[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
AstGrp
astgrp at cwkb.com
Fri Mar 12 21:58:34 MST 2004
Update...
I did some more testing today.. And with the same setup but one box
behind a Linksys router and another box behind a Pix firewall.. The
linksys works with no problems... So it appears to be how the PIX is
handling NAT & SIP... If any one has any thoughts on this , it would be
greatly appreciated.
And thank you James for the support you have given today.
Thanks,
gcc
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of AstGrp
Posted At: Friday, March 12, 2004 4:29 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
Do I need to associate the outside interface of the PIX with the phone
on the inside.. I don't remember doing this before...
Setup ----
* Server ---> PIX FW ---> WWW CLOUD ----> PIX FW ---> IP Phone
Again the only difference than before is the First PIX FW.... Old setup
was.... (Different server though)
* Server ----> Linksys Router ----> WWW CLOUD ----> PIX FW ----> IP
Phone
Any thoughts?
-----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 2:58 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
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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