[asterisk-gui] sip phones and asterisk 1.4.0

[Eric Swanson]

I have 3 sip phone 2 polycoms inside the firewall and 1 grandstream outside the firewall.  When I had asterisk 1.2.?, we were able to call from the polycom phones to the grandstream and talk.  Now when a call is made from one polycom phone to the other polycom phone, we can not hear each other.  I was able to fix this by added canreinvite=yes to each polycom phone.  

For the grandstream phone, they can call a polycom phone, and the polycom phone and grandstream phone can hear each other.  If a call is made from the polycom phone to the grandstream phone, each other can not hear the other.  Do I need to adjust the canreinvite status also?

What does the canreinvite status do in asterisk 1.4.0?

Here is the user.conf file:

[2001]
fullname = 2001
secret = ????
email=????
cid_number = 2001
zapchan =
context = numberplan-custom-1
hasvoicemail = yes
hasdirectory = yes
hassip = yes
hasiax = no
hasmanager = no
callwaiting = yes
threewaycalling = yes
mailbox = 2001
hasagent = yes
group =
host = dynamic
progressinband = no
nat=no
canreinvite=yes


[2002]
fullname = 2002
secret = ????
cid_number = 2002
zapchan =
context = numberplan-custom-1
hasvoicemail = yes
hasdirectory = yes
hassip = yes
hasiax = no
hasmanager = no
callwaiting = yes
threewaycalling = yes
mailbox = 2002
hasagent = yes
group =
host = dynamic
progressinband = no
nat=no
canreinvite=yes

[1131]
fullname = 1131
secret = ????
email = ?????
cid_number = 1131
zapchan =
context = numberplan-custom-1
hasvoicemail = yes
hasdirectory = yes
hassip = yes
hasiax = no
hasmanager = no
callwaiting = yes
threewaycalling = yes
mailbox = 1131
hasagent = yes
group =
host = dynamic
nat=yes
canreinvite=no

I appreciate your assistance.  Also does have the gui the capability to setting more sip parameters, instead of hand editing them?

Thanks

Eric

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Sent: Sat, 2/17/2007 1:00pm
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Today's Topics:

   1. Re: outside sip phones issues (Nandy Dagondon)


----------------------------------------------------------------------

Message: 1
Date: Sat, 17 Feb 2007 07:16:54 +0800
From: Nandy Dagondon <nandy1925 at gmail.com>
Subject: Re: [asterisk-gui] outside sip phones issues
To: Asterisk GUI project discussion <asterisk-gui at lists.digium.com>
Message-ID: <45D63B66.4040600 at gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

eric,

take a look at this website. they hv detailed explnation on SIP NAT setup:

http://www.fridu.org/index.php?option=com_content&task=category&sectionid=6&id=27&Itemid=55

Nandy



Eric Swanson wrote:
> Hello,
>
> I have some sip phones are outside the firewall.  I have manually 
> edited the user.conf file so that nat will be in use, but if I dial 
> these phones, the outside phones can hear the phone ringing and when 
> they pick it up and talk, I can not hear them.  From a phone inside 
> the firewall, if I call them, I do not hear a ringing sound.  If they 
> (an outside sip phone) call back, we can hear them.
>
> This was working with asterisk 1.2.?.  What have I done to screw up 
> this configuration?  It also appears that the nat flag is not being 
> displayed properly in the sip show peers.
>
> Any assistance would be appreciated.
>
> Thanks
>
> Eric
> ------------------------------------------------------------------------
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