asterisk users - Apr 2005 - using asterisk as a gateway for residential IP telephony clients

Hi

Do you think that asterisk could be use as a gateway for residential ip 
telephony clients ?

It is described as a PBX but could it be used to provide IP telephony to 
our cable modem users ?

Any experience in this field

Thanks

Patrick

Absolutely!   We're using it as such now.  If you'd like more
information, e-mail me off list and we could arrange a time to talk.

On Apr 5, 2005 12:44 PM, Video Dery / Internet du Royaume
<gestion@royaume.com> wrote:
> Hi
> 
> Do you think that asterisk could be use as a gateway for residential ip
> telephony clients ?
> 
> It is described as a PBX but could it be used to provide IP telephony to
> our cable modem users ?
> 
> Any experience in this field
> 
> Thanks
> 
> Patrick
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Yes, however other issues would affect its operation. One BitTorrent
and all of your users would be upset.

On Apr 5, 2005 11:44 AM, Video Dery / Internet du Royaume
<gestion@royaume.com> wrote:
> Hi
> 
> Do you think that asterisk could be use as a gateway for residential ip
> telephony clients ?
> 
> It is described as a PBX but could it be used to provide IP telephony to
> our cable modem users ?
> 
> Any experience in this field
> 
> Thanks
> 
> Patrick
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 

-- 
Andrew Latham

http://www.lathama.com
lathama@gmail.com
lathama@yahoo.com
lathama@lathama.com
If any of the above are not working,
we have bigger problems than my email.

Dear All,

We've got 1 set of Polycom V500 Video Conferencing Kit in my Office. I'm
trying to link it with Asterisk and is facing some issues. Would like to
seek your kind advise.

The Polycom V500 is unable to make the outgoing calls, and will always
report the "ENTER ERROR HERE".

"sip show peers" does not shows that the Polycom V500 being able to
register. The account is working alright as I've used the account on
Eyebeam and its working fine.

Here are the debug logs for the System

<-- SIP read from 192.168.100.146:5060:
INVITE sip:404@192.168.100.146 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655
Max-Forwards: 70
From: <sip:898@192.168.100.146>;epid=82042503E72EB0;tag=df8c4526
To: <sip:404@192.168.100.146>
Call-ID: e8c14000@192.168.100.146
CSeq: 1 INVITE
User-Agent: Polycom V500 Release 7.5 - 15Dec2004 10:12
Contact: <sip:192.168.100.146>
Content-Type: application/sdp
Content-Length: 899

v=0
o=Vigor11 1627471320 0 IN IP4 192.168.100.146
s=-
c=IN IP4 192.168.100.146
b=AS:384
t=0 0
m=audio 49178 RTP/AVP 99 98 97 102 101 103 9 15 0 8 18
a=rtpmap:99 SIREN14/16000
a=fmtp:99 bitrate=48000
a=rtpmap:98 SIREN14/16000
a=fmtp:98 bitrate=32000
a=rtpmap:97 SIREN14/16000
a=fmtp:97 bitrate=24000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:101 G7221/16000
a=fmtp:101 bitrate=24000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=16000
a=rtpmap:9 G722/8000
a=rtpmap:15 G728/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729A/8000
m=video 49180 RTP/AVP 109 34 96 31
b=TIAS:384000
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42800c max-mbps=10000
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2 F J T
a=rtpmap:31 H261/90000
a=fmtp:31 CIF=1 QCIF=1
m=data 49182 RTP/AVP 100
a=rtpmap:100 H224

--- (11 headers 35 lines)---
Using latest request as basis request
Sending to 192.168.100.146 : 5060 (non-NAT)
Apr 13 14:15:10 DEBUG[1496]: chan_sip.c:5947 check_user_full: Setting NAT
on RTP to 524288
Apr 13 14:15:10 DEBUG[1496]: chan_sip.c:5951 check_user_full: Setting NAT
on VRTP to 524288
Reliably Transmitting (NAT) to 192.168.100.146:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.100.146;branch=z9hG4bK1f784655;received=192.168.100.146;rport=5060
From: <sip:898@192.168.100.146>;epid=82042503E72EB0;tag=df8c4526
To: <sip:404@192.168.100.146>;tag=as36644353
Call-ID: e8c14000@192.168.100.146
CSeq: 1 INVITE
User-Agent: nVoice PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:404@61.14.78.47>
Proxy-Authenticate: Digest realm="nvoice", nonce="60b31ab3"
Content-Length: 0


---
Scheduling destruction of call 'e8c14000@192.168.100.146' in 15000 ms
Found user '898'
tannery*CLI>
<-- SIP read from 192.168.100.146:5060:
ACK sip:192.168.100.146 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.146;branch=z9hG4bK1f784655
Max-Forwards: 70
From: <sip:898@192.168.100.146>;epid=82042503E72EB0;tag=df8c4526
To: <sip:404@192.168.100.146>
Call-ID: e8c14000@192.168.100.146
CSeq: 1 ACK
Contact: <sip:192.168.100.146>
Content-Length: 0



Best Regards,

=============================David Choo
Systems Engineer
Business & Technology Division
"Engineered for Changing Businesses"
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel: 65-68487806
Fax : 65-6842 2724
E-mail :DavidChoo@Espore.com
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