Whenever I see the "Maximum retries" message it usually indicated a communication problem, like one way traffic. Last time I got it, I traced it to a bad firewall rule, dropped the firewall and it worked, the time before that when I received it, it was due to a routing error, the server could get the request but couldn't respond (wrong gateway). Rich Adamson wrote:>Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09 > >When call comes in and is sent to a Cisco 7960, I see: > > -- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new stack > -- Called 3000 >Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries > exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for seqno 102 > (Critical Request) > == No one is available to answer at this time > -- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new stack > -- Playing 'voicemail/default/3000/greet' (language 'en') > -- Playing 'vm-isunavail' (language 'en') > >but the phone doesn't ring. The 7960 is registered and can place >outbound calls. Same with multiple 7960's. > >Did I miss a mandatory config change, or is sip broken? > >Rich > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >. > > >
Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09
When call comes in and is sent to a Cisco 7960, I see:
-- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new
stack
-- Called 3000
Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries
exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for seqno 102
(Critical Request)
== No one is available to answer at this time
-- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new
stack
-- Playing 'voicemail/default/3000/greet' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
but the phone doesn't ring. The 7960 is registered and can place
outbound calls. Same with multiple 7960's.
Did I miss a mandatory config change, or is sip broken?
Rich
Hi All,
Looking for a recommendation. I was hoping to purchase a * "KIT" for a
small office. I have 4 lines and 4 extensions need phones so I need 4
phones. What phones would many of you recommend? Can you refer me to any
companies that have built a kit I can plugin and configure?
Thanks,
Jerry Roy
RemoteHand, Inc.
562-305-9545
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich
Adamson
Sent: Friday, August 27, 2004 7:15 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] sip change?
Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09
When call comes in and is sent to a Cisco 7960, I see:
-- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new
stack
-- Called 3000
Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum
retries
exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for
seqno 102
(Critical Request)
== No one is available to answer at this time
-- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new
stack
-- Playing 'voicemail/default/3000/greet' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
but the phone doesn't ring. The 7960 is registered and can place
outbound calls. Same with multiple 7960's.
Did I miss a mandatory config change, or is sip broken?
Rich
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Jerry,
If your talking sip phones...
I am using the Cisco 7960 phones and love them. The quality and
stability against Asterisk have been excellent.
Chad Brown - IdentityMine
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jerry Roy
Sent: Friday, August 27, 2004 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] sip change?
Hi All,
Looking for a recommendation. I was hoping to purchase a * "KIT" for a
small office. I have 4 lines and 4 extensions need phones so I need 4
phones. What phones would many of you recommend? Can you refer me to any
companies that have built a kit I can plugin and configure?
Thanks,
Jerry Roy
RemoteHand, Inc.
562-305-9545
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich
Adamson
Sent: Friday, August 27, 2004 7:15 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] sip change?
Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09
When call comes in and is sent to a Cisco 7960, I see:
-- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new
stack
-- Called 3000
Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum
retries
exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for
seqno 102
(Critical Request)
== No one is available to answer at this time
-- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new
stack
-- Playing 'voicemail/default/3000/greet' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
but the phone doesn't ring. The 7960 is registered and can place
outbound calls. Same with multiple 7960's.
Did I miss a mandatory config change, or is sip broken?
Rich
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users