search for: 481

...ing like shown below. > May 18 14:57:15] WARNING[8314]: rtp.c:2433 rtp_socket: Unable to allocate RTP socket: Too many open files > [May 18 14:57:15] WARNING[8314]: chan_sip.c:6710 sip_alloc: Unable to create RTP audio session: Too many open files > [May 18 14:57:15] ERROR[8314]: acl.c:481 ast_ouraddrfor: Cannot create socket > [May 18 14:57:15] ERROR[8314]: acl.c:481 ast_ouraddrfor: Cannot create socket > [May 18 14:57:15] WARNING[8314]: rtp.c:2433 rtp_socket: Unable to allocate RTP socket: Too many open files > [May 18 14:57:15] WARNING[8314]: chan_sip.c:6710 sip_alloc: Un...

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=481 ------- Additional Comments From joseba.beltran@gmail.com 2006-05-31 19:09 MET ------- Many pptp nat helper users are suffering this problem: http://www.chillispot.org/forum/viewtopic.php?p=4118&highlight=pptp http://www.chillispot.org/forum/viewtopic.php?t=958&highlight=pptp http://...

...:28:49 compaq slapd[487]: conn=271 op=0 BIND dn="cn=admin,dc=login" method=128 Mar 11 15:28:49 compaq slapd[487]: conn=271 op=0 BIND dn="cn=admin,dc=login" mech=simple ssf=0 Mar 11 15:28:49 compaq slapd[487]: conn=271 op=0 RESULT tag=97 err=0 text= Mar 11 15:28:49 compaq slapd[481]: conn=271 op=1 SRCH base="dc=login" scope=2 filter="(&(objectClass=sambaDomain)(sambaDomainName=SKOLE1))" Mar 11 15:28:49 compaq slapd[481]: conn=271 op=1 SRCH attr=sambaDomainName sambaNextRid sambaNextUserRid sambaNextGroupRid sambaSID sambaAlgorithmicRidBase objectCla...

Processing commands for control at bugs.debian.org: > reassign 469811 linux-2.6 Bug#469811: xen-hypervisor-3.0.3-1-i386-pae: kernel BUG at drivers/xen/core/evtchn.c:481 Bug reassigned from package `xen-hypervisor-3.0.3-1-i386-pae' to `linux-2.6'. > forcemerge 410807 469811 Bug#410807: linux-2.6/etch xen support - kernel BUG at drivers/xen/core/evtchn.c:481 Bug#469811: xen-hypervisor-3.0.3-1-i386-pae: kernel BUG at drivers/xen/core/evtchn.c:481 Bug#4139...

http://bugzilla.mindrot.org/show_bug.cgi?id=481 ------- Additional Comments From markus at openbsd.org 2003-02-02 00:11 ------- do you have a patch for the manpage? ------- You are receiving this mail because: ------- You are the assignee for the bug, or are watching the assignee.

I am sending this problem for 2nd time. Please help. Thanks _____ From: Omar McKenzie [mailto:omckenzie@trenetinc.com] Sent: Thursday, September 08, 2005 9:57 AM To: 'asterisk-users@lists.digium.com' Subject: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist" I am not able to get softphone registered (active) with * . new installation , new user Able to get server started , and phone appears to register . gets the SIP reponse 481 message Register SIP '4009' at 192.168.200.10 p...

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=481 ddehaan@omnexcontrols.com changed: What |Removed |Added ---------------------------------------------------------------------------- Status|RESOLVED |REOPENED Resolution|FIXED | ------- Additiona...

...|SIP Status | | |(5060) ------------------> (5061) | |907.590261| | INVITE SDP (g711U te |SIP Request | | |(5060) ------------------> (5061) | |907.591294| | 481 Call/Transaction |SIP Status | | |(5060) <------------------ (5061) | |907.591420| | ACK | |SIP Request | | |(5060) ------------------> (5061) | |907.591467|...

...152 (1 at 1): After 900s suddenly *asterisk* starts the timer reinvite - I would have expected the reinvite started by the provider as usual. The expected reinvite by the provider is started during authentication of the reinvite started by asterisk and is answered immediately by asterisk with sip 481. The answer of the provider after the resend of the reinvite came about 0.5s later and is sip 481, too. => The session obviously isn't known on both sides! Asterisk therefore now drops the call (bye). Does anybody has any idea about the reason why both members don't recognize the ex...

I am not able to get softphone registered (active) with * . new installation , new user Able to get server started , and phone appears to register . gets the SIP reponse 481 message Register SIP '4009' at 192.168.200.10 port 2199 expires 120 Unregistered SIP '4009' Register SIP '4009' at 192.168.200.10 port 9428 expires 120 Saved useragent "RTC/1.24949 " for peer 4009 Got SIP response 481 "Call Leg/Transactio...

Hi all, Anyone know how to make asterisk properly reply to options keep-alive? Or just force a 200 OK somehow? I recently took over a server and they have ~80 pap2 devices that send nat keep-alive and * always replies with 481 No subscription. It's more of an annoyance, I know but I like to keep my pcap's clean. S. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110513/fa335093/attachment.htm>

...ough we get the same behaviour using other phones (X-lite, Kphone). It behaves the same regardless of whether the incoming call is from a SIP extension or an external PSTN call through the CAPI Channel. The SIP debug (see below) shows that the response to the SIP "Cancel" request is "481 no such call". The call-ID seems to match the corresponding invite. I suspect it has something to do with the NAT setup:- phone1__ | |--NAT---internet---NAT---phone2 asterisk__| Calls to phone1 work fine but to phone2 I get the 481 response to a cancel. The S...

...several months, and I recently set it up again to work with Asterisk. Everything works fine for several minutes -- I am able to receive and make calls as expected. However, after a few minutes, and every few minutes thereafter, I get the following message on the console: -- Got SIP response 481 "Call Does Not Exist" back from 192.168.1.98 After this message, the phone no longer works properly -- if at all. When attempting to dial, the phone seems to ignore my dial plan, accepting no more than five digits (I get the tone burst indicating that it is done accepting digits), and i...

...t Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday, July 01, 2005 4:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Got SIP response 481 "Invalid CSeq Number" backfrom X.X.X.X I get this error message when sending calls to a Cisco Gateway AS 5300, one call out of 10. Is there any configuration hack either on Asterisk or the Cisco that would this problem go away?? Federico Alves __________________________________________...

One of our remote user's phone reports frequently: Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from <IP> What can I do ??? bye Ronald

...eneral section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004

...t if I call from the Polycom to my mobile and then I hang up the mobile the Polycom thinks the call is still active. However doing a show sip channels shows the the call has ended. Further to that doing a tcpdump shows that Asterisk sends a SIP BYE to the phone but the phone responds with: Status 481 Call Leg/Transaction does not exist. The Polycom is currently associated with 2 sip servers (using 2 lines on the phone) because I am currently in the progress of migrating from one server to another. So the asterisk server is having issues with is on Line 2 and it works perfectly well on Line 1...

http://bugzilla.mindrot.org/show_bug.cgi?id=481 Summary: Extend man page with documentation Product: Portable OpenSSH Version: older versions Platform: All URL: http://www.openbsd.org/cgi-bin/man.cgi?query=ssh-agent OS/Version: All Status: NEW Severity: norm...

-- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207 == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1' I am getting this error on an outgoing call to a SIP host. The call just disconnects .. is there any way around it ? thanks Dave

I have a PCPHONELINE SIS phone set it up to asterisk Registered SIP '205' at 24.172.221.22 port 2770 expires 120 (Port changes every time) Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 24.172.221.22(24.172.221.22 is my phones IP) Anyone have an idea what the problem is? Jeff