[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries
exceeded on call
James Sizemore
james at deny.org
Fri Mar 12 09:50:15 MST 2004
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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