in advance...
Cheers,
Nitesh
2010 Apr 19
0
RTP Timeouts not clearing calls
...systems when RTP timeouts occur.
It appears that asterisk doesn't send a BYE when it decides to terminate a
call because of a RTP Timeout - is this a configuration problem? if so what
need changing? or is this a bug/feature? if so is there a work around?
The setup here is calls come from our Nextone softswitch and connect to
remote Asterisk boxes and this all works fine unless the Asterisk box
decides there is an RTP timeout - it clears the call down as far as the
remote (asterisk) end is concerned but the Nextone never sees the BYE so
holds the channel up until a Max-Duration timer clears i...
2014 Apr 14
1
how to configure callcentric peer
...ctelnum>@ss.callcentric.com>
> i: 18075985-3606475083-968100 at msw2.telengy.net
> CSeq: 1 INVITE
> Max-Forwards: 8
> m: <sip:f1eb8ab7586b3f2b72742b5e4d43d78d at 204.11.192.161:5060;transport=udp>
> Supported: timer
> c: application/sdp
> l: 350
>
> v=0
> o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.161
> s=sip call
> c=IN IP4 204.11.192.161
> t=0 0
> m=audio 50960 RTP/AVP 18 0 8 101
> a=fmtp:18 annexb=no
> a=fmtp:101 0-15
> a=rtpmap:101 telephone-event/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G7...
2005 Oct 04
3
Asterisk as H323 gateway
Is there anyone who is currently using Asterisk as a production H323
gateway?
And using which combination of asterisk and H323 (chan_h323, chan_oh323?)
The main issue is interoperability with other H323 parties (Cisco AS53xx,
Nextone, etc).
Searching the mailing list it seems that both h323 and oh323 are not so
stable, is it only an impression or using h323 is really not so advisable?
Francesco Pellegrini
pellegrini@frameweb.it
2004 Aug 04
5
H323 Call Dropping
Hello All,
I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the
configuration:
CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK
My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk,
and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however
the gatekeeper drops the call
2004 Aug 11
7
H323 call dropped when answered
Hi All.
I'm using RedHat 9
I configured the chan_h323 and asterisk from CVS.
This is the scenario SJ_lab_phone(sip) ---------------> Asterisk
-------------> H323 GK --------------> PSTN
I have tried all codec's and always the same result, the called phone
will ring without dropping for how ever I allow it to but as soon as it
is answered it immediately gets disconnected.
2004 Jul 15
3
SIP to H323 call timeout
Hi all,
I have the following setup:
UAs ------------SER ------------------------ ASTERISK
---------------------GNUGK --------------- GWs
SER is configured to route call requests from UAs to Asterisk. Asterisk is
configured to receive the call on SIP channel and dial out to GNUGK over
H323 channel. The problem I'm facing is that asterisk sends out the call
request to GNUGK and times out
2004 Jan 29
1
WTB E100P or better
If anyone is looking at upgrading their Digium hardware, or has an
excess of any of these cards, please let me know. I am not interested in
non-Digium hardware, but thanks for asking :)
E100P
E400P
TE410P
I only need one card, in that preference order. Would also prefer if the
card is already in Australia. I'm in Sydney/Australia if that matters to
you.
Regards,
Adam
--
Adam Goryachev
2007 Jul 03
1
Session Border Controller time...
Come on you carriers on the list... Give up the dibs what are you using
and why?
About to sledgehammer these SELECT * FROM GARBAGE WHERE SBC = 'nCite'
Don't bother shooting me off Newport Networks stuff... Too pricey
--
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
echo infiltrated.net|sed
2008 Jan 18
0
Maximum retries/no reply to our critical packet
...Response)
Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1245 retrans_pkt: Hanging up
call 20e5feaa-1d3d7ef1 at 192.168.1.2 - no reply to our critical packet.
Customer can receive inbound calls without any disconnections, its just
when he tries to make outbound calls.
All outbound calls are sent to Nextone SoftSwitch and default codec is
G729a. Customer has Linksys SPA-2102 - firmware ver 3.3.6 and Asterisk
version 1.2.18.
Thanking in advance...
Cheers,
Nitesh
2010 Apr 10
2
Sending RTP media to a different server than SIP Signaling
Greetings list
i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with
one SIP Signaling server and Two Media servers ..
googled for a week and didn't find a way to do this.. so my question. is it possible to be done?
Asterisk server 1.4.26.3
_________________________________________________________________
The
2003 Jan 14
0
Oplocks_break FAILURE in 2.2.7? hmm..
...g profiles, but I'm experiencing problems with w2k, NT4/XP
just not test because i'm testing on my box at home and when it'll
work fine then I'll go to clients from which smaller part has XPs
installed on PC. I think if it works on w2k so will be working on XP
too. My problem is nextone: Logon to domain from w2k box is OK(NO
ERROR). Drives are mapped and everything works fine. Just smb_audit
tells that the .Desktop.ini cannot found at directory "." ?!? I'm
ignoring it, but when I logout the system begin process of storing
profile on Samba server and report that...
2005 Mar 11
2
Re: Incoming echo cancel
...gt;
> Hello,
>
> we're in process of testing an interconnection with a trans-european
> carrier. But the carrier wants the SIP signalling to server 1 and the
> RTP stream to server 2. How do I configure asterisk to work with that
> type of installation. It seems they are using NexTone as SIP Signaling
> and RTP servers. Can someone help me???
>
> Regards,
>
> Marc
> --
> CTO Marc Storck
> MS Networks SA mstorck@msnetworks.lu
> IT Service Provider http://www.msnetworks.lu
> 15, route d'Esch...
2004 Nov 02
0
Number of routing tables..255 limit?
Hello,
I have a question about Linux Advanced Routing..Is it possible to have more than 252 user-defined routing tables?
I see the table id is an unsigned char everywhere, is there a reason to have it limited to 255? Can I change it to more than that? Any adverse effects?
Thanks!
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