[Freeswitch-users] Aastra phone registration lost
Aloysius Lloyd
lloyd.aloysius at gmail.com
Fri Feb 18 23:43:21 MSK 2011
I did some traces and figure it out the problem losing the Registrations.
1. FreeSWITCH Send the response to a wrong port . Is this a Bug?
2. Why FreeSWITCH + Aastra Registration contact not have the port number ?
3. See the below ...
203 & 202 Aastra Phones on Domain - *aastra.mydomain.com*
But user Agent Shows Polycom. BTW in the same LAN I have two polycom phones
on different domain *foo.mydomain.com*
==============
SIP/2.0 200 OK
Via: SIP/2.0/UDP 173.230.136.12;rport;branch=z9hG4bKZ6DDSDNZ11gvS
*From: <sip:203 at aastra.mydomain.com>;tag=3jFNS2KBvNX0S*
*To: "Ext 202" <sip:203 at aastra.mydomain.com>;tag=B550915E-F491D63B*
CSeq: 8639912 NOTIFY
Call-ID: cdad5068-b542-122e-ef86-fefdade6880c
*Contact: <sip:202 at 192.168.240.128:5060>*
Event: message-summary
*User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.3.1.0769*
Accept-Language: en
Content-Length: 0
=============
Any help on this why sending wrong ports?
Thanks
Lloyd
On Wed, Feb 16, 2011 at 10:34 PM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:
> Hi All,
>
>
> I stay away with Aastra phone for a long time and today I did some tests.
> All of my test ... the phones not reliable with FreeSWITCH
>
> I try both 6731i and 57i with the most recent firmware
>
> Here is the configuration files
>
> *aastra.cfg*
>
> dhcp: 1
> sip digit timeout: 3
> sip dial plan:
> "x+#|xx+*|[2-9]XX[2-9]XXXXXX|1[2-9]XX[2-9]XXXXXX|1XXXXXXXXXX|[2-3]XX|67[2-9]XX[2-9]XXXXXX"
> sip rport: 1
> sip customized
> codec:payload=9;ptime=20;silsupp=off,payload=0;ptime=20;silsupp=off,payload=18;ptime=20;silsupp=off
> #sip registration period: 120
> #sip registration renewal timer: 15
> headset tx gain: -3
> headset sidetone gain: -3
> handset tx gain: -3
> handset sidetone gain: -3
> handsfree tx gain: 0
> handset volume: 5 #RX volumes - user adjustable, so easily changed
> speaker volume: 5
> ringer volume: 3
> web interface enabled: 1
> live dialpad: 1
> missed calls indicator disabled: 1
> suppress dtmf playback: 0
> #audio mode: 2 #0 = speaker (default)1 = headset 2 = speaker/headset 3 =
> headset/speaker
> time server disabled: 0
> time server1: pool.ntp.org
> #directory
> directory 1: internal_list.csv
> directory 2: external_list.csv
>
> *mac.cfg*
>
> directed call pickup: 1
> directed call pickup prefix: **
> #
> sip line1 screen name: Ext 203
> sip line1 display name: Ext 203
> sip line1 auth name: 203
> sip line1 user name: 203
> sip line1 password: *********
> sip line1 vmail: *97
> sip line1 mode: 0
> #
> sip line1 proxy ip: aastra.mydomain.com
> #sip line1 proxy port: 5060
> sip line1 registrar ip: aastra.mydomain.com
> sip line1 registration period: 300
> #sip line1 registrar port: 5060
>
> ------------
>
> 1. FreeSWITCH Registration shows two entires in the internal profile and
> when I try to call to the extension two lines on the phones rings .... no
> idea why this is happen?
>
> Call-ID: d8fd4c8795801cd2
> User: 203 at aastra.mydomain.com
> Contact: "Ext 203" <sip:203 at 192.168.240.106:5060
> ;transport=udp;fs_nat=yes;fs_path=sip%3A203%40173.33.178.49%3A1627%3Btransport%3Dudp>
> Agent: Aastra 6731i/2.6.0.2010
> Status: Registered(UDP-NAT)(unknown) EXP(2011-02-16 22:24:36)
> EXPSECS(310)
> Host: li176-12
> IP: 173.33.178.49
> Port: 1627
> Auth-User: 203
> Auth-Realm: aastra.mydomain.com
> MWI-Account: 203 at aastra.mydomain.com
>
> Call-ID: d8fd4c8795801cd2
> User: 203 at aastra.mydomain.com
> Contact: "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
> Agent: Aastra 6731i/2.6.0.2010
> Status: Registered(UDP)(unknown) EXP(2011-02-16 22:24:36)
> EXPSECS(310)
> Host: li176-12
> IP: 173.33.178.49
> Port: 1627
> Auth-User: 203
> Auth-Realm: aastra.mydomain.com
> MWI-Account: 203 at aastra.mydomain.com
>
>
> 2. After first registration expires .... FreeSWITCH internal registration
> status shows the following entry. When I dial the extension now there is a
> long delay ... FreeSWITCH dialing and waiting then goes to voicemail.
>
> Call-ID: d8fd4c8795801cd2
> User: 203 at aastra.mydomain.com
> Contact: "Ext 203" <sip:203 at 173.33.178.49:1627;transport=udp>
> Agent: Aastra 6731i/2.6.0.2010
> Status: Registered(UDP)(unknown) EXP(2011-02-16 22:31:25)
> EXPSECS(266)
> Host: li176-12
> IP: 173.33.178.49
> Port: 1631
> Auth-User: 203
> Auth-Realm: aastra.mydomain.com
> MWI-Account: 203 at aastra.mydomain.com
>
> 3. Phones Goes randomly "No Service" .... I see this problem several times
> in the past.
>
> -------------
>
> At the same time in my lab Linksys and Polycom Phones working without any
> issues with the default settings.
>
> -------------
>
> my question what is Aastra Doing differently?
>
> Thanks
> Lloyd
>
> On Wed, Feb 16, 2011 at 10:53 AM, Tim St. Pierre <
> fs-list at communicatefreely.net> wrote:
>
>> I have been setting my expires to 600.
>>
>> Brian West wrote:
>> > Are you setting your expires to > 300 seconds?
>> >
>> > /b
>> >
>> > On Feb 15, 2011, at 9:42 AM, Aloysius Lloyd wrote:
>> >
>> >>
>> >>
>> >> I have the same issue, around 275 phones in the field. I want the 275
>> >> phones work with FreeSWITCH.
>> >>
>> >>
>> >> Thanks
>> >> Lloyd
>> >
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
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>> >
>>
>>
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>
>
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