Displaying 7 results from an estimated 7 matches for "spa1000".
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sja1000
2005 Jul 26
2
sip+oh323 - no voice at sip side
...0 100 Trying
To: <sip:48224789000@172.16.13.169:5060>
From: "datacom1234, 224782479 " <sip:224782479@192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351@192.168.0.252
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc
Server: Sipura/SPA1000-2.0.13(g)
Content-Length: 0
8 headers, 0 lines
Sip read:
SIP/2.0 180 Ringing
To: <sip:48224789000@172.16.13.169:5060>;tag=b857823bdad08738i0
From: "datacom1234, 224782479 " <sip:224782479@192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351@192.168.0.252...
2008 Apr 29
1
Annoying Sipura problem?
This may not be the right place to ask, but I have an annoying issue with
a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The
system is remote to me, so I've only been able to observe this by dialling
into a VoIP phone on-site, then run commands on the box remotely!)
First of all it's all working fine connected to an Asterisk box and the
user can make/tak...
2005 May 23
3
ISPCON Mini-emergency: ATA186 Power Cube OK on SPA841?
Guess who's here to do an Asterisk demo this week without the power
supply for his SPA-841.
I have an ATA186 with me. Both phones use a 5v supply. Does anyone
know whether the supplies are interchangeable?
Thanks in advance; sorry for the noise.
B.
2004 Jun 29
1
Asterisk and Sipura SPA-1000 configs
Anyone had any experience here on how to config both ends, asterisk and
the sipura SPA1000
TIA
2004 Jun 14
2
inviting an spa-x000
...ead:
SIP/2.0 404 Not Found
To: <sip:42.7.11.194>
From: "asterisk" <sip:asterisk@128.9.0.39>;tag=as39d40d19
Call-ID: 37a3a2bc5811012d44645e861979d27e@128.9.0.39
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 128.9.0.39:5060;branch=z9hG4bK43efe1d7
Server: Sipura/SPA1000-2.0.8(GW)
Content-Length: 0
the sip.conf entry for the spa is
[spa0p]
callerid="SPA Phone" <102>
type=friend
host=dynamic
port=5060
username=spa0p
auth=md5
secret=seekret
qualify=1000
dtmfmode=rfc2833
canreinvite=no
context=in-99...
2005 May 19
1
Newbie X100P question
Hello,
I just bought a X100P from digitnetworks.
It is supposed to be a FXO card, but there are 2 rj-11 plug on the card.
One is labelled "phone" and the other "pstn". When i plug the "pstn" on
the wall and the "phone" on my analog phone, everything (incoming and
outgoing calls) works like before (without asterisk).
AFAIU, i should have an FXS card in my
2005 Mar 01
6
Broadvoice + Videosupport=yes - Fails!
Hi All
First time poster, long time reader.
I just ran into something really bizarre. I've enabled videosupport and
have been testing sip calls with Xten Eyebeam software, it works
(mostly)
However, when I have
Videosupport=yes
In the [general] section of my sip.conf, broadvoice calls fail w/ "We're
sorry your call cannot be completed at this time"
So... I've