thr3ads.net - asterisk users - [Asterisk-Users] Got SIP response 302 "Moved Temporarily" back.... [Apr 2005]

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etiennep@kingsley.co.za

2005-Apr-18 01:16 UTC

[Asterisk-Users] Got SIP response 302 "Moved Temporarily" back....

Hello everyone.
How was your weekend?

Anyway...
'Got SIP response 302 "Moved Temporarily" back from
192.168.10.24'

Lately I've been getting this error... well i am at a loss as to why I am
getting this when on Friday I was able to make a pass-through call with no
problems.

+----------+       +-----------------+     +-----------+     +---------+
|Net2Phone |======>|sip.Net2Phone.com|====>|Asterisk(*)|====>|SIP
Phone|
|MAX IP10  |       +-----------------+     +-----------+     |GS BT-100|
+----------+                                 (GateWay)       +---------+
[ip 196.x.x.x]      [ip 66.33.157.12]      [ip 165.x.x.x]    [ip 192.168.10.24]

Asterisk Server(GateWay) has two eth cards - one with the external ip of
165.x.x.x
via ppp0 and the other and internal ip of 192.x.x.x

Now on Friday this setup worked 100% for a pass through - but now, I keep on
getting this "302" error and I can't see how SIP is ending up in a
HAIRPIN
senario.

DialPlan is simple:
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Dial(SIP/Receprion|20|tr)

Asterisk(*) Output:
    -- Executing Answer("SIP/3828106029-29bb", "") in new
stack
    -- Executing Wait("SIP/3828106029-29bb", "1") in new
stack
    -- Executing Dial("SIP/3828106029-29bb",
"SIP/Reception|20|tr") in new stack
    -- Called Reception
Apr 18 09:46:22 NOTICE[1841]: channel.c:1812 ast_set_write_format: Unable to
find a path from slin to g723
Apr 18 09:46:22 WARNING[1841]: indications.c:78 playtones_alloc: Unable to set
'SIP/3828106029-29bb' to signed linear format (write)
    -- Got SIP response 302 "Moved Temporarily" back from
192.168.10.204
    -- SIP/Reception-e6bf is busy
  == Everyone is busy/congested at this time (1:1/0/0)

Any help on this issue will be apreciated. Thank you.

Kindly,
Etienne Pretorius

etiennep@kingsley.co.za

2005-Apr-18 02:16 UTC

head link

[Asterisk-Users] Got SIP response 302 "Moved Temporarily" back....

Got some debug info... please see attachement.

Quoting etiennep@kingsley.co.za:
> Hello everyone.
> How was your weekend?
>
> Anyway...
> 'Got SIP response 302 "Moved Temporarily" back from
192.168.10.24'
>
> Lately I've been getting this error... well i am at a loss as to why I
am
> getting this when on Friday I was able to make a pass-through call with no
> problems.
>
> +----------+       +-----------------+     +-----------+     +---------+
> |Net2Phone |======>|sip.Net2Phone.com|====>|Asterisk(*)|====>|SIP
Phone|
> |MAX IP10  |       +-----------------+     +-----------+     |GS BT-100|
> +----------+                                 (GateWay)       +---------+
> [ip 196.x.x.x]      [ip 66.33.157.12]      [ip 165.x.x.x]    [ip
> 192.168.10.24]
>
> Asterisk Server(GateWay) has two eth cards - one with the external ip of
> 165.x.x.x
> via ppp0 and the other and internal ip of 192.x.x.x
>
> Now on Friday this setup worked 100% for a pass through - but now, I keep
on
> getting this "302" error and I can't see how SIP is ending up
in a HAIRPIN
> senario.
>
> DialPlan is simple:
> exten => s,1,Answer
> exten => s,2,Wait(1)
> exten => s,3,Dial(SIP/Receprion|20|tr)
>
> Asterisk(*) Output:
>     -- Executing Answer("SIP/3828106029-29bb", "") in
new stack
>     -- Executing Wait("SIP/3828106029-29bb", "1") in
new stack
>     -- Executing Dial("SIP/3828106029-29bb",
"SIP/Reception|20|tr") in new
> stack
>     -- Called Reception
> Apr 18 09:46:22 NOTICE[1841]: channel.c:1812 ast_set_write_format: Unable
to
> find a path from slin to g723
> Apr 18 09:46:22 WARNING[1841]: indications.c:78 playtones_alloc: Unable to
> set
> 'SIP/3828106029-29bb' to signed linear format (write)
>     -- Got SIP response 302 "Moved Temporarily" back from
192.168.10.204
>     -- SIP/Reception-e6bf is busy
>   == Everyone is busy/congested at this time (1:1/0/0)
>
> Any help on this issue will be apreciated. Thank you.
>
> Kindly,
> Etienne Pretorius
>
>
>
> _______________________________________________
> 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
>



-------------- next part --------------
SIP Debugging Enabled for IP: 192.168.10.24:5060
    -- Executing Answer("SIP/3828106029-8e32", "") in new
stack
    -- Executing Wait("SIP/3828106029-8e32", "1") in new
stack
    -- Executing Dial("SIP/3828106029-8e32",
"SIP/Reception|20|tr") in new stack
We're at 192.168.10.1 port 14468
Answering/Requesting with root capability 0x1 (g723)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.10.24:5060:
INVITE sip:Reception@192.168.10.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0bf1b64b
From: "asterisk" <sip:Reception@192.168.10.1>;tag=as4a953271
To: <sip:Reception@192.168.10.24>
Contact: <sip:Reception@192.168.10.1>
Call-ID: 6cd8dba94dacf4fd01c065e0620fb84d@192.168.10.1
CSeq: 102 INVITE
User-Agent: X-Lite release 1103m
Date: Mon, 18 Apr 2005 09:11:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 2433 2433 IN IP4 192.168.10.1
s=session
c=IN IP4 192.168.10.1
t=0 0
m=audio 14468 RTP/AVP 4 101
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called Reception
Apr 18 11:11:22 NOTICE[2433]: channel.c:1812 ast_set_write_format: Unable to
find a path from slin to g723
Apr 18 11:11:22 WARNING[2433]: indications.c:78 playtones_alloc: Unable to set
'SIP/3828106029-8e32' to signed linear format (write)
adsl-test*CLI>
<-- SIP read from 192.168.10.24:5060:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0bf1b64b
From: "asterisk" <sip:Reception@192.168.10.1>;tag=as4a953271
To: <sip:Reception@192.168.10.24>;tag=6fe736daf4223205
Call-ID: 6cd8dba94dacf4fd01c065e0620fb84d@192.168.10.1
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.5.18
Contact: sip:@192.168.10.1
Diversion: <sip:Reception@192.168.10.24>;reason=unconditional
Content-Length: 0


--- (10 headers 0 lines)---
    -- Got SIP response 302 "Moved Temporarily" back from
192.168.10.24
Transmitting (no NAT) to 192.168.10.24:5060:
ACK sip:Reception@192.168.10.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0bf1b64b
From: "asterisk" <sip:Reception@192.168.10.1>;tag=as4a953271
To: <sip:Reception@192.168.10.24>;tag=6fe736daf4223205
Contact: <sip:Reception@192.168.10.1>
Call-ID: 6cd8dba94dacf4fd01c065e0620fb84d@192.168.10.1
CSeq: 102 ACK
User-Agent: X-Lite release 1103m
Content-Length: 0


---
    -- SIP/Reception-fe13 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
Destroying call '6cd8dba94dacf4fd01c065e0620fb84d@192.168.10.1'

asterisk users - Apr 2005 - Got SIP response 302 "Moved Temporarily" back....

[Asterisk-Users] Got SIP response 302 "Moved Temporarily" back....

[Asterisk-Users] Got SIP response 302 "Moved Temporarily" back....