[Asterisk-Dev] IWATSU IX-6 SIP Based IP phone
Masakazu Nakano
n-mack at md.neweb.ne.jp
Mon Apr 26 22:15:52 MST 2004
Hi bkw.
and all thanks mark.
I checked and I think that problem is maybe fixed.
Please close that ticket :-)
IWATSU IX6 still has another problem following that.
-- Got SIP response 481 "Call Leg/Transaction Dose Not Exist" back from
192.168.XX.XX
Sip read:
SIP/2.0 481 Call Leg/Transaction Dose Not Exist
Via: SIP/2.0/UDP 192.168.XX.X:5060;branch=z9hG4bK394b4fea
Call-ID: 0ce9e7803771c3ac08aaf01349bebc05 at 192.168.XX.X
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk at 192.168.XX.X>;tag=as1c141207
To: <sip:20000700 at 192.168.XX.XX>;tag=00505101178f-1064967542-iu-V1.52.07.VX-00001-r
Server: IWATSU_SIP/V1.52.07.VX
Content-Length: 0
uhm...typo found in this firmware...
BTW
Is latest cvs works well with sipphone.com ?
Apr 27 13:48:07 NOTICE[9226]: chan_sip.c:3268 sip_reg_timeout: Registration for '17473861230 at 198.65.166.131' timed out, trying again
Apr 27 13:48:07 NOTICE[9226]: chan_sip.c:3268 sip_reg_timeout: Registration for '17473861053 at 198.65.166.131' timed out, trying again
Apr 27 13:48:13 WARNING[9226]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 542289ec6de91b1838437fdb7644a45c at 210.194.200.158 for seqno 132 (
Critical Request)
Apr 27 13:48:13 WARNING[9226]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 79a1deaa75c6c33a12e685fb70c6a529 at 210.194.200.158 for seqno 132 (
Critical Request)
Currently,iptel and inphonex are okay...uhm.... strange :-?
they are work with SER.
Regards.
mack_jpn
On Mon, 26 Apr 2004 22:30:52 -0600
"brian k. west" <brian at bkw.org> wrote:
> http://bugs.digium.com/bug_view_page.php?bug_id=0001333
>
> Please confirm if it works or not so we can close it out.
>
> Thanks,
> Brian
> ----- Original Message -----
> From: "Masakazu Nakano" <n-mack at md.neweb.ne.jp>
> To: <asterisk-dev at lists.digium.com>
> Sent: Saturday, April 24, 2004 8:57 PM
> Subject: [Asterisk-Dev] IWATSU IX-6 SIP Based IP phone
>
>
> >
> > Hi All.
> >
> > now I'm try to regist that phone.
> > but it had funny REGISTER header like that.because it cannot to regist
> > to asterisk.( multi-lined Proxy-Authorization: )
> >
> > Sip read:
> > REGISTER sip:192.168.10.1 SIP/2.0
> > Call-ID: 96C3C59796C3C597.10 at 192.168.10.32
> > CSeq: 2 REGISTER
> > From: "2000XXXX" <sip:2000XXXX at 192.168.10.1>;tag=00505101178f-108286159
> > 4-iu-V1.52.07.VX-00070-o
> > To: "2000XXXX" <sip:2000XXXX at 192.168.10.1>
> > Via: SIP/2.0/UDP
> 192.168.10.32:5060;branch=z9hG4bK-00505101178f-1082861594-iu-V1.52.07.VX-000
> 70-o-2
> > Max-Forwards: 70
> > User-Agent: IWATSU_SIP/V1.52.07.VX
> > Expires: 300
> > Contact: sip:2000XXXX at 192.168.10.32;expires=300
> > Date: Sun, 25 Apr 2004 02:53:14 GMT
> > Proxy-Authorization: Digest realm="asterisk",
> > nonce="3f1fcbf3",
> > opaque="",
> > cnonce="",
> > nc=00000001,
> > uri="sip:192.168.10.1",
> > username="2000XXXX",
> > response="a000630f3e2b2f94308250f76a1dee73"
> > Content-Length: 0
> >
> > Does anyone already fixed chan_sip.c?
> >
> > mack_jpn
> >
> > _______________________________________________
> > Asterisk-Dev mailing list
> > Asterisk-Dev at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
>
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
---
代理店どっとこむ 代表 中野
ポータル&自動音声応答装置、IP電話、IPPBX複合システム開発、
UMS・業務支援・応対窓口・在宅コールセンター開発請負
http://www.dairiten.com/
More information about the asterisk-dev
mailing list