[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
AstGrp
astgrp at cwkb.com
Thu Mar 11 08:59:41 MST 2004
Here's a copy of the cisco config....
------ Current *FLASH* Configuration ------
Platform : Cisco IP Phone 7940
Elasped Time: 00:01:37
dhcp_server : 10.100.0.2
my_ip_addr : 10.100.0.150
subnet_mask : 255.255.255.0
defaultgw : 10.100.0.2
dyn_dns_addr_1 : 0.0.0.0
dyn_dns_addr_2 : 0.0.0.0
dns_addr : 10.100.254.7
dns_backup_1: 24.93.68.65
tftp_addr : 66.64.246.36
dyn_tftp_addr : 0.0.0.0
my_mac_addr : 000f:23ac:4559
domain_name : tnessentials.com
my_name : SIP000F23AC4559
Status Flags : 12300000
image_version : "P0S3-06-2-00"
FirmLoadID : "PC030301"
DSPLoadID : "PS03AT38"
network_media_type : Auto
network_port2_type : Hub/Switch
tos_media : 5
phone_label : "TNE PBX VOIP"
tftp_cfg_dir : ""
phone_password : **********
phone_prompt : "SIP Phone"
language : english
sntp_mode : DirectedBroadcast
sntp_server :
time_zone : EST
dst_offset : 1
dst_start_month : April
dst_start_day : 0
dst_start_day_of_week : Sun
dst_start_week_of_month : 1
dst_start_time : 02
dst_stop_month : Oct
dst_stop_day : 0
dst_stop_day_of_week : Sunday
dst_stop_week_of_month : 8
dst_stop_time : 2
dst_auto_adjust : 1
time_format_24hr : 1
date_format : M/D/Y
nat_enable : 1
nat_address :
voip_control_port : 5060
start_media_port : 16456
end_media_port : 17456
sync : "1"
xml_card_dir : ""
xml_card_file : "CARD.XML"
telnet_level : 2
services_url : ""
directory_url : ""
logo_url : ""
http_proxy_addr :
http_proxy_port : 80
enable_vad : 0
dial_template : "dialplan"
callerid_blocking : 0
anonymous_call_block : 0
autocomplete : 1
messages_uri : "55"
dnd_control : 0
preferred_codec : g711ulaw
dtmf_outofband : avt
dtmf_avt_payload : 101
dtmf_db_level : 3
dtmf_inband : 1
line1_name : "khome"
line2_name : "UNPROVISIONED"
line1_authname : "khome"
line2_authname : "UNPROVISIONED"
line1_password : **********
line2_password : **********
line1_shortname : "UNPROVISIONED"
line2_shortname : "UNPROVISIONED"
line1_displayname : "Kyle Elworthy"
line2_displayname : ""
proxy1_address : "66.64.246.36"
proxy2_address : ""
proxy1_port : 5060
proxy2_port : 5060
sip_retx : 10
sip_invite_retx : 6
timer_t1 : 500
timer_t2 : 4000
timer_invite_expires : 180
timer_register_expires : 3600
proxy_register : 1
proxy_backup : ""
proxy_emergency : ""
proxy_backup_port : 5060
proxy_emergency_port : 5060
outbound_proxy :
outbound_proxy_port : 5060
nat_received_processing : 1
mwi_status : 0
call_waiting : 1
user_info : none
cnf_join_enable : 1
remote_party_id : 0
semi_attended_transfer : 1
call_hold_ringback : 0
stutter_msg_waiting : 0
cfwd_url : ""
call_stats : 1
auto_answer : 0
local_cfwd_enable : 1
timer_register_delta : 5
-----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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