[Spce-user] Can not Install sipwise on debian 8.6
Cedric Le
cedric at hoiio.com
Fri Jan 20 05:27:30 EST 2017
Hi Team,
Still not working when i change sip_ext to eth1. It alway use eth0 but
peering on subnet 172.16.19.0 with int eth1.
INVITE sip:01677140640 at 172.16.19.10:5060;transport=udp SIP/2.0'
Via: SIP/2.0/UDP 172.16.16.111:5080;branch=z9hG4bKFIsFaadd;rport'
*From: <sip:84901774554 at 172.16.16.111 <sip%3A84901774554 at 172.16.16.111>>*
;tag=28D6F513-5881DBF3000EBCB1-DAEEE700'
To: <sip:01677140640 at 172.16.19.10>'
CSeq: 10 INVITE'
Call-ID: Y2UzMGE2ZGYzZDI0ZTczOWVjNDQzYTBiYTA4ZTljNWI._b2b-1'
Route: <sip:172.16.16.111:5060;received='sip:172.16.19.10:
5060;lr;transport=udp';lr>'
Max-Forwards: 70'
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE'
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
P-Asserted-Identity: <sip:84901774554 at 172.16.16.111>'
P-D-Uri: sip:172.16.16.111:5060;received='sip:172.16.19.10:
5060;lr;transport=udp''
Content-Type: application/sdp'
Contact: <sip:172.16.16.111:5080;transport=udp>'
Content-Length: 287'
'
v=0'
o=Z 0 0 IN IP4 192.168.1.52'
s=Z'
c=IN IP4 203.205.28.49'
t=0 0'
m=audio 8000 RTP/AVP 3 110 8 0 98 101'
a=rtpmap:110 speex/8000'
a=rtpmap:98 iLBC/8000'
a=fmtp:98 mode=20'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=sendrecv'
a=oldmediaip:192.168.1.52'
a=direction:active'
May i know how to fix this issue?
On Fri, Jan 20, 2017 at 5:49 PM, Cedric Le <cedric at hoiio.com> wrote:
> Hi Walter,
>
> Still not working when i change sip_ext to eth1. It alway use eth0 but
> peering on subnet 172.16.19.0 with int eth1.
>
> INVITE sip:01677140640 at 172.16.19.10:5060;transport=udp SIP/2.0'
> Via: SIP/2.0/UDP 172.16.16.111:5080;branch=z9hG4bKFIsFaadd;rport'
> *From: <sip:84901774554 at 172.16.16.111 <sip%3A84901774554 at 172.16.16.111>>*
> ;tag=28D6F513-5881DBF3000EBCB1-DAEEE700'
> To: <sip:01677140640 at 172.16.19.10>'
> CSeq: 10 INVITE'
> Call-ID: Y2UzMGE2ZGYzZDI0ZTczOWVjNDQzYTBiYTA4ZTljNWI._b2b-1'
> Route: <sip:172.16.16.111:5060;received='sip:172.16.19.10:
> 5060;lr;transport=udp';lr>'
> Max-Forwards: 70'
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE'
> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
> P-Asserted-Identity: <sip:84901774554 at 172.16.16.111>'
> P-D-Uri: sip:172.16.16.111:5060;received='sip:172.16.19.10:
> 5060;lr;transport=udp''
> Content-Type: application/sdp'
> Contact: <sip:172.16.16.111:5080;transport=udp>'
> Content-Length: 287'
> '
> v=0'
> o=Z 0 0 IN IP4 192.168.1.52'
> s=Z'
> c=IN IP4 203.205.28.49'
> t=0 0'
> m=audio 8000 RTP/AVP 3 110 8 0 98 101'
> a=rtpmap:110 speex/8000'
> a=rtpmap:98 iLBC/8000'
> a=fmtp:98 mode=20'
> a=rtpmap:101 telephone-event/8000'
> a=fmtp:101 0-15'
> a=sendrecv'
> a=oldmediaip:192.168.1.52'
> a=direction:active'
>
>
> May i know how to fix this issue?
>
> Thanks
>
> On Fri, Jan 20, 2017 at 5:28 PM, Walter Klomp <walter at myrepublic.net>
> wrote:
>
>> Because your sip_ext is on eth0 … it will always use that interface.
>>
>>
>>
>>
>> On 20 Jan 2017, at 5:22 PM, Cedric Le <cedric at hoiio.com> wrote:
>>
>> Hi Team,
>>
>> May i know how to create route and change the route. We are facing with
>> issue can not make out going call. We are peering with ip on 172.16.19.10
>> and it connect to eth1 but It alway use the route on eth0, did not use
>> eth1.
>>
>> eth0:
>> ip: 172.16.16.111
>> netmask: 255.255.255.0
>> type:
>> - web_ext
>> - ssh_ext
>> - sip_ext
>> - rtp_ext
>> - sip_int
>> - rtp_int
>> - web_int
>> eth1:
>> ip: 172.16.19.57
>> netmask: 255.255.255.0
>> type:
>> - web_ext
>> - web_int
>> - mon_ext
>>
>>
>> Trunk to peer with ip 172.16.19.10
>>
>>
>> INVITE sip:01677140640 at 172.16.19.10:5060;transport=udp SIP/2.0'
>> Via: SIP/2.0/UDP 172.16.16.111:5080;branch=z9hG4bKSj2LNadd;rport'
>> From: <sip:84901774554 at 172.16.16.111>;tag=00A97DAD-5881D5E30005651
>> A-50A10700'
>> To: <sip:01677140640 at 172.16.19.10>'
>> CSeq: 10 INVITE'
>> Call-ID: NjhmYTBhZDEwYThjODY4MDdkMmI0YTZkZDMyMjMwNGI._b2b-1'
>> Route: <sip:172.16.16.111:5060;received='sip:172.16.19.10:5060;lr;
>> transport=udp';lr>'
>> Max-Forwards: 70'
>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE'
>> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
>> P-Asserted-Identity: <sip:84901774554 at 172.16.16.111>'
>> P-D-Uri: sip:172.16.16.111:5060;received='sip:172.16.19.10:5060;lr;
>> transport=udp''
>> Content-Type: application/sdp'
>> Contact: <sip:172.16.16.111:5080;transport=udp>'
>> Content-Length: 287'
>> '
>> v=0'
>> o=Z 0 0 IN IP4 192.168.1.52'
>> s=Z'
>> c=IN IP4 203.205.28.49'
>> t=0 0'
>> m=audio 8000 RTP/AVP 3 110 8 0 98 101'
>> a=rtpmap:110 speex/8000'
>> a=rtpmap:98 iLBC/8000'
>> a=fmtp:98 mode=20'
>> a=rtpmap:101 telephone-event/8000'
>> a=fmtp:101 0-15'
>> a=sendrecv'
>> a=oldmediaip:192.168.1.52'
>> a=direction:active'
>>
>>
>> On Thu, Jan 19, 2017 at 9:34 PM, Alex Lutay <alutay at sipwise.com> wrote:
>>
>>> Hi,
>>>
>>> I do not see anything which can cause such error.
>>> Is it still topical? Do you use proxy?
>>>
>>> P.S. Can you please send logs in plain-text next time,
>>> it simplifies quoting. Thank you!
>>>
>>> On 01/10/2017 03:50 AM, Cedric Le wrote:
>>> > We are facing with another problem. Please help to advise
>>>
>>> --
>>> Alex Lutay
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> https://lists.sipwise.com/listinfo/spce-user
>>>
>>
>>
>>
>> --
>> Thanks and Regard,
>> Cedric Le
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> https://lists.sipwise.com/listinfo/spce-user
>>
>>
>>
>
>
> --
> Thanks and Regard,
> Cedric Le
>
--
Thanks and Regard,
Cedric Le
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