Re: [Siproxd-users] Configuration help needed for Cisco 7905

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I am coming back to the list with this old problem, because although I have made some progress I still can't get my Cisco 7905 handset to work behind my NAT router and siproxd, though more modern softphones are fine.

The problem that I originally reported below had to do with the Cisco not being able to reach DNS - that is now fixed, and I can now make outgoing calls (even without siproxd).  The problem comes with incoming calls.

In order to receive incoming calls I'm registering the Cisco with siproxd, and the registration is showing up correctly in the siproxd_registrations file:

***:1:1376210552
sip
cisco
192.168.15.122                   # this is the local network address of the phone
5060
sip
cisco
128.107.224.32                   # this is the external IP address

sip
cisco
my.sip.server.net                 # this is my SIP server

When I try to dial in my other softphones will ring, but the Cisco won't.  The siproxd log shows:

Aug 11 16:18:17 servalan siproxd[27573]: plugin_logcall.c:126 INFO:Outgoing Call: 000...@my... -> cisco@192.168.15.200 [Req: cisco@192.168.15.200]
Aug 11 16:18:18 servalan siproxd[27573]: plugin_logcall.c:126 INFO:ACK Call: 000...@my... -> cisco@192.168.15.200 [Req: cisco@192.168.15.200]

(192.168.15.200 is the machine running siproxd.)  A packet dump shows this:

U my.sip.server.net:5060 -> 192.168.15.200:5060
INVITE sip:cisco@192.168.15.200:5060 SIP/2.0.
Max-Forwards: 14.
Via: SIP/2.0/UDP my.sip.server.net:5060;rport;branch=z9hG4bK1412142834.
From: "qirtaiba" <sip:000...@my...:5060>;tag=1715387067.
To: <sip:cisco@192.168.15.200:5060>.
Call-ID: 897...@my....
CSeq: 143 INVITE.
User-Agent: YATE/4.1.0.
Contact: <sip:000...@my...:5060>.
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO.
Content-Type: application/sdp.
Content-Length: 209.
.
v=0.
o=yate 1376210100 1376210100 IN IP4 85.234.150.85.
s=SIP Call.
c=IN IP4 my.sip.server.net.
t=0 0.
m=audio 32108 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.

#
U 192.168.15.200:5060 -> my.sip.server.net:5060
SIP/2.0 408 Request Timeout.
Via: SIP/2.0/UDP my.sip.server.net:5060;rport;branch=z9hG4bK1412142834.
From: "qirtaiba" <sip:000...@my...:5060>;tag=1715387067.
To: <sip:cisco@192.168.15.200:5060>.
Call-ID: 897...@my....
CSeq: 143 INVITE.
Content-Length: 0.
.

#
U my.sip.server.net:5060 -> 192.168.15.200:5060
ACK sip:cisco@192.168.15.200:5060 SIP/2.0.
Via: SIP/2.0/UDP my.sip.server.net:5060;rport;branch=z9hG4bK1412142834.
From: "qirtaiba" <sip:000...@my...:5060>;tag=1715387067.
To: <sip:cisco@192.168.15.200:5060>.
Call-ID: 897...@my....
CSeq: 143 ACK.
Max-Forwards: 14.
Contact: <sip:000...@my...:5060>.
User-Agent: YATE/4.1.0.
Content-Length: 0.
.

#
U 192.168.15.200:5060 -> my.sip.server.net:5060
SIP/2.0 408 Request Timeout.
Via: SIP/2.0/UDP my.sip.server.net:5060;rport;branch=z9hG4bK1412142834.
From: "qirtaiba" <sip:000...@my...:5060>;tag=1715387067.
To: <sip:cisco@192.168.15.200:5060>.
Call-ID: 897...@my....
CSeq: 143 ACK.
Content-Length: 0.

Any more help would be much appreciated!

On 07/04/2013, at 3:54 PM, Jeremy Malcolm <Je...@Ma...> wrote:

> My configuration is like this one:
> 
> http://siproxd.sourceforge.net/siproxd_guide/siproxd_guide_c7s3.html
> 
> However my phone is different; it's a Cisco 7905.  The relevant settings that the phone allows me to configure are these:
> 
> UID	 cisco
> PWD	 mypassword
> Proxy	 my.sip.server.net
> SIPRegOn     1
> SIPRegInterval	 3600
> StaticIP    192.168.15.122
> OutBoundProxy	 192.168.15.200 [my siproxd machine]
> StaticRoute    192.168.15.1 [my dumb masquerading router, with siproxd set as the DMZ]
> 
> Here is my configuration for siproxd:
> 
> if_inbound  = eth0
> if_outbound = eth0
> host_outbound = name-that-resolves-to-my-external-ip.net
> 
> It doesn't work, here is what happens:
> 
> 15:31:35 INFO:siproxd.c:233 siproxd-0.8.1-53 i486-pc-linux-gnu starting up
> 15:31:35 INFO:plugins.c:112 Plugin 'plugin_logcall' [Logs calls to syslog] loaded with success, exemask=0x40
> 07:31:35 INFO:rtpproxy_relay.c:121 Current thread stacksize is 8192 kB
> 07:31:35 INFO:sock.c:131 bound to port 5060
> 07:31:35 INFO:siproxd.c:344 siproxd-0.8.1-53 i486-pc-linux-gnu started
> 07:31:57 INFO:plugin_logcall.c:126 Outgoing Call: ci...@my... -> 888...@my... [Req: 888...@my...]
> 07:31:57 INFO:plugin_logcall.c:126 ACK Call: ci...@my... -> 888...@my... [Req: 888...@my...]
> 
> Also by using ngrep I see this:
> 
> interface: eth0 (192.168.15.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
> #
> U 192.168.15.122:5060 -> 192.168.15.200:5060
> REGISTER sip:my.sip.server.net SIP/2.0.
> Via: SIP/2.0/UDP 192.168.15.122:5060.
> From: sip:ci...@my...;tag=3840067541.
> To: sip:ci...@my....
> Call-ID: 4035595390@192.168.15.122.
> CSeq: 1 REGISTER.
> Contact: Jeremy Malcolm <sip:cisco@192.168.15.122:5060;transport=udp>;expires=3600.
> User-Agent: Cisco-CP7905/1.01-030807A.
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER.
> Content-Length: 0.
> .
> 
> #
> U 192.168.15.200:5060 -> 192.168.15.122:5060
> SIP/2.0 408 Request Timeout.
> Via: SIP/2.0/UDP 192.168.15.122:5060.
> From: <sip:ci...@my...>;tag=3840067541.
> To: <sip:ci...@my...>.
> Call-ID: 4035595390@192.168.15.122.
> CSeq: 1 REGISTER.
> Content-Length: 0.
> .
> 
> #
> U 192.168.15.122:5060 -> 192.168.15.200:5060
> INVITE sip:888...@my...;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 192.168.15.122:5060.
> From: Jeremy Malcolm <sip:ci...@my...>;tag=816758886.
> To: <sip:888...@my...;user=phone>.
> Call-ID: 2441701698@192.168.15.122.
> CSeq: 1 INVITE.
> Contact: Jeremy Malcolm <sip:cisco@192.168.15.122:5060;transport=udp>.
> User-Agent: Cisco-CP7905/1.01-030807A.
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER.
> Expires: 300.
> Content-Length: 262.
> Content-Type: application/sdp.
> .
> v=0.
> o=cisco 15826 15826 IN IP4 192.168.15.122.
> s=Cisco 7905 SIP Call.
> c=IN IP4 192.168.15.122.
> t=0 0.
> m=audio 16384 RTP/AVP 18 8 0 101.
> a=rtpmap:18 G729/8000/1.
> a=rtpmap:8 PCMA/8000/1.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> 
> #
> U 192.168.15.200:5060 -> 192.168.15.122:5060
> SIP/2.0 408 Request Timeout.
> Via: SIP/2.0/UDP 192.168.15.122:5060.
> From: Jeremy Malcolm <sip:ci...@my...>;tag=816758886.
> To: <sip:888...@my...;user=phone>.
> Call-ID: 2441701698@192.168.15.122.
> CSeq: 1 INVITE.
> Content-Length: 0.
> .
> 
> #
> U 192.168.15.122:5060 -> 192.168.15.200:5060
> ACK sip:888...@my...;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 192.168.15.122:5060.
> From: Jeremy Malcolm <sip:ci...@my...>;tag=816758886.
> To: <sip:888...@my...;user=phone>.
> Call-ID: 2441701698@192.168.15.122.
> CSeq: 1 ACK.
> User-Agent: Cisco-CP7905/1.01-030807A.
> Content-Length: 0.
> 
> Interestingly there is no sign of "name-that-resolves-to-my-external-ip.net" above.  I have tried running a similar command on my.sip.server.net (because it's my own SIP server running yate), but it never sees anything.
> 
> Also interestingly, I have sip softphones on the same network which work fine, even without using siproxd as the proxy.  I guess they are a bit smarter about NAT, and can work fine even though my router does not have a sip ALG built-in.
> 
> Any help appreciated...
> 
> PS. Public IP addresses/domains have been changed for privacy/security reasons.

--
Jeremy Malcolm PhD LLB (Hons) B Com
Internet and Open Source lawyer, consumer advocate and geek
host -t NAPTR 5.9.8.5.2.8.2.2.1.0.6.e164.org|awk -F! '{print $3}'

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