sr-users November 2006

sr-users@lists.kamailio.org

189 participants
306 discussions

by Daniel-Constantin Mierla

Hello, On 11/22/06 17:57, Kelvin Williams wrote: > > Yesterday, something happened—the exact I do not know just yet. > However, when looking at the logging that I have my OpenSER performing > I seen a ton of INVITES, CANCELs, OpenSER became unusable for a time > until I was able to block all SIP traffic at the firewall and let it > calm down enough to restart it. > > During this process calls could not be made (in or out). Is there a > way to prevent this type of behavior? It looked as though there were > only a handful of calls that caused a continuous loop. > that means that something is wrong with the configuration file or the sip request. The message was not properly handled and looped. If there is no need for loops in your config, you can have a check at the beginning of the script denying requests from local IP. If you need loops, then try to lower maxfwd value so it will loop less. However, the best is to detect which was the problem and fix it, and have the above fixes to prevent similar cases in future. If you faced a flood attack, the best is to look at pike module. Cheers, Daniel > What logic is recommended to test for a call that has been handled too > many times by the proxy, and to immediately drop it? > > Thanks in advance. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

17 years, 5 months

[Users] help in Call parking......

by raviprakash sunkara

Hello Users I'm Doing working on Both OpenSER and Asterisk ....... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=>115,1,Park() exten =>115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says " U parked on 701 extension " After When 9001 dial 701 ..... it Say " 483 too many parameters ... in X-lite " , Actual it has to ring 9003, -- Executing Park("SIP/9003-085d9e10", "") in new stack == Parked SIP/9003-085d9e10 on 701. Will timeout back to extension [from-sip] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/9003-085d9e10' == Spawn extension (from-sip, s, 1) exited KEEPALIVE on 'SIP/9003-085d9e10' Nov 22 18:18:45 NOTICE[3289]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.2.5 -- Stopped music on hold on SIP/9003-085d9e10 -- Registered extension context 'park-dial' -- Added extension 'SIP/9003' priority 1 to park-dial == Timeout for SIP/9003-085d9e10 parked on 701. Returning to park-dial,SIP/9003,1 -- Executing Dial("SIP/9003-085d9e10", "SIP/9003||t") in new stack -- Called 9003 -- Got SIP response 482 "Loop Detected" back from 192.168.2.76 -- Now forwarding SIP/9003-085d9e10 to 'Local/9003@from-sip' (thanks to SIP/9003-085df878) Nov 22 18:19:20 NOTICE[3949]: chan_local.c:498 local_alloc: No such extension/context 9003@from-sip creating local channel Nov 22 18:19:20 NOTICE[3949]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/9003@from-sip' (cause = 0) == Everyone is busy/congested at this time (1:0/0/1) Nov 22 18:19:30 WARNING[3949]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'park-dial' -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

17 years, 5 months

[Serusers] aaa

by Phan Van Duc

aaa

17 years, 5 months

[Users] resent packets in siptrace

by Mark Price

Hi, I am logging text/html messages with siptrace, and seeing that certain packets are being retransmitted several times, but am left with no way to detect that they are retransmissions and not legitimate messages. I.e. the contents of the msg field are the same, and the other fields in the table seem to be of no help. Is there a way to detect within openser that a packet is either a retransmission or being sent to multiple destinations, so that only the first one is recorded? The larger problem I am trying to solve is the above one. However, the specific one is interesting, too. The specific circumstances under which this is happening is that I'm trying to communicate between gaim 2.0 beta5 and a counterpath xten phone. 1. the gaim client will only login using tcp (it's udp transport seems to be broken.. it won't send the Authorization header). 2. messages sent from gaim arrive at the destination, but gaim does not see incoming messages... openser complains about tcp errors. 7(32169) ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s 7(32169) ERROR: tcpconn_connect: tcp_blocking_connect failed 7(32169) ERROR: tcp_send: connect failed 7(32169) msg_send: ERROR: tcp_send failed 7(32169) ERROR:tm:t_forward_nonack: sending request failed In this context, the MESSAGE packets get sent several times, and get logged several times. Thanks, Mark Price

17 years, 5 months

[Serusers] Problem with MTU size in SER with MIPv6.

by Jose Angel Calvo

Thank you for your help and sorry for my english. I have a problem with mipv6 and a server SIP, SER (SIP Express Router). The mipv6 path is "mipv6-2.0.2" in kernel 2.6.16.24-Ubuntu. The problem occurs when a packet is too big (it must be fragmented), and go to the node what makes function of P-CSCF (Proxy in SIP) and Correspondent Node. If the P-CSCF is not a CN, all is OK. But if the node is CN, then i have the problem. The SER logs are: ................ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: SER: new INVITE Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: parse_headers: flags=-1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: ERROR: warning_builder: buffer size exceeded Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING: warning skipped -- too big Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING:vqm_resize: resize(0) called Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: reply sent out. buf=0x8106270: SIP/2.0 1..., shmem=0xb5e8b330: SIP/2.0 1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: _reply_light: finished Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: mk_proxy: doing DNS lookup... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: DBG: tcp_main_loop: dead child 2 (shutting down?) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: child process 8051 exited by a signal 11 Nov 20 18:28:25 pcscfD kernel: [4296472.851000] <0>skb_under_panic: text:c02d2aad len:321 put:14 head:d62cfc00 data:d62cfbea tail:d62cfd2b end:d62cfd80 dev:eth0 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: core was not generated Nov 20 18:28:25 pcscfD kernel: [4297162.639000] ------------[ cut here ]------------ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: INFO: terminating due to SIGCHLD Nov 20 18:28:25 pcscfD kernel: [4297162.639000] kernel BUG at net/core/skbuff.c:112! Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8057]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8058]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8060]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8061]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8062]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8063]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8064]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8065]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8066]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8067]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8050]: INFO: signal 15 received Nov 20 18:28:25 pcscfD kernel: [4297162.639000] invalid opcode: 0000 [#3] Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: Memory status (pkg): ....................... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2055 fragments no.: 1, unused: 0 ^I^I bucket size: 1048576 - 2097152 (first 1572864) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2059 fragments no.: 1, unused: 0 ^I^I bucket size: 16777216 - 33554432 (first 31934160) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: TOTAL: 26 free fragments = 33537656 free bytes Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: ----------------------------- Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: shm_mem_destroy Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: destroying the shared memory lock Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: terminating due to SIGCHLD ############################################################## The problem is in the MTU size, because if the packet is not fragmented, there is not a problem. But the INVITE packet needs to fragment, what it means, i always have the same problem. My questions are: - Have anybody the same o similar problem? - The MTU size is a problem in SER? - Have anybody sent fragmented packets with SER? Thanks for your help. ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

17 years, 5 months

Re: [Users] $ct and $hdr(Content-Type) failing

by Daniel-Constantin Mierla

Hello, On 11/22/06 18:33, Mark Price wrote: > Daniel, > > I'm using openser 1.1.0. > Using $cT works, as well as $hdr(Content-Type). > I must have had a typo in my $hdr() command yesterday. > Thanks for the help. to make it more interactive and allow users to contribute/enhance to these documents, I moved to dokuwiki latest version so it can be edited by anybody. From time to time we may generate a static version to have it as reference in case someone tries to "poison" the content. http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.1.x http://www.openser.org/dokuwiki/doku.php/pseudovariables:devel Cheers, Daniel > > Mark Price > > > On 11/22/06, *Daniel-Constantin Mierla* <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>> wrote: > > Hello, > > I tried and seems to work. There was a typo in pseudo-variables > documentation (fixed now), the short name is $cT for Content-Type > -- $ct > stands for Contact header body. > > xlog("L_ALERT","**** $hdr(Content-Type) | $cT | $rb\n"); > > **** text/plain;charset=UTF-8 | text/plain;charset=UTF-8 | test > > I used devel version, what version are you using ? > > Cheers, > Daniel > > On 11/22/06 01:36, Mark Price wrote: > > Hi, > > > > I am trying to log MESSAGE packets differentiated the > content-type of > > the packet. > > I can see in the packet dump that they have types such as > "text/html" > > and so on. > > I'm using the following statement in the code: > > if(method=="MESSAGE") { > > xlog("L_ALERT","$ct $rb\n"); > > } > > The message body specified by $rb is printing out fine, but $ct is > > printing out "<null>". > > I get the exact same result if I say: > > xlog("L_ALERT","$hdr(Content-Type) $rb\n"); > > > > Is this a bug in openser? > > > > Thanks, > > Mark Price > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

17 years, 5 months

[Serusers] Problem with MTU size in SER with MIPv6.

by jose angel calvo

17 years, 5 months

[Serusers] NAT and ACK

by Benito Caracuel

Hello all , This is my scenario: I have SER with Private address behind a Cisco router which Has NAT implemented. I have two UAs with public address each one. UA1 ----------------------> Router Cisco with NAT (public IP) ------------------------> SER with private IP UA2 <----------------------Router Cisco with NAT (public IP) <------------------------ SER with private IP The problem is that when UA1 sends an INVITE with SDP, UA2 reply with 200 ok with SDP but after that, SER doesn't see the ACK. I know that UA1 sends the ACK but this ACK doesn't reach SER. I think that maybe the problem is NAT. Is possible to run SER with private address? How can i solve this? Help me please. Thank you very much. Ben. _________________________________________________________________ Consigue el nuevo Windows Live Messenger http://get.live.com/messenger/overview

17 years, 5 months

Re: [Serusers] Re: where is the vm.so for voice mail feature ?:

by lawrence k.y. lin

Hello list, I searched for the voice mail feature and found an example "Example 5-3. Example ser Config File" at the Admin's Guide in the following url: http://old.iptel.org/ser/doc/seruser/seruser.html#AEN1364 Inside that example, there is a statement to load vm.so as follows: loadmodule "modules/vm/vm.so" However, I cannot find vm.so under the ser-0.9.6. Please advise where I can find vm.so for voice mail feature. Thanks and Regards, Larry _________________________________________________________________ View Athletes Collections with Live Search http://sportmaps.live.com/index.html?source=hmemailtaglinenov06&FORM=MGAC01

17 years, 5 months

[Users] polycom phone proxy failover question

by Zahid Mehmood

Hi, I am having difficulty configuring polycom phone to failover when one proxy server is not available. I've configured the phone to use sip naptr records that provides two proxy server names. I've configured the "retry timeout" and "retry max count" however, when proxy1 is not available, polycom phone does not send requests to proxy2. Is there any other option I need to configure? I've tried this on polycom 430 and 601 models. Thanks is advance for your help. -- Zahid Mehmood CUIT Network Systems

17 years, 5 months

← Newer
1
...
7
8
9
10
11
12
13
...
31
Older →

Jump to page:

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users November 2006