RE: [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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 pgp0.pgp Description: signature
RE: [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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Varga Sent: Saturday, March 13, 2004 10:28 AM To: [EMAIL PROTECTED] 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [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 pgp0.pgp Description: signature
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 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
Re[2]: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Let me start by saying I have no cisco phones, and no idea how to configure them. I will speak to the use of asterisk behind a NAT'ing firewall, which I believe to be your setup. Asterisk is pretty picky about how SIP and RTP packets are handled by a NAT firewall. Basically you need to maintain the same udp port for incoming and outgoing udp packets. I will attempt to illustrate this from what I have seen with my OpenBSD firewall: The SIP UA sends a packet registration packet to the Asterisk server, giving a port number with which it will accept reply packets. Our NAT however is in the middle and does this. SIP UA Register my port: 5060 (or 2842) sent to Asterisk port 5060 (or 2842) Firewall, ah outgoing UDP connection, to Asterisk port 5060, I will choose a random high port and rewrite the packet with my IP and the UDP port SomeHighPort. Asterisk gets this and responds to the ORIGINAL port (5060) and the firewall expects this to arrive on the UDP port SomeHighPort. For some resion even if the port 5060 is forwarded to the SIP UA, this packet gets lost. So you need to tell your firewall to somehow use the same port as the UA for the re-written packets. On openbsd I use this directive: nat on $ext_if inet proto udp from any to any - ($ext_if) static-port (someone please comment on how to do this under linux for our other readers) which says do not rewrite UDP port numbers. This is also necessary for RTP to work properly. Some people have success with using the qualify= directive in sip.conf to keep the session alive in the firewall by sending packets before it times out. But I believe this to be a far better solution, as the path for the UDP packets always exists, alive or dead. And this works nice if you have the following setup, 1 Asterisk - NAT - (n) SIP UA Since SIP will setup our calls, and Asterisk will assign different RTP UDP ports for different calls to different SIP UA. -- Best regards, Scottmailto:[EMAIL PROTECTED] fwd: 253984 Friday, March 12, 2004, 11:34:51 AM, you wrote: A Ok.. Let me start by saying that SJPhone works fine through NAT and the A Cisco phones inside the internal network work fine also... It's just the A Cisco phones on the outside using NAT. A For Testing I opened the Firewall open on the IP for the * Server. I A have done, everything you recommended below, but still no go... When the A phone registers with port 2842? Not the standard 5060? Any ideas? I A believe this is where my problem sits... A Thanks, A -gcc A -Original Message- A From: [EMAIL PROTECTED] A [mailto:[EMAIL PROTECTED] On Behalf Of James A Sizemore A Posted At: Friday, March 12, 2004 9:03 AM A Posted To: Asterisk User Group A Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum A retries exceeded on call A Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum A retries exceeded on call A Make sure your using qualify=500 in the sip.conf along with nat=yes, A make sure any firewalls allow 5060 udp and tcp and random ports above A 1 in form your PBX. A If you have all that it should work. A AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 A 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: A 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI
RE: [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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post
RE: [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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [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 : 1230 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman
RE: [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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 [EMAIL PROTECTED] 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 smime.p7s Description: S/MIME Cryptographic Signature
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
