John Bittner
2004-Mar-02 21:45 UTC
[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@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of dkwok > Sent: Wednesday, March 03, 2004 7:04 AM > To: asterisk-users@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@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 >
dkwok
2004-Mar-03 05:03 UTC
[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@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 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040302/27a7088b/smime.bin
AstGrp
2004-Mar-10 21:06 UTC
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
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@lists.digium.com [mailto:asterisk-users-admin@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@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of dkwok > Sent: Wednesday, March 03, 2004 7:04 AM > To: asterisk-users@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@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 >_______________________________________________ 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
AstGrp
2004-Mar-11 08:56 UTC
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Yes .... -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@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@lists.digium.com >[mailto:asterisk-users-admin@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@lists.digium.com >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of dkwok >>Sent: Wednesday, March 03, 2004 7:04 AM >>To: asterisk-users@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@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 >> >> >> > >_______________________________________________ >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 > >_______________________________________________ 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
AstGrp
2004-Mar-11 08:59 UTC
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
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@lists.digium.com [mailto:asterisk-users-admin@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@lists.digium.com >[mailto:asterisk-users-admin@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@lists.digium.com >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of dkwok >>Sent: Wednesday, March 03, 2004 7:04 AM >>To: asterisk-users@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@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 >> >> >> > >_______________________________________________ >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 > >_______________________________________________ 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
AstGrp
2004-Mar-12 09:34 UTC
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
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@lists.digium.com [mailto:asterisk-users-admin@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@lists.digium.com >[mailto:asterisk-users-admin@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 behindNAT.>> >> > > > >>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@lists.digium.com >>[mailto:asterisk-users-admin@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@lists.digium.com >>>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of dkwok >>>Sent: Wednesday, March 03, 2004 7:04 AM >>>To: asterisk-users@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@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 >>> >>> >>> >>> >>> >>_______________________________________________ >>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 >> >> >> >> > > >_______________________________________________ >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 > >_______________________________________________ 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
AstGrp
2004-Mar-12 10:45 UTC
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
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@lists.digium.com [mailto:asterisk-users-admin@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@lists.digium.com >[mailto:asterisk-users-admin@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@lists.digium.com >>[mailto:asterisk-users-admin@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@lists.digium.com >>>[mailto:asterisk-users-admin@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@lists.digium.com >>>>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of dkwok >>>>Sent: Wednesday, March 03, 2004 7:04 AM >>>>To: asterisk-users@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@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 >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>_______________________________________________ >>>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 >>> >>> >>> >>> >>> >>> >>_______________________________________________ >>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 >> >> >> >> > > >_______________________________________________ >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 > >_______________________________________________ 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
AstGrp
2004-Mar-12 14:28 UTC
[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@lists.digium.com [mailto:asterisk-users-admin@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@lists.digium.com >[mailto:asterisk-users-admin@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? Anyideas?>> >> > > > >>I believe this is where my problem sits... >> >>Thanks, >> >>-gcc >> >> >>-----Original Message----- >>From: asterisk-users-admin@lists.digium.com >>[mailto:asterisk-users-admin@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@lists.digium.com >>>[mailto:asterisk-users-admin@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@lists.digium.com >>>>[mailto:asterisk-users-admin@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 andset>>>> >>>> > > > >>>>proxy_register to 0. >>>> >>>>I hope this helps. >>>> >>>>John Bittner >>>>Simlab.net >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>>-----Original Message----- >>>>>From: asterisk-users-admin@lists.digium.com >>>>>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of dkwok >>>>>Sent: Wednesday, March 03, 2004 7:04 AM >>>>>To: asterisk-users@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@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 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>_______________________________________________ >>>>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 >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>_______________________________________________ >>>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 >>> >>> >>> >>> >>> >>> >>_______________________________________________ >>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 >> >> >> >> > > >_______________________________________________ >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 > >_______________________________________________ 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
AstGrp
2004-Mar-12 21:58 UTC
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
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@lists.digium.com [mailto:asterisk-users-admin@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@lists.digium.com [mailto:asterisk-users-admin@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@lists.digium.com >[mailto:asterisk-users-admin@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? Anyideas?>> >> > > > >>I believe this is where my problem sits... >> >>Thanks, >> >>-gcc >> >> >>-----Original Message----- >>From: asterisk-users-admin@lists.digium.com >>[mailto:asterisk-users-admin@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@lists.digium.com >>>[mailto:asterisk-users-admin@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@lists.digium.com >>>>[mailto:asterisk-users-admin@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 andset>>>> >>>> > > > >>>>proxy_register to 0. >>>> >>>>I hope this helps. >>>> >>>>John Bittner >>>>Simlab.net >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>>-----Original Message----- >>>>>From: asterisk-users-admin@lists.digium.com >>>>>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of dkwok >>>>>Sent: Wednesday, March 03, 2004 7:04 AM >>>>>To: asterisk-users@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@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 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>_______________________________________________ >>>>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 >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>_______________________________________________ >>>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 >>> >>> >>> >>> >>> >>> >>_______________________________________________ >>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 >> >> >> >> > > >_______________________________________________ >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 > >_______________________________________________ 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
AstGrp
2004-Mar-13 13:21 UTC
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Thank you... I found that document last night.. And I have the pix configured this way with fixup sip... But still no go.. I am going to try and upgrade the pix tonight and see if that helps. gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Stephen Varga Sent: Saturday, March 13, 2004 10:28 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call On Friday 12 March 2004 09:28 pm, AstGrp wrote:> 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?You may want to look at this page from Cisco http://www.cisco.com/en/US/products/hw/vpndevc/ps2030/ products_configuration_example09186a00801fc74a.shtml It looks like it will take care of the PAT/NATing issues. I have not have the luxury of trying it. HTH, Steve
Stephen Varga
2004-Mar-13 14:20 UTC
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
On Saturday 13 March 2004 08:21 pm, AstGrp wrote:> Thank you... I found that document last night.. And I have the pix > configured this way with fixup sip... But still no go.. I am going to > try and upgrade the pix tonight and see if that helps. >The only suggestion I have now is to start doing network sniffs before and after the PIX to see what is actually happening on the wire. Hopefully it will give you clue to missing part of the your puzzle. Steve -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040313/bf09862c/attachment.pgp
AstGrp
2004-Mar-16 17:54 UTC
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Ok.. After upgrading the PIX to version PIX6.3(3). I can register the phone, but I am having related issue of sorts... Here's the low down.. The outside interface of the PIX is doing PAT. And I have one to one NAT translation for the * Server... But if I configure everything this way... I get an Unreachable... But if I put the PAT IP in for the NAT IP in the SIP file, it registers fine, but then no sound is heard through the phone.... Any ideas.. gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of James Sizemore Posted At: Monday, March 15, 2004 5:24 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 Some firewalls when doing nat will alter the return address (need to make nat work) but not recalculate the header checksum, (Sonic walls come to mind.), Linux will proply delete any tcp/udp packet that fails its checksum at the kernel level, and send an error to the app. If this is happening to you Asterisk should log some kind of error. AstGrp wrote:>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@lists.digium.com >[mailto:asterisk-users-admin@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@lists.digium.com >[mailto:asterisk-users-admin@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: > > >_______________________________________________ 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