[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call

James Sizemore james at deny.org
Fri Mar 12 12:57:56 MST 2004 

The pings are pinging the out side port on the nat device,  You don't 
have a
rule in your nat table to associate it with a device on the inside.  You 
should
reset the phone and then see if the qualify shows a return time.  You will
need to make the phone register every time you change you config till the
qualify shows a time. A good way to do this is to reboot the phone.
Your nat device will have a default time that it keep nat rules in its 
table.
Your qualify time will need to be lower then this value.

AstGrp wrote:

>Ok...
>
>If put in the qualify=500... It says it is unreachable... But ping
>times.... Are fine...
>
>PING 69.133.182.77 (69.133.182.77) from 10.100.254.21 : 56(84) bytes of
>data. 64 bytes from 69.133.182.77: icmp_seq=1 ttl=241 time=54.9 ms 64
>bytes from 69.133.182.77: icmp_seq=2 ttl=241 time=52.0 ms 64 bytes from
>69.133.182.77: icmp_seq=3 ttl=241 time=54.2 ms 64 bytes from
>69.133.182.77: icmp_seq=5 ttl=241 time=57.9 ms 64 bytes from
>69.133.182.77: icmp_seq=6 ttl=241 time=56.0 ms 64 bytes from
>69.133.182.77: icmp_seq=7 ttl=241 time=54.0 ms
>
>Any thoughts there?
>
>-----Original Message-----
>From: asterisk-users-admin at lists.digium.com
>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of James
>Sizemore
>Posted At: Friday, March 12, 2004 11:50 AM
>Posted To: Asterisk User Group
>Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
>retries exceeded on call
>Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
>retries exceeded on call
>
>
>I have noticed that sometimes you need to comment out profiles with
>nat=yes on and then reload, then uncomment them and reload, for Asterisk
>to clean out historical settings. Try that.  I have run phones before on
>odd port with out trouble, so I don't think that is your problem.
>
>AstGrp wrote:
>
>  
>
>>Ok.. Let me start by saying that SJPhone works fine through NAT and the
>>    
>>
>
>  
>
>>Cisco phones inside the internal network work fine also... It's just 
>>the Cisco phones on the outside using NAT.
>>
>>For Testing I opened the Firewall open on the IP for the * Server.  I 
>>have done, everything you recommended below, but still no go... When 
>>the phone registers with port 2842?  Not the standard 5060?  Any ideas?
>>    
>>
>
>  
>
>>I believe this is where my problem sits...
>>
>>Thanks,
>>
>>-gcc
>>
>>
>>-----Original Message-----
>>From: asterisk-users-admin at lists.digium.com
>>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of James 
>>Sizemore Posted At: Friday, March 12, 2004 9:03 AM
>>Posted To: Asterisk User Group
>>Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
>>retries exceeded on call
>>Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
>>retries exceeded on call
>>
>>
>>Make sure your using qualify=500 in the sip.conf along with nat=yes, 
>>make sure any firewalls allow 5060 udp and tcp  and random ports above 
>>10000 in form your PBX.
>>
>>If you have all that it should work.
>>
>>AstGrp wrote:
>>
>> 
>>
>>    
>>
>>>Yes ....
>>>
>>>
>>>
>>>-----Original Message-----
>>>From: asterisk-users-admin at lists.digium.com
>>>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of James
>>>Sizemore Posted At: Thursday, March 11, 2004 10:47 AM
>>>Posted To: Asterisk User Group
>>>Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
>>>retries exceeded on call
>>>Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
>>>retries exceeded on call
>>>
>>>
>>>You do have :
>>>nat_enable: "1"
>>>nat_received_processing: "1"
>>>
>>>On the Ciscos?
>>>
>>>AstGrp wrote:
>>>
>>>
>>>
>>>   
>>>
>>>      
>>>
>>>>I am having a similar problem... I get the same message, but inbound 
>>>>calls can go through.... This is only Cisco phones that are behind
>>>>     
>>>>
>>>>        
>>>>
>>NAT.
>> 
>>
>>    
>>
>>>>  
>>>>
>>>>     
>>>>
>>>>        
>>>>
>>>   
>>>
>>>      
>>>
>>>>I have tried your recommendations from below, but still no luck.. 
>>>>User can make outbound calls, just can't receive any.  Any ideas 
>>>>would be greatly appreciated.. I even tried to change the timeout 
>>>>value in chan_sip, but it just waits longer to fail.. Just dosen't 
>>>>seem to want
>>>>     
>>>>
>>>>        
>>>>
>> 
>>
>>    
>>
>>>>to communicate...
>>>>
>>>>Thanks,
>>>>
>>>>gcc
>>>>
>>>>-----Original Message-----
>>>>From: asterisk-users-admin at lists.digium.com
>>>>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of John 
>>>>Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To:
>>>>     
>>>>
>>>>        
>>>>
>>Asterisk
>> 
>>
>>    
>>
>>>>  
>>>>
>>>>     
>>>>
>>>>        
>>>>
>>>   
>>>
>>>      
>>>
>>>>User Group
>>>>Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum 
>>>>retries exceeded on call
>>>>Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
>>>>retries exceeded on call
>>>>
>>>>
>>>>Are you using Cisco phones. ?
>>>>
>>>>I had this issue with my cisco phones. I didn't had any issues with 
>>>>dropped calls. All I did to fix this was set a prefered_codex and set
>>>>        
>>>>
>
>  
>
>>>>proxy_register to 0.
>>>>
>>>>I hope this helps.
>>>>
>>>>John Bittner
>>>>Simlab.net
>>>>
>>>>
>>>>
>>>>
>>>>  
>>>>
>>>>     
>>>>
>>>>        
>>>>
>>>>>-----Original Message-----
>>>>>From: asterisk-users-admin at lists.digium.com
>>>>>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of dkwok
>>>>>Sent: Wednesday, March 03, 2004 7:04 AM
>>>>>To: asterisk-users at lists.digium.com
>>>>>Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum 
>>>>>retries
>>>>>       
>>>>>
>>>>>          
>>>>>
>> 
>>
>>    
>>
>>>>>exceeded on call
>>>>>
>>>>>*CLI> Mar  3 12:55:05 WARNING[1150495040]: chan_sip.c:495
>>>>>retrans_pkt:
>>>>>Maximum retries exceeded on call 
>>>>>5946b8292887d22017623f85018dcfa4 at 192.168.1.143 for seqno 102
>>>>>       
>>>>>
>>>>>          
>>>>>
>>(Request)
>> 
>>
>>    
>>
>>>>>This has been brought up in the previous post but it does not seem 
>>>>>to have an answer for it so far.
>>>>>
>>>>>I cvs the stable v1.0 this morning after compiling and installing I 
>>>>>have calls drop 1 minutes into the connection with the above 
>>>>>message.
>>>>>
>>>>>If anyone has any idea of this occurrence.
>>>>>
>>>>>I have set up sip.conf:
>>>>>
>>>>>canreinvite=no
>>>>>
>>>>>--
>>>>>David Kwok
>>>>>Tel: 612 99292086 ext 1002
>>>>>Iaxtel/FWD # 17001813482 ext 1002
>>>>>
>>>>> 
>>>>>
>>>>>    
>>>>>
>>>>>       
>>>>>
>>>>>          
>>>>>
>>>>_______________________________________________
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>>>>
>>>>  
>>>>
>>>>     
>>>>
>>>>        
>>>>
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>>>
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>
>
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