[asterisk-biz] Asterisk Router Question, kind of long, sorry

[Jim Houser]

Here's my issue:

I am using Mikrotik for the gateway router with Asterisk behind it.  All has
been fine until recently as my VoIP provider stopped providing IAX and only
does SIP.  I changed to SIP and only have 1-way audio.  I've been going back
and forth verifying my Piaf configuration with my SIP provider and it ended
up with the item in question being my router.  I am behind a NAT, (as you
would guess), and have SIP port forwarding in place but do not have a HUGE
range of RTP ports forwarded.  Read further to understand why.

Here's my first question:
If you look at the attached SIP debug info it seem pretty clear that my SIP
messages are getting changed when they reach my SIP provider.  I have NO
idea how this is happening.  My Mikrotik has next to no programming in it
for my home office.  It simply NATs inside from outside, has IAX and SIP
forwarding in place.  ???  Has this happened to other people using Mikrotik?

Here's my second question:
Let's assume I need to replace my Mikrotik.  I was wondering what everyone
uses for routers.  If you would be so kind you can answer this question at
http://www.surveymonkey.com/s/WSH6B2D   This is simply a survey asking "what
router do you use with Asterisk?".  I will be happy to share the results.
If I forgot your brand, please forgive me and add it in the "other"
selection.

Thanks to all who reply.
 

NOTE: the SIP message account, provider url and public IP are changed as I
did not have their permission to post this.

This is what my Voip provider is seeing.
Here's why I do not have a TON of RTP ports forwarded.
I am glad the provider is willing to deal with this as the RTP range is so
wide.
They say I should be setting Piaf with no externip= value and let them worry
about NAT.

Comment from SIP provider:

There IS externip or externhost parameter set. Look at registration request
from you, it has external IP address in Via and Contact headers: 

My reply to SIP provider with SIP debug attached.
This is what Asterisk says it is sending.
This was copied from a SSH session at CLI>

REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 38.99.70.232:5060:
REGISTER sip:sip.PROVIDER.com SIP/2.0
Via: SIP/2.0/UDP 192.168.111.110:5060;branch=z9hG4bK75f7e979;rport
From: <sip:(MY_ACCOUNT)@sip.PROVIDER.com>;tag=as260ed57a
To: <sip:MY_ACCOUNT at sip.PROVIDER.com>
Call-ID: 0c725c37189dbe7a24de0d9c537977f6 at 127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="MY_ACCOUNT", realm="sip.PROVIDER.com",
algorithm=MD5, uri="sip:sip.PROVIDER.com",
nonce="4b4a910900001d4f8bde7829027049cd6a7e67724f78fe10",
response="6c91845c0fb80d385dbcc17a48a0a189" Expires: 120
Contact: <sip:s at 192.168.111.110>
Event: registration
Content-Length: 0

Ha, looks like your router knows something about SIP protocol, substitutes
addresses and breaks everything :-) 

Here is what server getting for the same request (call id
0c725c37189dbe7a24de0d9c537977f6 at 127.0.0.1) :

Jan 10 21:43:18 t2h /usr/local/sbin/kamailio[19956]: SIP message from
udp:72.198.223.192:5060
REGISTER sip:sip.PROVIDER.com SIP/2.0
Via: SIP/2.0/UDP MY_PUBLIC_IP:5060;branch=z9hG4bK75f7e979;rport
From: <sip:MY_ACCOUNT at sip.PROVIDER.com>;tag=as260ed57a
To: <sip:MY_ACCOUNT at sip.PROVIDER.com>
Call-ID: 0c725c37189dbe7a24de0d9c537977f6 at 127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 69
Authorization: Digest username="MY_ACCOUNT", realm="sip.PROVIDER.com",
algorithm=MD5, uri="sip:sip.PROVIDER.com", nonce="4b4a
910900001d4f8bde7829027049cd6a7e67724f78fe10",
response="6c91845c0fb80d385dbcc17a48a0a189"
Expires: 120
Contact: <sip:s at MY_PUBLIC_IP:5060>
Event: registration
Content-Length: 0