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Chris

2003-May-02 09:41 UTC

[Asterisk-Users] SIP Peers unreachable

Hi Everyone,

I'm new to * and I'm trying to setup a small configuration of SIP
clients.
Eventually when I get this working I plan on expanding with a Digium
developers kit to add analog phones and PSTN access.

My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both
peers seem to register with * but I cannot call to one another. When I dial
the associated extension, the call goes to the programmed voicemail
extension (busy) yet if I create an extension to call out through the proxy
(IX66), I can still reach my destination. It's just calling within * there
is a problem. I suspect it's because the status is unreachable but I'm
not
sure how to fix it.

Here is the sip show peers output.

Name/username    Host                 Mask             Port     Status
sipset/sipset    192.200.14.31    (D)  255.255.255.255  5060     UNREACHABLE
sippc/sippc      192.200.14.33    (D)  255.255.255.255  5060     UNREACHABLE

Here is the sip.conf settings:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
register => 9055551212@somewhere.homeip.net

[sippc]
type=friend
username=sippc
secret=blah
host=dynamic
qualify=3000

[sipset]
type=friend
username=sipset
secret=blah
host=dynamic
qualify=3000

Here is the extensions.conf settings:
exten => 421,1,Dial(SIP/sipset)  	  ; Mitel 5055 SIP Phone
exten => 421,2,Voicemail(u421)
exten => 421,102,Voicemail(b421)
exten => 422,1,Dial(SIP/sippc)  	  ; Xten client
exten => 422,2,Voicemail(u422)
exten => 422,102,Voicemail(b422)

exten => 444,1,Dial(SIP/tony@somewhere.homeip.net)  ; friends MSN (4.6)
account registered to IX66


These are the console messages when I dial 421 from 422

    -- Executing Dial("SIP/sippc-b5f6", "SIP/sipset") in new
stack
  == Everyone is busy at this time
    -- Executing VoiceMail("SIP/sippc-b5f6", "b421") in new
stack
  == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
'/etc/asterisk/voicemail.conf': Found
    -- Playing 'vm-theperson'
    -- Playing 'digits/4'
    -- Playing 'digits/2'
    -- Playing 'digits/1'
    -- Playing 'vm-isonphone'
    -- Playing 'vm-intro'
  == Spawn extension (default, 421, 102) exited non-zero on
'SIP/sippc-b5f6'


Any help is appreciated.

Thanks.

Chris

John Todd

2003-May-02 10:33 UTC

head link

[Asterisk-Users] SIP Peers unreachable

I would suggest firing up "tethereal port 5060" on the command line 
in a different window and see if your SIP requests are making it to 
the destination device when you create a call.  If you don't have 
tethereal, get the "ethereal" packages and install them.

Remove the "qualify" lines until you get things working.

Launch asterisk with "asterisk -vvvgc" to look at some additional 
debugging information when attempting to dial.

What happens when you dial 444 from one of your phones?

JT



>Hi Everyone,
>
>I'm new to * and I'm trying to setup a small configuration of SIP
clients.
>Eventually when I get this working I plan on expanding with a Digium
>developers kit to add analog phones and PSTN access.
>
>My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both
>peers seem to register with * but I cannot call to one another. When I dial
>the associated extension, the call goes to the programmed voicemail
>extension (busy) yet if I create an extension to call out through the proxy
>(IX66), I can still reach my destination. It's just calling within *
there
>is a problem. I suspect it's because the status is unreachable but
I'm not
>sure how to fix it.
>
>Here is the sip show peers output.
>
>Name/username    Host                 Mask             Port     Status
>sipset/sipset    192.200.14.31    (D)  255.255.255.255  5060     UNREACHABLE
>sippc/sippc      192.200.14.33    (D)  255.255.255.255  5060     UNREACHABLE
>
>Here is the sip.conf settings:
>[general]
>port = 5060
>bindaddr = 0.0.0.0
>context = default
>register => 9055551212@somewhere.homeip.net
>
>[sippc]
>type=friend
>username=sippc
>secret=blah
>host=dynamic
>qualify=3000
>
>[sipset]
>type=friend
>username=sipset
>secret=blah
>host=dynamic
>qualify=3000
>
>Here is the extensions.conf settings:
>exten => 421,1,Dial(SIP/sipset) 	  ; Mitel 5055 SIP Phone
>exten => 421,2,Voicemail(u421)
>exten => 421,102,Voicemail(b421)
>exten => 422,1,Dial(SIP/sippc) 	  ; Xten client
>exten => 422,2,Voicemail(u422)
>exten => 422,102,Voicemail(b422)
>
>exten => 444,1,Dial(SIP/tony@somewhere.homeip.net)  ; friends MSN (4.6)
>account registered to IX66
>
>
>These are the console messages when I dial 421 from 422
>
>     -- Executing Dial("SIP/sippc-b5f6", "SIP/sipset")
in new stack
>   == Everyone is busy at this time
>     -- Executing VoiceMail("SIP/sippc-b5f6", "b421") in
new stack
>   == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
>'/etc/asterisk/voicemail.conf': Found
>     -- Playing 'vm-theperson'
>     -- Playing 'digits/4'
>     -- Playing 'digits/2'
>     -- Playing 'digits/1'
>     -- Playing 'vm-isonphone'
>     -- Playing 'vm-intro'
>   == Spawn extension (default, 421, 102) exited non-zero on
'SIP/sippc-b5f6'
>
>
>Any help is appreciated.
>
>Thanks.
>
>Chris
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users

Mark Spencer

2003-May-02 12:42 UTC

head link

[Asterisk-Users] SIP Peers unreachable

Try turning off the "qualify". It's possible they don't
support "OPTIONS"
or something.

Mark

On Fri, 2 May 2003, Chris wrote:
> Hi Everyone,
>
> I'm new to * and I'm trying to setup a small configuration of SIP
clients.
> Eventually when I get this working I plan on expanding with a Digium
> developers kit to add analog phones and PSTN access.
>
> My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both
> peers seem to register with * but I cannot call to one another. When I dial
> the associated extension, the call goes to the programmed voicemail
> extension (busy) yet if I create an extension to call out through the proxy
> (IX66), I can still reach my destination. It's just calling within *
there
> is a problem. I suspect it's because the status is unreachable but
I'm not
> sure how to fix it.
>
> Here is the sip show peers output.
>
> Name/username    Host                 Mask             Port     Status
> sipset/sipset    192.200.14.31    (D)  255.255.255.255  5060    
UNREACHABLE
> sippc/sippc      192.200.14.33    (D)  255.255.255.255  5060    
UNREACHABLE
>
> Here is the sip.conf settings:
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = default
> register => 9055551212@somewhere.homeip.net
>
> [sippc]
> type=friend
> username=sippc
> secret=blah
> host=dynamic
> qualify=3000
>
> [sipset]
> type=friend
> username=sipset
> secret=blah
> host=dynamic
> qualify=3000
>
> Here is the extensions.conf settings:
> exten => 421,1,Dial(SIP/sipset)  	  ; Mitel 5055 SIP Phone
> exten => 421,2,Voicemail(u421)
> exten => 421,102,Voicemail(b421)
> exten => 422,1,Dial(SIP/sippc)  	  ; Xten client
> exten => 422,2,Voicemail(u422)
> exten => 422,102,Voicemail(b422)
>
> exten => 444,1,Dial(SIP/tony@somewhere.homeip.net)  ; friends MSN (4.6)
> account registered to IX66
>
>
> These are the console messages when I dial 421 from 422
>
>     -- Executing Dial("SIP/sippc-b5f6", "SIP/sipset")
in new stack
>   == Everyone is busy at this time
>     -- Executing VoiceMail("SIP/sippc-b5f6", "b421") in
new stack
>   == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
> '/etc/asterisk/voicemail.conf': Found
>     -- Playing 'vm-theperson'
>     -- Playing 'digits/4'
>     -- Playing 'digits/2'
>     -- Playing 'digits/1'
>     -- Playing 'vm-isonphone'
>     -- Playing 'vm-intro'
>   == Spawn extension (default, 421, 102) exited non-zero on
'SIP/sippc-b5f6'
>
>
> Any help is appreciated.
>
> Thanks.
>
> Chris
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Chris

2003-May-02 14:46 UTC

head link

[Asterisk-Users] RE: SIP Peers unreachable

(Sorry if this breaks the thread. I receive the list mail by the daily
digest.)

I removed the qualifier from all my SIP extensions in the sip.conf as
suggested by both John and Mark. Still the same issue.

It's interesting, Mark, that you mentioned that the clients may not support
OPTIONS. When I turn on SIP debug, I get 3 retransmitions of CSeq: 102
OPTIONS in the console for each extension. Not sure what this means.

Here is the dump from tethereal while calling 421 from 422. It does look
like the SIP request are making it to the destination.

[root@homegw asterisk requirements]# tethereal port 5060
Capturing on eth0
  0.000000     pc-00033 -> homegw.defrag.homeip.net SIP/SDP Request: INVITE
sip:
421@192.200.14.251, with session description
  0.036408 homegw.defrag.homeip.net -> pc-00033     SIP Status: 407 Proxy
Authen
tication Required
  0.042191     pc-00033 -> homegw.defrag.homeip.net SIP Request: ACK
sip:421@172
.20.14.251
  0.045378 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
  0.049398 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
  0.252140     pc-00033 -> homegw.defrag.homeip.net SIP/SDP Request: INVITE
sip:
421@192.200.14.251, with session description
  0.363390 homegw.defrag.homeip.net -> pc-00033     SIP Status: 100 Trying
  0.710967 homegw.defrag.homeip.net -> pc-00033     SIP/SDP Status: 200 OK,
with
 session description
  0.772759     pc-00033 -> homegw.defrag.homeip.net SIP Request: ACK
sip:421@172
.20.14.251
  1.050840 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
  1.051029 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
  2.060714 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
  2.060900 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
  3.070620 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
  3.070807 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
  4.734937     pc-00033 -> homegw.defrag.homeip.net SIP Request: BYE
sip:421@172
.20.14.251
  4.737003 homegw.defrag.homeip.net -> pc-00033     SIP Status: 200 OK
  4.759445     pc-00033 -> homegw.defrag.homeip.net SIP Request: ACK
sip:421@172
.20.14.251
 14.081490 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
 14.085292 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
 15.089388 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
 15.089571 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
 16.099288 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
 16.099471 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
 17.109195 homegw.defrag.homeip.net -> pc-00031     SIP Request: OPTIONS
sip:
 17.109388 homegw.defrag.homeip.net -> pc-00033     SIP Request: OPTIONS
sip:
[root@homegw asterisk requirements]#

Here is a capture from the console with SIP Debug turned on.

 to 192.200.14.33:5060
Sip read: >
ACK sip:421@192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc@192.200.14.251>;tag=939212736
To: <sip:421@192.200.14.251>;tag=as23f24230
Contact: <sip:sippc@192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C@192.200.14.33
CSeq: 21726 ACK
Max-Forwards: 70
Content-Length: 0


9 headers, 0 lines
Sip read: >
INVITE sip:421@192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc@192.200.14.251>;tag=939212736
To: <sip:421@192.200.14.251>
Contact: <sip:sippc@192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C@192.200.14.33
CSeq: 21727 INVITE
Proxy-Authorization: Digest
username="sippc",realm="asterisk",nonce="788a7172",response="1b6ff26ec4f095b
723d429306edca5d9",uri="sip:421@192.200.14.251"
Content-Type: application/sdp
Content-Length: 229

v=0
o=sippc 39160289 39160289 IN IP4 192.200.14.33
s=X-Lite
c=IN IP4 192.200.14.33
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:126 x-pro-encrypted/8000

10 headers, 10 lines
Using latest request as basis request
Sending to 192.200.14.33 : 5060 (non-NAT)
Capabilities: us - 14, them - 4, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 421 in default
list_route: hop: <sip:sippc@192.200.14.33:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc@192.200.14.251>;tag=939212736
To: <sip:421@192.200.14.251>;tag=as011374f6
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C@192.200.14.33
CSeq: 21727 INVITE
User-Agent: Asterisk PBX
Contact: <sip:421@192.200.14.251>
Content-Length: 0


 to 192.200.14.33:5060
    -- Executing Dial("SIP/sippc-040b", "SIP/sipset") in new
stack
  == Everyone is busy at this time
    -- Executing VoiceMail("SIP/sippc-040b", "b421") in new
stack
We're at 192.200.14.251 port 11980
Answering with capability 4
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc@192.200.14.251>;tag=939212736
To: <sip:421@192.200.14.251>;tag=as011374f6
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C@192.200.14.33
CSeq: 21727 INVITE
User-Agent: Asterisk PBX
Contact: <sip:421@192.200.14.251>
Content-Type: application/sdp
Content-Length: 191

v=0
o=root 24171 24171 IN IP4 192.200.14.251
s=session
c=IN IP4 192.200.14.251
t=0 0
m=audio 11980 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 192.200.14.33:5060
  == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
'/etc/asterisk/voicemail.conf': Found
Sip read: >
ACK sip:421@192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc@192.200.14.251>;tag=939212736
To: <sip:421@192.200.14.251>;tag=as011374f6
Contact: <sip:sippc@192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C@192.200.14.33
CSeq: 21727 ACK
Max-Forwards: 70
Content-Length: 0


9 headers, 0 lines
    -- Playing 'vm-theperson'
Retransmitting #1 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK18327382
From: "asterisk" <sip:asterisk@192.200.14.251>;tag=as4dbb6c0a
To: <sip:>
Contact: <sip:asterisk@192.200.14.251>
Call-ID: 4683962a3c7615a76bf2643b0823b915@192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.31:5060
Retransmitting #1 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK0ab60e7d
From: "asterisk" <sip:asterisk@192.200.14.251>;tag=as26d1d313
To: <sip:>
Contact: <sip:asterisk@192.200.14.251>
Call-ID: 62829f7e5be8b4292b9760a477045992@192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.33:5060
Retransmitting #2 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK18327382
From: "asterisk" <sip:asterisk@192.200.14.251>;tag=as4dbb6c0a
To: <sip:>
Contact: <sip:asterisk@192.200.14.251>
Call-ID: 4683962a3c7615a76bf2643b0823b915@192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.31:5060
Retransmitting #2 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK0ab60e7d
From: "asterisk" <sip:asterisk@192.200.14.251>;tag=as26d1d313
To: <sip:>
Contact: <sip:asterisk@192.200.14.251>
Call-ID: 62829f7e5be8b4292b9760a477045992@192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.33:5060
    -- Playing 'digits/4'
Sip read: >
BYE sip:421@192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc@192.200.14.251>;tag=939212736
To: <sip:421@192.200.14.251>;tag=as011374f6
Contact: <sip:sippc@192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C@192.200.14.33
CSeq: 21728 BYE
Content-Length: 0


8 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc@192.200.14.251>;tag=939212736
To: <sip:421@192.200.14.251>;tag=as011374f6
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C@192.200.14.33
CSeq: 21728 BYE
User-Agent: Asterisk PBX
Contact: <sip:421@192.200.14.251>
Content-Length: 0


 to 192.200.14.33:5060
  == Spawn extension (default, 421, 102) exited non-zero on
'SIP/sippc-040b'
Sip read:
ACK sip:421@192.200.14.251 SIP/2.0
Via: SIP/2.0/UDP 192.200.14.33:5060
From: sippc <sip:sippc@192.200.14.251>;tag=939212736
To: <sip:421@192.200.14.251>;tag=as011374f6
Contact: <sip:sippc@192.200.14.33:5060>
Call-ID: 26A99B50-F10D-4D38-98F2-A824FA14B84C@192.200.14.33
CSeq: 21728 ACK
Max-Forwards: 70
Content-Length: 0


9 headers, 0 lines
Retransmitting #3 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK18327382
From: "asterisk" <sip:asterisk@192.200.14.251>;tag=as4dbb6c0a
To: <sip:>
Contact: <sip:asterisk@192.200.14.251>
Call-ID: 4683962a3c7615a76bf2643b0823b915@192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.31:5060
Retransmitting #3 (no NAT):
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP 192.200.14.251:5060;branch=z9hG4bK0ab60e7d
From: "asterisk" <sip:asterisk@192.200.14.251>;tag=as26d1d313
To: <sip:>
Contact: <sip:asterisk@192.200.14.251>
Call-ID: 62829f7e5be8b4292b9760a477045992@192.200.14.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.200.14.33:5060
homegw*CLI>

Thanks for your assitance.

Cheers,

Chris

On Fri, 2 May 2003, Chris wrote:
> Hi Everyone,
>
> I'm new to * and I'm trying to setup a small configuration of SIP
clients.
> Eventually when I get this working I plan on expanding with a Digium
> developers kit to add analog phones and PSTN access.
>
> My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both
> peers seem to register with * but I cannot call to one another. When I
dial> the associated extension, the call goes to the programmed voicemail
> extension (busy) yet if I create an extension to call out through the
proxy> (IX66), I can still reach my destination. It's just calling within *
there
> is a problem. I suspect it's because the status is unreachable but
I'm not
> sure how to fix it.
>
> Here is the sip show peers output.
>
> Name/username    Host                 Mask             Port     Status
> sipset/sipset    192.200.14.31    (D)  255.255.255.255  5060
UNREACHABLE> sippc/sippc      192.200.14.33    (D)  255.255.255.255  5060
UNREACHABLE>
> Here is the sip.conf settings:
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = default
> register => 9055551212@somewhere.homeip.net
>
> [sippc]
> type=friend
> username=sippc
> secret=blah
> host=dynamic
> qualify=3000
>
> [sipset]
> type=friend
> username=sipset
> secret=blah
> host=dynamic
> qualify=3000
>
> Here is the extensions.conf settings:
> exten => 421,1,Dial(SIP/sipset)  	  ; Mitel 5055 SIP Phone
> exten => 421,2,Voicemail(u421)
> exten => 421,102,Voicemail(b421)
> exten => 422,1,Dial(SIP/sippc)  	  ; Xten client
> exten => 422,2,Voicemail(u422)
> exten => 422,102,Voicemail(b422)
>
> exten => 444,1,Dial(SIP/tony@somewhere.homeip.net)  ; friends MSN (4.6)
> account registered to IX66
>
>
> These are the console messages when I dial 421 from 422
>
>     -- Executing Dial("SIP/sippc-b5f6", "SIP/sipset")
in new stack
>   == Everyone is busy at this time
>     -- Executing VoiceMail("SIP/sippc-b5f6", "b421") in
new stack
>   == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
> '/etc/asterisk/voicemail.conf': Found
>     -- Playing 'vm-theperson'
>     -- Playing 'digits/4'
>     -- Playing 'digits/2'
>     -- Playing 'digits/1'
>     -- Playing 'vm-isonphone'
>     -- Playing 'vm-intro'
>   == Spawn extension (default, 421, 102) exited non-zero on
'SIP/sippc-b5f6'>
>
> Any help is appreciated.
>
> Thanks.
>
> Chris
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>


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Chris

2003-May-02 19:56 UTC

head link

[Asterisk-Users] RE: SIP Peers unreachable

I've just noticed in my last posting that the tethereal capture was from a
2nd setup I have here. In any event, I'm seeing the same problem on both my
setups.

Sorry for the confusion.

Uriel Carrasquilla

2003-May-03 08:00 UTC

head link

[Asterisk-Users] SIP Peers unreachable

I have exactly the same problem using Xten.  I have tried with different
codecs such as ulaw, 711 and gsm.  My extension does ring and after two
rings it hangs up.

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris
Sent: Friday, May 02, 2003 12:42 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Peers unreachable

Hi Everyone,

I'm new to * and I'm trying to setup a small configuration of SIP
clients.
Eventually when I get this working I plan on expanding with a Digium
developers kit to add analog phones and PSTN access.

My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both
peers seem to register with * but I cannot call to one another. When I dial
the associated extension, the call goes to the programmed voicemail
extension (busy) yet if I create an extension to call out through the proxy
(IX66), I can still reach my destination. It's just calling within * there
is a problem. I suspect it's because the status is unreachable but I'm
not
sure how to fix it.

Here is the sip show peers output.

Name/username    Host                 Mask             Port     Status
sipset/sipset    192.200.14.31    (D)  255.255.255.255  5060     UNREACHABLE
sippc/sippc      192.200.14.33    (D)  255.255.255.255  5060     UNREACHABLE

Here is the sip.conf settings:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
register => 9055551212@somewhere.homeip.net

[sippc]
type=friend
username=sippc
secret=blah
host=dynamic
qualify=3000

[sipset]
type=friend
username=sipset
secret=blah
host=dynamic
qualify=3000

Here is the extensions.conf settings:
exten => 421,1,Dial(SIP/sipset)  	  ; Mitel 5055 SIP Phone
exten => 421,2,Voicemail(u421)
exten => 421,102,Voicemail(b421)
exten => 422,1,Dial(SIP/sippc)  	  ; Xten client
exten => 422,2,Voicemail(u422)
exten => 422,102,Voicemail(b422)

exten => 444,1,Dial(SIP/tony@somewhere.homeip.net)  ; friends MSN (4.6)
account registered to IX66

These are the console messages when I dial 421 from 422

    -- Executing Dial("SIP/sippc-b5f6", "SIP/sipset") in new
stack
  == Everyone is busy at this time
    -- Executing VoiceMail("SIP/sippc-b5f6", "b421") in new
stack
  == Parsing '/etc/asterisk/voicemail.conf':   == Parsing
'/etc/asterisk/voicemail.conf': Found
    -- Playing 'vm-theperson'
    -- Playing 'digits/4'
    -- Playing 'digits/2'
    -- Playing 'digits/1'
    -- Playing 'vm-isonphone'
    -- Playing 'vm-intro'
  == Spawn extension (default, 421, 102) exited non-zero on
'SIP/sippc-b5f6'

Any help is appreciated.

Thanks.

Chris

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