[Asterisk-Dev] SIP Registration failed at Chan_sip.c Line 4854

[Pathuri]

Hi,

I have installed Asterisk PNX on Redhad Linux 7.3 machine.
I am trying to test whether two SIP softphones {extensions} can talk to each
other.

I Do NOT have VOCAL or SER etc.. I just have my Linux box and other PC on My
LAN.

I am using UBIQUITY's softphone to test. I always get Registratio Failed
in "chan_sip.c" Line 4854.

I am appending my sip & extensiosn conf file for your reference.

Please help.

Best Regards.
Hanu
-----------------------------------------------------------

;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr =172.16.2.99           ; Address to bind to
context = default               ; Default for incoming calls
;srvlookup = yes                ; Enable SRV lookups on outbound calls
;pedantic = yes                 ; Enable slow, pedantic checking for Pingtel
;tos=lowdelay
;tos=184
;maxexpirey=3600                ; Max length of incoming registration we
allow
;defaultexpirey=120             ; Default length of incoming/outoing
registratio
n
;notifymimetype=text/plain      ; Allow overriding of mime type in NOTIFY
;videosupport=yes               ; Turn on support for SIP video
;
;register => 1234 at mysipprovider.com     ; Register with a SIP provider
;register => 2345 at mysipprovider.com/1234 ; Register 2345 at sip provider as
1234
 here.
;
;[snomsip]
;type=friend
;secret=blah
;host=dynamic
;dtmfmode=inband                ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59
;mailbox=1234,2345              ; Mailbox for message waiting indicator

;[pingtel]
;type=friend
;username=pingtel
;secret=blah
;host=dynamic
;qualify=1000                   ; Consider it down if it's 1 second to reply
;callgroup=1,3-4
;pickupgroup=1,3-4
;defaultip=192.168.0.60

;[cisco]
;type=friend
;username=cisco
;secret=blah
;nat=yes                        ; This phone may be natted
;host=dynamic
;canreinvite=no                 ; Cisco poops on reinvite sometimes
;qualify=200                    ; Qualify peer is no more than 200ms away
;defaultip=192.168.0.4

;[cisco1]
;type=friend
;username=cisco1
;fromuser=markster              ; Specify user to put in "from" instead of
calle
rid
;secret=blah
;host=dynamic
;defaultip=192.168.0.4
;amaflags=default               ; Choices are default, omit, billing,
documentat
ion
;accountcode=markster           ; Users may be associated with an
accountcode tp
 ease billing
[hanu]
type=friend
username=hanu
secret=hanu
host=172.16.2.189


[reddy]
type=friend
username=reddy
secret=reddy



----------------- I added the following lines in the deafult
extensions.conf -------- file


exten => reddy,1,Dial(SIP/reddy,20)
exten => hanu,1,Dial(SIP/hanu,20)


-----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com]On Behalf Of Benjamin Miller
Sent: Thursday, August 14, 2003 10:34 AM
To: asterisk-dev at lists.digium.com
Subject: RE: [Asterisk-Dev] Interface names (in AddQueueMember)


This seems to be an important request.
Adding a member that is the generic version and not the specific version
to the structure would be _very_ valuable.  There are a number of other
places that the "-XXXX" has to be stripped to do comparisons, such as in
the manager interface, etc.
There are times when the specific instance of the channel are needed
(such as call monitoring), and other where the generic reference are
needed, this being an excellent example.
Mark, would this be too difficult to do or are we thinking about this
the wrong way?
Ben

-----Original Message-----
From: Jordyn Buchanan [mailto:jbuchanan at registrypro.pro]
Sent: Wednesday, August 13, 2003 12:16 PM
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] Interface names (in AddQueueMember)


[My first attempts to send this to the list seems not to have worked; I
apologize for any duplicates.]

Hello:

I've been playing a bit with the AddQueueMember application and have
realized that its handling of SIP interfaces (and probably other types
that use a similar channel naming convention) is a bit wrong.

Basically, if you don't specify an interface to be added to the queue,
it will take the name of the calling channel and add that to the queue.
  In the case of a SIP channel, this is something like "SIP/blah-XXXX"
where XXXX is some random string to differentiate between various calls
to/from the same entity.  Adding this whole thing

I've put together a fix to this that looks for SIP interfaces and
truncates the name beginning with the final "-".  This works fine, but
won't work for other types of channels that use similar naming
conventions unless I manually add similar logic for each of them.

So, finally, to a question:  is there a generalized way to derive the
callable name of a given channel other than stripping off some of the
text, as I've been doing?  I was hoping that the ast_channel structure
would contain something useful, but I don' t see anything that will
obviously help me out.

Jordyn
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