[Freeswitch-users] Aastra phone registration lost

Aloysius Lloyd lloyd.aloysius at sunteltech.ca
Sat Feb 19 04:39:40 MSK 2011


FreeSWITCH is on public IP and Phones Behind the NAT. I see sometimes
FreeSWITCH send the reply to a wrong port or not responses for the
registration request.

Please see the following contact

Call-ID:        1bb9e78e1cba12f8
User:           202 at aastra.mydomain.com
*Contact:        "Ext 202" <sip:202 at 173.33.178.49:1287;transport=udp>*
Agent:          Aastra 6731i/2.6.0.2010
Status:         Registered(UDP)(unknown) EXP(2011-02-18 20:42:53)
EXPSECS(358)
Host:           li176-12
IP:             173.33.178.49
Port:           1287
Auth-User:      202
Auth-Realm:     aastra.mydomain.com
MWI-Account:    202 at aastra.mydomain.com


---------

I solve the problem using the <param name="sip-force-contact" value="
NDLB-connectile-dysfunction"/> in the user directory.

But why this is happening to Aastra Phones only? Please let me know.

Thanks
Lloyd
On Fri, Feb 18, 2011 at 8:29 PM, Michael Collins <msc at freeswitch.org> wrote:

> Get the SIP trace of the phone doing its registration. Also, is the phone
> on the same LAN as FreeSWITCH? Is there any NAT involved?
> -MC
>
>
> On Fri, Feb 18, 2011 at 12:43 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>> 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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>
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