Reconfigured my router/gateway w/ different ip(s) and setup gre vpn config
and now im back to having problems with receiving incoming calls. Same
symptoms, i can call the AA from an outside pstn line but when i ask it to
transfer me to a user extension, it just stays silent for a minute then goes
busy. This time around though, im getting a 503 service unavailable error
instead of the 401 in the debug. Outgoing calls still working. I've attached
my cisco gateway config and results of the debug. I hope someone can shed
some light on this. thnx.
On Mon, Apr 5, 2010 at 7:51 PM, Jean-Hugues Royer <jhro...@joher.com> wrote:
> Hi,
>
> Glad I could help.
>
> Now you got it is working I advise you to take some time when you can to
> investigate why the Cisco was unable by itself (without a configured route)
> to decide where to send the INVITE, you probably have an underlying problem
> that you masked by this quick fix that might reappear in other conditions.
>
> Regards.
>
>
> ronald teng wrote:
>
> Hi Jean,
> Thnx for highlighting to me why it's not working...i got it working
> now....im soooo happy i feel like crying.
> It was an issue w/ one of my dialpeers on cisco.
>
> my previous config was:
>
> dial-peer voice 10 voip
> description *** SIP trunk from router to SipX ***
> *destination-pattern 100*
> voice-class codec 1
> session protocol sipv2
> session target ipv4:10.9.20.254
> dtmf-relay rtp-nte
> no vad
>
> it seems this is the only dialpeer i have pointing from the cisco gateway
> to the sipX machine....so when we ask the gateway to take us to other
> extensions (e.g. 210) it doesn't know where to send it.
> I corrected it to "destination-pattern ..." and everything's now up and
> running. Will be trying to feature sets from here on out. Thnx so much again
> for your and everyone elses input. Couldn't have fixed this w/o ya'll.
>
> Regards,
> Ron
>
>
>
>
>
>
>
> On Sat, Apr 3, 2010 at 8:02 PM, Jean-Hugues Royer <jhro...@joher.com>wrote:
>
>> Hi,
>>
>> I will assume that you meant still not working.
>>
>> From the cisco logs the only thing in which was weird at first sight was
>> that you had IPs in every SIP URI (request/from/to/contact) _but_ in the
>> REFER Refer-To field where you had "2...@alsterph.lan" <2...@alsterph.lan>.
>>
>> Since the cisco accepts the REFER (202 Accepted) but he doesn't generate
>> an INVITE to "2...@alsterph.lan" <2...@alsterph.lan> in response and issue
>> a "503 Service Unavailable", the most obvious answer is that he does support
>> REFER but he is unable to decide where to send the INVITE to
>> "2...@alsterph.lan" <2...@alsterph.lan>.
>>
>> I have very basic cisco knowledge, but the answer probably lies in how the
>> cisco internally decides where to send an INVITE to
>> "2...@alsterph.lan"<2...@alsterph.lan>
>> .
>>
>> Normally if he has no pre-configured way to know it, he should issue a DNS
>> SRV request for "alsterph.lan", from this DNS SRV request he will get the
>> transport (tcp/udp), the host and port where he has to send his INVITE
>> packet.
>>
>> So what I offered you was to first check/create a DNS SRV for
>> "aslterph.lan", or in last resort to use an IP instead of a host/domain in
>> the "2...@alsterph.lan" <2...@alsterph.lan> transfer target which is
>> probably the easiest (but not nicest) way to solve your problem.
>>
>> Regards.
>>
>>
>> ronald teng wrote:
>>
>>> UPDATE: DNS has been fixed so i can now ping just
>>> 'alsterph.lan'...incoming still now working (T_T)
>>>
>>> would i need to use the sip call transfer commands of cisco for this?
>>> According to this link i posted before, i'll need to have this script named
>>> " app_h450_transfer.tcl " in my flash. Any ideas as to where i can find a
>>> copy of this?
>>>
>>>
>>>
>
*Apr 13 14:58:50.363: %SYS-5-CONFIG_I: Configured from console by vty0 (172.16.1
0.200)
AMS-RTR2651#
*Apr 13 14:59:01.982: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
REFER sip:172.16.10.1:5060;x-sipX-nonat SIP/2.0
Via: SIP/2.0/TCP 172.16.10.254;branch=z9hG4bK-XX-437f7n6_KtpCj6ud_z`dzJwQGg
Via: SIP/2.0/UDP 172.16.10.254:15060;rport=15060;branch=z9hG4bK26H3c253482eH;id=
6763-29
Max-Forwards: 20
From: <sip:3...@172.16.10.254>;tag=r8FcmFB0572pe
To: <sip:172.16.10.1>;tag=17AA106-1340
Call-Id: ddf77104-464311df-805add57-37a71...@172.16.10.1
Cseq: 129539492 REFER
Contact: <sip:3...@172.16.10.254:15060;transport=udp;x-sipX-nonat>
Expires: 3600
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER
, NOTIFY
Supported: timer, precondition, path, replaces
Proxy-Authorization: Digest username="~~id~media", realm="ourcompany.lan",
nonce="
355e0cd903af41df87df3feacde470704bc6bcc6", cnonce="VntTWsMBEi2ok5DmutgP/w", algo
rithm=MD5, uri="sip:172.16.10.1:5060;x-sipX-nonat", response="3d7f30b8830e19a367
8fb817ee1f2d3e", qop=auth, nc=00000001
Refer-To: "Ako
Pogi"<sip:3...@ourcompany.lan?x-sipx-authidentity=%3csip%3a%7e%7eid%
7Emedia%40ourcompany.lan%3Bsignature%3D4BC6BCC6%253A%253A8445e1619d6b590673fd345c7
5434ba4%3E&REFERENCES=DDF77104-464311DF-805ADD57-37A714A5%40172.16.10.1%3Brel%3D
refer>
Referred-By: <sip:172.16.10.254:15060>
Content-Length: 0
Date: Thu, 15 Apr 2010 07:14:14 GMT
*Apr 13 14:59:02.026: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
NOTIFY sip:3...@172.16.10.254:15060;transport=udp;x-sipX-nonat SIP/2.0
Via: SIP/2.0/UDP 172.16.10.1:5060;branch=z9hG4bK211D47
From: <sip:172.16.10.1>;tag=17AA106-1340
To: <sip:3...@172.16.10.254>;tag=r8FcmFB0572pe
Call-ID: ddf77104-464311df-805add57-37a71...@172.16.10.1
CSeq: 102 NOTIFY
Max-Forwards: 15
Date: Tue, 13 Apr 2010 14:59:02 GMT
User-Agent: Cisco-SIPGateway/IOS-12.x
Event: refer
Subscription-State: pending;expires=180
Route: <sip:172.16.10.254:5060;lr;sipXecs-CallDest=AA;sipXecs-rs=%2Aauth%7E.%2Af
rom%7EMTdBQTEwNi0xMzQw.900_ntap%2Aid%7ENjc2My0yOQ%60%60%21237d8ae0582de2db3617eb
5d18300cec;x-sipX-done>
Contact: <sip:172.16.10.1:5060>
Content-Type: message/sipfrag
Content-Length: 22
SIP/2.0 100 Trying
*Apr 13 14:59:02.034: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 202 Accepted
Via: SIP/2.0/TCP 172.16.10.254;branch=z9hG4bK-XX-437f7n6_KtpCj6ud_z`dzJwQGg,SIP/
2.0/UDP 172.16.10.254:15060;rport=15060;branch=z9hG4bK26H3c253482eH;id=6763-29
From: <sip:3...@172.16.10.254>;tag=r8FcmFB0572pe
To: <sip:172.16.10.1>;tag=17AA106-1340
Date: Tue, 13 Apr 2010 14:59:01 GMT
Call-ID: ddf77104-464311df-805add57-37a71...@172.16.10.1
CSeq: 129539492 REFER
Content-Length: 0
Contact: <sip:172.16.10.1:5060>
AMS-RTR2651#
*Apr 13 14:59:02.063: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.1:5060;branch=z9hG4bK211D47;id=6763-29
From: <sip:172.16.10.1>;tag=17AA106-1340
To: <sip:3...@172.16.10.254>;tag=r8FcmFB0572pe
Call-Id: ddf77104-464311df-805add57-37a71...@172.16.10.1
Cseq: 102 NOTIFY
Contact: <sip:3...@172.16.10.254:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER
, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0
Date: Thu, 15 Apr 2010 07:14:14 GMT
AMS-RTR2651#
*Apr 13 14:59:20.116: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
NOTIFY sip:3...@172.16.10.254:15060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.10.1:5060;branch=z9hG4bK221153
From: <sip:172.16.10.1>;tag=17AA106-1340
To: <sip:3...@172.16.10.254>;tag=r8FcmFB0572pe
Call-ID: ddf77104-464311df-805add57-37a71...@172.16.10.1
CSeq: 103 NOTIFY
Max-Forwards: 15
Date: Tue, 13 Apr 2010 14:59:20 GMT
User-Agent: Cisco-SIPGateway/IOS-12.x
Event: refer
Subscription-State: terminated;reason=noresource
Route: <sip:172.16.10.254:5060;lr;sipXecs-CallDest=AA;sipXecs-rs=%2Aauth%7E.%2Af
rom%7EMTdBQTEwNi0xMzQw.900_ntap%2Aid%7ENjc2My0yOQ%60%60%21237d8ae0582de2db3617eb
5d18300cec;x-sipX-done>
Contact: <sip:172.16.10.1:5060>
Content-Type: message/sipfrag
Content-Length: 35
SIP/2.0 503 Service Unavailable
*Apr 13 14:59:20.128: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.1:5060;branch=z9hG4bK221153;id=6763-29
From: <sip:172.16.10.1>;tag=17AA106-1340
To: <sip:3...@172.16.10.254>;tag=r8FcmFB0572pe
Call-Id: ddf77104-464311df-805add57-37a71...@172.16.10.1
Cseq: 103 NOTIFY
Contact: <sip:3...@172.16.10.254:15060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER
, NOTIFY
Supported: timer, precondition, path, replaces
Content-Length: 0
Date: Thu, 15 Apr 2010 07:14:32 GMT
*Apr 13 14:59:20.140: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:172.16.10.1:5060;x-sipX-nonat SIP/2.0
Via: SIP/2.0/UDP 172.16.10.254;branch=z9hG4bK-XX-4396UmGzD5bQc_kt_Mllt92C8Q
Via: SIP/2.0/UDP 172.16.10.254:15060;rport=15060;branch=z9hG4bK3FBveXp71HS1c
Max-Forwards: 20
From: <sip:3...@172.16.10.254>;tag=r8FcmFB0572pe
To: <sip:172.16.10.1>;tag=17AA106-1340
Call-Id: ddf77104-464311df-805add57-37a71...@172.16.10.1
Cseq: 129539493 BYE
Contact: <sip:3...@172.16.10.254:15060;transport=udp;x-sipX-nonat>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER
, NOTIFY
Supported: timer, precondition, path, replaces
Proxy-Authorization: Digest username="~~id~media", realm="ourcompany.lan",
nonce="
355e0cd903af41df87df3feacde470704bc6bcc6", cnonce="VntTWsMBEi2ok5DmutgP/w", algo
rithm=MD5, uri="sip:172.16.10.1:5060;x-sipX-nonat", response="1ebda020a66e00f9e9
302dfa78845c59", qop=auth, nc=00000002
Content-Length: 0
Date: Thu, 15 Apr 2010 07:14:32 GMT
AMS-RTR2651#
*Apr 13 14:59:20.168: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.254;branch=z9hG4bK-XX-4396UmGzD5bQc_kt_Mllt92C8Q,SIP/
2.0/UDP 172.16.10.254:15060;rport=15060;branch=z9hG4bK3FBveXp71HS1c
From: <sip:3...@172.16.10.254>;tag=r8FcmFB0572pe
To: <sip:172.16.10.1>;tag=17AA106-1340
Date: Tue, 13 Apr 2010 14:59:20 GMT
Call-ID: ddf77104-464311df-805add57-37a71...@172.16.10.1
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 129539493 BYE
Reason: Q.850;cause=16
Content-Length: 0
!
voice service voip
allow-connections sip to sip
sip
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
!
!
!
!
!
!
!
!
!
!
!
voice source-group secured
access-list 1
disconnect-cause call-reject
!
!
!
!
!
!
archive
log config
hidekeys
!
!
crypto isakmp policy 1
encr 3des
authentication pre-share
group 2
crypto isakmp key ourcompany address 100.0.0.254
!
!
crypto ipsec transform-set ESP-3DES-SHA esp-3des esp-sha-hmac
!
crypto map SDM_CMAP_1 1 ipsec-isakmp
description Apply the crypto map on the peer router's interface having IP addre
ss 100.0.0.253 that connects to this router.
set peer 100.0.0.254
set transform-set ESP-3DES-SHA
match address SDM_1
!
!
!
!
!
!
!
interface Tunnel0
ip address 10.11.12.253 255.255.255.0
ip mtu 1420
tunnel source FastEthernet0/0
tunnel destination 100.0.0.254
tunnel path-mtu-discovery
crypto map SDM_CMAP_1
!
interface FastEthernet0/0
ip address 100.0.0.253 255.255.255.0
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
crypto map SDM_CMAP_1
!
interface FastEthernet0/1
ip address 10.1.1.1 255.255.255.0
duplex auto
speed auto
!
interface FastEthernet0/1.10
encapsulation dot1Q 10
ip address 172.16.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
!
interface FastEthernet0/1.20
encapsulation dot1Q 20
ip address 172.16.20.1 255.255.255.0
ip nat inside
ip virtual-reassembly
!
interface FastEthernet0/1.30
encapsulation dot1Q 30
ip address 172.16.30.1 255.255.255.0
ip nat inside
ip virtual-reassembly
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 100.0.0.254
!
!
ip http server
no ip http secure-server
ip nat inside source route-map SDM_RMAP_1 interface FastEthernet0/0 overload
!
ip access-list extended DATA
permit ip 172.16.20.0 0.0.0.255 192.168.20.0 0.0.0.255
permit ip 172.16.20.0 0.0.0.255 192.168.30.0 0.0.0.255
permit ip 172.16.30.0 0.0.0.255 192.168.20.0 0.0.0.255
permit ip 172.16.30.0 0.0.0.255 192.168.30.0 0.0.0.255
ip access-list extended NAT
remark SDM_ACL Category=18
deny gre host 100.0.0.253 host 100.0.0.254
deny ip 172.16.0.0 0.0.255.255 192.168.0.0 0.0.255.255
permit ip 172.16.0.0 0.0.255.255 any
ip access-list extended SDM_1
remark SDM_ACL Category=4
permit gre host 100.0.0.253 host 100.0.0.254
ip access-list extended VOICE
permit ip 172.16.10.0 0.0.0.255 192.168.10.0 0.0.0.255
!
!
!
!
route-map SDM_RMAP_1 permit 1
match ip address NAT
!
!
!
control-plane
!
!
!
voice-port 1/1/0
supervisory disconnect dualtone pre-connect
supervisory answer dualtone
input gain 8
no vad
cptone PH
connection plar 300
!
voice-port 1/1/1
!
!
!
!
dial-peer cor custom
!
!
!
dial-peer voice 21 pots
description *** Outbound to PSTN using FXO ***
destination-pattern .T
port 1/1/0
!
dial-peer voice 10 voip
description *** SIP trunk from router to SipX ***
destination-pattern ...
voice-class codec 1
session protocol sipv2
session target ipv4:172.16.10.254
dtmf-relay rtp-nte
no vad
!
dial-peer voice 20 voip
description *** SIP trunk from SipX to router ***
voice-class codec 1
session protocol sipv2
session target sip-server
incoming called-number .T
dtmf-relay rtp-nte
fax protocol pass-through g711alaw
no vad
!
!
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 1200
!
sip-ua
max-forwards 15
sip-server ipv4:172.16.10.254
!
!
!
end
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