Re: [OpenSIPS-Users] No RADIUS traffic
Hi, This is rather strange. Are there any knows bugs for the libradiusclient package for the platform your are using? I would try to recompile that package. I remember there was a broken package in debian a couple of month ago, maybe on other platforms too. If not I have no further ideas anymore. Good luck BR Uwe Leon Li schrieb: Uwe, Strace output, nothing comes when I tries to register an endpoint. [...@cbr-a-sysdev1 lli]$ sudo /usr/local/bin/strace -f -e open /usr/local/sbin/opensips open(/etc/ld.so.preload, O_RDONLY)= -1 ENOENT (No such file or directory) open(/etc/ld.so.cache, O_RDONLY) = 3 open(/lib/libdl.so.2, O_RDONLY) = 3 open(/lib/libresolv.so.2, O_RDONLY) = 3 open(/lib/tls/libc.so.6, O_RDONLY)= 3 open(/usr/local/etc/opensips/opensips.cfg, O_RDONLY) = 3 open(/dev/urandom, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/signaling.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/sl.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/tm.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/rr.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/maxfwd.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/usrloc.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/registrar.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/textops.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/mi_fifo.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/uri_db.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/uri.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/xlog.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/acc.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/auth.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/auth_radius.so, O_RDONLY) = 4 open(/etc/ld.so.cache, O_RDONLY) = 4 open(/usr/local/lib/libradiusclient-ng.so.2, O_RDONLY) = 4 open(/lib/libcrypt.so.1, O_RDONLY)= 4 open(/lib/libnsl.so.1, O_RDONLY) = 4 open(/etc/resolv.conf, O_RDONLY) = 4 open(/etc/nsswitch.conf, O_RDONLY)= 4 open(/etc/ld.so.cache, O_RDONLY) = 4 open(/lib/libnss_files.so.2, O_RDONLY) = 4 open(/etc/host.conf, O_RDONLY)= 4 open(/etc/hosts, O_RDONLY)= 4 open(/etc/hosts, O_RDONLY)= 4 Listening on udp: 202.158.197.134 [202.158.197.134]:5060 tcp: 202.158.197.134 [202.158.197.134]:5060 Aliases: tcp: cbr-a-sysdev1:5060 tcp: cbr-a-sysdev1.aarnet.net.au:5060 udp: cbr-a-sysdev1:5060 udp: cbr-a-sysdev1.aarnet.net.au:5060 open(/etc/localtime, O_RDONLY)= 4 open(/dev/zero, O_RDWR) = 5 open(/usr/local/etc/radiusclient-ng/radiusclient.conf, O_RDONLY) = 6 open(/usr/local/etc/radiusclient-ng/dictionary, O_RDONLY) = 6 Process 25956 attached Process 25957 attached Process 25958 attached [pid 25958] open(/tmp/opensips_fifo, O_RDONLY|O_NONBLOCK) = 9 [pid 25958] open(/tmp/opensips_fifo, O_WRONLY|O_NONBLOCK) = 11 Ldd auth_radiu.so: [...@cbr-a-sysdev1 lli]$ /usr/bin/ldd /usr/local/lib/opensips/modules/auth_radius.so libradiusclient-ng.so.2 = /usr/local/lib/libradiusclient-ng.so.2 (0x00ee5000) libc.so.6 = /lib/tls/libc.so.6 (0x00111000) libcrypt.so.1 = /lib/libcrypt.so.1 (0x00eb) libnsl.so.1 = /lib/libnsl.so.1 (0x0064c000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x0067b000) Thanks, Leon -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: Friday, 26 June 2009 5:26 PM To: Leon Li Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] No RADIUS traffic Leon, Could you post the output of the strace call? And could you please post the output of ldd auth_radius.so ? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS-mediaproxy, Asterisk (packet2packet bridge) - problem with RTP
Hi, I have implemented the following scenario: [incoming pstn]---[opensips]--[asterisk] ---[sip phone] | [outgoing pstn]---[opensips]--| Opensips acts as SBC with mediaproxy functionality. Moreover, I use the LCR module to route calls. The Asterisk is located at the public domain and we have activated the packet2packet bridge. A soft phone is registered to asterisk and we have created a ring group that sends an incoming call to soft phone and external line (outbound pstn) that rings simultaneous both devices. Opesips version 1.4.5 or 1.5 Asterisk version 1.6 Single calls without ring gourp: Incoming calls from PSTN to asterisk through Opensips with mediaproxy enabled. It works properly. Outgoing calls from Asterisk to PSTN through Opensips with mediaproxy enabled. It works properly. Calls with ring group enabled: Incoming call from PSTN to asterisk through opensips with mediaproxy enabled. The incoming call activate the asterisk's ring group and sends the call to sip phone and external line – outgoing pstn call. Both devices ring simultaneous. When hang-up: A) soft phone, the signaling and media work properly. B) External line, the signaling works properly but the media is not open. The system (opensips/mediaproxy) generates two media sessions(incoming and outgoing) but the ip of asterisk at both sessions has value Unknown. The mediaproxy/opensips tries to connect the two legs through asterisk. But this does not work because the asterisk acts as packet2packet bridge. Please, can you provide any help/sugestion about this problem? Regards, Dimitris ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Xlite+Opensips
Hi there Pls check this before trying to capture 127.0.0.1 On Fri, Jun 26, 2009 at 6:51 AM, nOeLiA noep...@gmail.com wrote: Hi! Yes I have both in my laptop. I tried configurying the 127.0.0.1 but it did nit run... I will try your other option and I will come back to you! Many many thanks On Fri, Jun 26, 2009 at 4:56 AM, Cao Lei-MNW784 lei@motorola.comwrote: Hi Noelia, You should put the IP to the external IP, even 192.168.X.X would work, but not 127.0.0.1. Also, did you put the correct port? Thanks Regards Cao, Charles -- *From:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *On Behalf Of *nOeLiA *Sent:* Thursday, June 25, 2009 10:52 PM *To:* users@lists.opensips.org *Subject:* [OpenSIPS-Users] Xlite+Opensips Hi! I have installed Xlite in my laptop to ensure that my opensips is working. I have also installed wireshark to sniff the network and being able to check if the Xlite is communicating to the Opensips. When I start Opensips I see it is listening in IP:127.0.0.1, so I configureg the XLite with SIP/Proxy address:127.0.0.1 but I cannot see any communication between them. Can anybody help me please? Thanks! Noelia ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Xlite+Opensips
Sorry Dangvinh, what I have to try? Many thanks for your help!!:) On Mon, Jun 29, 2009 at 8:33 AM, DangVinh Nguyen dangvinh.ngu...@gmail.comwrote: Hi there Pls check this before trying to capture 127.0.0.1 On Fri, Jun 26, 2009 at 6:51 AM, nOeLiA noep...@gmail.com wrote: Hi! Yes I have both in my laptop. I tried configurying the 127.0.0.1 but it did nit run... I will try your other option and I will come back to you! Many many thanks On Fri, Jun 26, 2009 at 4:56 AM, Cao Lei-MNW784 lei@motorola.comwrote: Hi Noelia, You should put the IP to the external IP, even 192.168.X.X would work, but not 127.0.0.1. Also, did you put the correct port? Thanks Regards Cao, Charles -- *From:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *On Behalf Of *nOeLiA *Sent:* Thursday, June 25, 2009 10:52 PM *To:* users@lists.opensips.org *Subject:* [OpenSIPS-Users] Xlite+Opensips Hi! I have installed Xlite in my laptop to ensure that my opensips is working. I have also installed wireshark to sniff the network and being able to check if the Xlite is communicating to the Opensips. When I start Opensips I see it is listening in IP:127.0.0.1, so I configureg the XLite with SIP/Proxy address:127.0.0.1 but I cannot see any communication between them. Can anybody help me please? Thanks! Noelia ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] [NEW] REGISTAR module enhancements
Hi,
There were couple of changes pushed to the REGISTRAR module to allow
more flexibility in configuring the behaviour of the module. More or
less, these changes aimed to move most of the global options (per module
options) to per AOR or function options. So, you can configure the
behaviour per AOR (like how many contacts are allowed, if branches
should be appended, if PATH support should be used and how, etc).
A brief summary of the changes:
1. Lookup() function accepts a set of char flags:
'm' - enable/disable method filtering
'b' - disable/enable the appending of branches
See:
http://www.opensips.org/html/docs/modules/devel/registrar.html#id271073
Ex:
lookup(location);
lookup(location,b);
lookup(location,,$var(my_aor));
2. Save() function - the binary flags (for mem only saving and no reply
options) were moved as char flags ('m' and 'r'). More char flags were added:
'cnn' - maximum number of contacts for the AOR (like 'c10' , 'c2' ,
'c0') - 0 disables the check
's' - socket header (replacement for the sock_flag global parameter)
'p0' , 'p1' , 'p2', 'v' - controls the path mode
See:
http://www.opensips.org/html/docs/modules/devel/registrar.html#id228571
Ex:
save(location);
save(location,c5mp0);
save(location,m, $var(my_aor));
3. Global module param removed:
sock_flag - moved as param of save()
use_path, path_mode, path_use_received - moved as param for save()
method_filtering - moved as param for lookup()
append_branches - moved as param for lookup()
Regards,
Bogdan
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] LDAP Authentication
Hi Alan, I'm not an LDAP expert to get into details about how ldap should be configured or soWhat I can tell is that the bind is static (only once done at the beginning at that's it)Can you send me a link or something to read more about what this dynamic bind means in LDAP ? Thanks and regards, Bogdan Alan Rubin wrote: Bogdan, Apparently the email administrator had a regex on the SMTP gateway to reject messages with pass (and) word (combined) because of previous users succumbing to phishing exercises. It may work now, but I will continue to check the archives. Oh well. Regarding: Now, going to the actual issue, the problem is related to password - about how the client and server (ldap) are keeping the password - do they both keep it same format (like plain text) ? Regards, Bogdan I think I've figured out the issue, although I don't believe there is a solution. Hopefully you can verify, either way. The bind user in the ldap.cfg file does not have the privilege to retrieve the pass word field from our LDAP directory. The only way our LDAP setup is supposed to work is by binding using the user-to-be-authenticated directly with the LDAP directory server. It is my understanding, and this is where you can verify or correct me, that opensips and the LDAP module can not change the bind user dynamically. Regards, Alan Rubin -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Alan Rubin Sent: Wednesday, 24 June 2009 8:10 AM To: Bogdan-Andrei Iancu Cc: users@lists.opensips.org Subject: [OpenSIPS-Users] LDAP Authentication Bogdan, The LDAP messages from the mailing list are still not reaching my mailbox, which is unusual. I am checking the mail services on my end. Still managed to pick up your last message from the Archive. After making the changes suggested for my config file, I'm still failing with a 401 - Unauthorized. Here are the relevant logs: Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:ldap:lds_search: [sipaccounts]: performing LDAP search: dn [o=ntg], scope [2], filter [((cn=oh5)(departmentNumber=66)(ntguserstatus=Active))], client_timeout [500] usecs Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:ldap:ldap_params_search: [sipaccounts]: [1] LDAP entries found Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:auth:check_nonce: comparing [4a4155840004dcd97551d7189591cf32402f006987b9] and [4a4155840004dcd97551d7189591cf32402f006987b9] Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:auth:reserve_nonce_index: second= 9, sec_monit= -1, index= 5 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:auth:build_auth_hf: nonce index= 5 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:auth:build_auth_hf: 'WWW-Authenticate: Digest realm=155.205.69.126, nonce=4a415584000573fd091deb999f17423ea6b4be4cb6e2 ' Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:core:parse_headers: flags= Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:core:check_via_address: params 155.205.26.124, 155.205.26.124, 0 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:core:destroy_avp_list: destroying list (nil) Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30653]: DBG:core:receive_msg: cleaning up dcshub1:/usr/local/opensips/etc/opensips # dcshub1:/usr/local/opensips/etc/opensips # dcshub1:/usr/local/opensips/etc/opensips # grep 07:51:26 /var/log/localmessages | less dcshub1:/usr/local/opensips/etc/opensips # dcshub1:/usr/local/opensips/etc/opensips # grep 07:51:26 /var/log/localmessages Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_msg: SIP Request: Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_msg: method: REGISTER Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_msg: uri: sip:155.205.69.126 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_msg: version: SIP/2.0 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_headers: flags=2 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_via_param: found param type 232, branch = z9hG4bK-d8754z-02350078246c1c6a-1---d8754z-; state=6 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_via_param: found param type 235, rport = n/a; state=17 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_via: end of header reached, state=5 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_headers: via found, flags=2 Jun 24 07:51:26 dcshub1 /usr/local/opensips/sbin/opensips[30646]: DBG:core:parse_headers: this
[OpenSIPS-Users] Update = dlg_handlers vs. dlg_db_handler DIALOG not deleted on BYE
Hi, If I use db_mode=1 the dialog is deleted from the database (as expected) but not from the memory. I will test with a fresh installation and maybe open a bug report. BR Uwe Uwe Kastens schrieb: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. Unfort I was not able to find out what the states and the events means. So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [NEW] REGISTAR module enhancements
Hi Brett,
yes, one by one, I will replace the binary flags (for functions) with
char flags - as you said, it is more user friendly :)...
Regards,
Bogdan
Brett Nemeroff wrote:
Is this move to char flags going to be global? It's nice to see the
config syntax becoming a little more friendly. Are we losing anything
in performance that way?
Thanks for all the great work!!
-Brett
On Mon, Jun 29, 2009 at 7:39 AM, Bogdan-Andrei Iancu
bog...@voice-system.ro mailto:bog...@voice-system.ro wrote:
Hi,
There were couple of changes pushed to the REGISTRAR module to allow
more flexibility in configuring the behaviour of the module. More or
less, these changes aimed to move most of the global options (per
module
options) to per AOR or function options. So, you can configure the
behaviour per AOR (like how many contacts are allowed, if branches
should be appended, if PATH support should be used and how, etc).
A brief summary of the changes:
1. Lookup() function accepts a set of char flags:
'm' - enable/disable method filtering
'b' - disable/enable the appending of branches
See:
http://www.opensips.org/html/docs/modules/devel/registrar.html#id271073
Ex:
lookup(location);
lookup(location,b);
lookup(location,,$var(my_aor));
2. Save() function - the binary flags (for mem only saving and no
reply
options) were moved as char flags ('m' and 'r'). More char flags
were added:
'cnn' - maximum number of contacts for the AOR (like 'c10' , 'c2' ,
'c0') - 0 disables the check
's' - socket header (replacement for the sock_flag global
parameter)
'p0' , 'p1' , 'p2', 'v' - controls the path mode
See:
http://www.opensips.org/html/docs/modules/devel/registrar.html#id228571
Ex:
save(location);
save(location,c5mp0);
save(location,m, $var(my_aor));
3. Global module param removed:
sock_flag - moved as param of save()
use_path, path_mode, path_use_received - moved as param for save()
method_filtering - moved as param for lookup()
append_branches - moved as param for lookup()
Regards,
Bogdan
___
Users mailing list
Users@lists.opensips.org mailto:Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Feature request: Loop detection
Hi Brett, This will be kind of pike but instead of using as input the source IP string, it should use a custom string you build form script, right ? this string will be a kind of key (logical one) to identify the loop. Regards, Bogdan Brett Nemeroff wrote: Hey All, I was wanting to submit a feature request for loop detection. Specifically NOT SIP loop detection, but when another technology / B2BUA is involved where max-forwards can't be used. This is for big loops. The idea is similar to the pike module. However, you bascically look at the to_did and the source IP and if you see more than X calls in Y period, begin to reject them for Z seconds. Simple enough. This has come up a dozen times for me and for now I have to handle it with kludge of memcache, and perl scripts to detect these issues in my cdr. The loops are a bit nuts and are always the results of someone doing something stupid (but hey, it does happen). The loops are like, my customer sends me a call to one of thier own DIDs (they've misrouted it to me) and I send it to my carrier, who sends it to the pstn, back to my customer, back to me, etc.. There may be a ss7 portion in there so it keeps looking like a new call on the SIP side. So without anything, this can clog up my call paths pretty quickly, the proposed feature would blacklist the source_ip to_did combination for a period of time to kill the loop. Thoughts? -Brett ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Feature request: Loop detection
Yeah, that's a great idea actually, I could just concatenate some PVs to form a key like $si-$rU. On Mon, Jun 29, 2009 at 8:53 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Brett, This will be kind of pike but instead of using as input the source IP string, it should use a custom string you build form script, right ? this string will be a kind of key (logical one) to identify the loop. Regards, Bogdan Brett Nemeroff wrote: Hey All, I was wanting to submit a feature request for loop detection. Specifically NOT SIP loop detection, but when another technology / B2BUA is involved where max-forwards can't be used. This is for big loops. The idea is similar to the pike module. However, you bascically look at the to_did and the source IP and if you see more than X calls in Y period, begin to reject them for Z seconds. Simple enough. This has come up a dozen times for me and for now I have to handle it with kludge of memcache, and perl scripts to detect these issues in my cdr. The loops are a bit nuts and are always the results of someone doing something stupid (but hey, it does happen). The loops are like, my customer sends me a call to one of thier own DIDs (they've misrouted it to me) and I send it to my carrier, who sends it to the pstn, back to my customer, back to me, etc.. There may be a ss7 portion in there so it keeps looking like a new call on the SIP side. So without anything, this can clog up my call paths pretty quickly, the proposed feature would blacklist the source_ip to_did combination for a period of time to kill the loop. Thoughts? -Brett ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Error to subscribe message
Hi Kiran,
Your SUBSCRIBER contains:
RURI: ims.wimaxlab.com
Route: scscf.ims.wimaxlab.com:6060
I gues both ims.xx scscf.ims.xxx point to the same machine,
right ? if so, just add in the begining of your script:
alias=cscf.ims.wimaxlab.com:6060
Regards,
Bogdan
Kiran Kumar wrote:
Hi,
I did not completely understand what I should add in the alias.
The below is the transcript of the logs that I am seeing on opensips.
Jun 24 14:25:07 [7277] DBG:core:parse_msg: method: SUBSCRIBE
Jun 24 14:25:07 [7277] DBG:core:parse_msg: uri:
sip:use...@ims.wimaxlab.com
Jun 24 14:25:07 [7277] DBG:core:parse_msg: version: SIP/2.0
Jun 24 14:25:07 [7277] DBG:core:parse_headers: flags=2
Jun 24 14:25:07 [7277] DBG:core:parse_via_param: found param type 232,
branch = z9hG4bK0325.f1f6a745.0; state=16
Jun 24 14:25:07 [7277] DBG:core:parse_via: end of header reached, state=5
Jun 24 14:25:07 [7277] DBG:core:parse_headers: via found, flags=2
Jun 24 14:25:07 [7277] DBG:core:parse_headers: this is the first via
Jun 24 14:25:07 [7277] DBG:core:receive_msg: After parse_msg...
Jun 24 14:25:07 [7277] DBG:core:receive_msg: preparing to run routing
scripts...
Jun 24 14:25:07 [7277] DBG:maxfwd:is_maxfwd_present: value = 9
Jun 24 14:25:07 [7277] DBG:core:parse_headers: flags=8
Jun 24 14:25:07 [7277] DBG:core:parse_via_param: found param type 232,
branch = z9hG4bK0325.55278e3.0; state=16
Jun 24 14:25:07 [7277] DBG:core:parse_via: end of header reached, state=5
Jun 24 14:25:07 [7277] DBG:core:parse_headers: via found, flags=8
Jun 24 14:25:07 [7277] DBG:core:parse_headers: parse_headers: this is
the second via
Jun 24 14:25:07 [7277] DBG:core:parse_via_param: found param type 232,
branch = z9hG4bK0325.d1f6a745.0; state=16
Jun 24 14:25:07 [7277] DBG:core:parse_via: end of header reached, state=5
Jun 24 14:25:07 [7277] DBG:core:parse_headers: via found, flags=8
Jun 24 14:25:07 [7277] DBG:core:parse_to: end of header reached, state=10
Jun 24 14:25:07 [7277] DBG:core:parse_to: display={},
ruri={sip:use...@ims.wimaxlab.com}
Jun 24 14:25:07 [7277] DBG:core:get_hdr_field: To [31];
uri=[sip:use...@ims.wimaxlab.com]
Jun 24 14:25:07 [7277] DBG:core:get_hdr_field: to body
[sip:use...@ims.wimaxlab.com
]
Jun 24 14:25:07 [7277] DBG:uri:has_totag: no totag
Jun 24 14:25:07 [7277] DBG:core:parse_headers: flags=78
Jun 24 14:25:07 [7277] DBG:core:get_hdr_field: cseq CSeq: 2
SUBSCRIBE
Jun 24 14:25:07 [7277] DBG:tm:t_lookup_request: start searching:
hash=21040, isACK=0
Jun 24 14:25:07 [7277] DBG:tm:matching_3261: RFC3261 transaction
matching failed
Jun 24 14:25:07 [7277] DBG:tm:t_lookup_request: no transaction found
Jun 24 14:25:07 [7277] DBG:core:parse_headers: flags=200
Jun 24 14:25:07 [7277] DBG:rr:is_preloaded: is_preloaded: Yes
Jun 24 14:25:07 [7277] DBG:core:grep_sock_info: checking if host==us:
9==9 [127.0.0.1] == [127.0.0.1]
Jun 24 14:25:07 [7277] DBG:core:grep_sock_info: checking if port 7060
matches port 7060
Jun 24 14:25:07 [7277] DBG:rr:after_loose: Topmost route URI:
'sip:127.0.0.1:7060;lr' is me
Jun 24 14:25:07 [7277] DBG:rr:after_loose: URI to be processed:
'sip:iscm...@scscf.ims.wimaxlab.com:6060;lr;s=1;h=0;d=2;a=7369703a757365725f3140696d732e77696d61786c61622e636f6d'
Jun 24 14:25:07 [7277] DBG:rr:after_loose: Next URI is a loose router
Jun 24 14:25:07 [7277] DBG:core:parse_to_param: tag=ea7a4226
Jun 24 14:25:07 [7277] DBG:core:parse_to: end of header reached, state=29
Jun 24 14:25:07 [7277] DBG:core:parse_to: display={},
ruri={sip:ki...@ims.wimaxlab.com}
Attempt to route with preloaded Route's
[sip:ki...@ims.wimaxlab.com/sip:use...@ims.wimaxlab.com/sip:use...@ims.wimaxlab.com/ZjdmOGNjOTIwNTNlMTJmOGFjZjQzYzUwOGVjOTMzNzk.]Jun
24 14:25:07 [7277] DBG:core:parse_headers: flags=
Jun 24 14:25:07 [7277] DBG:core:get_hdr_field: content_length=0
Jun 24 14:25:07 [7277] DBG:core:get_hdr_field: found end of header
Jun 24 14:25:07 [7277] DBG:core:check_via_address: params
10.142.139.4, 10.142.139.4, 0
Jun 24 14:25:07 [7277] DBG:core:destroy_avp_list: destroying list (nil)
Jun 24 14:25:07 [7277] DBG:core:receive_msg: cleaning up
Message: 2
Date: Wed, 24 Jun 2009 19:47:44 +0200
From: I?aki Baz Castillo i...@aliax.net
Subject: Re: [OpenSIPS-Users] FW: Error to subscribe message
Cc: users@lists.opensips.org users@lists.opensips.org
Message-ID:
cc1f582e0906241047hbdf8852h4b252cc994681...@mail.gmail.com
Content-Type: text/plain; charset=UTF-8
2009/6/24 Kiran Kumar k.k.balasubraman...@ftel.co.uk:
Hi,
I am trying to setup presence server on the new version and I am
getting an error 403 Preload Route denied when a subscribe message
is sent.
Please help me to solve this problem.
The SUBSCRIBE contains a Route header and the IP or domain in that
Route header is not local to OpenSIPS.
Make sure you have added that domain
Re: [OpenSIPS-Users] Help with enabling Authorization
Hi, do rm -f modules/db_mysql/*.d and try again Regards, Bogdan srikanth R wrote: I am just starting out in this field so might need a bit more of guidance from you guys. I have a red hat 9 linux machine and am using Opensips as a proxy server. I got the 1.5.1 package and followed a documentation that I found online for red hat 5 and the install instructions at opensips website. First I ignored all mysql stuff and tried running the proxy server without any authentication database and it worked fine. Then I tried to follow the steps to implement authentication. I followed the steps given here http://www.voip-info.org/wiki/view/How+to+install+opensips+in+a+Red+Hat+Enterprise+Linux+5+server+with+mysql+support but got stuck at make prefix=/usr/local install include_module=mysql I get this as the error message make[1]: Entering directory `/usr/src/opensips/modules/db_mysql' dbase.d:1: *** missing separator. Stop. make[1]: Leaving directory `/usr/src/opensips/modules/db_mysql' make: *** [modules] Error 2 The mysql package that I have is mysql-server-3.23.54a-11 Thanks fro your help. Srikanth Cricket on your mind? Visit the ultimate cricket website. Enter now! http://in.rd.yahoo.com/tagline_cricket_1/*http://cricket.yahoo.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 1.5.1 with Postgres
I'm trying to get OpenSIPS 1.5.1 running with Postgres (8.1) on SLES10 SP2 I've created the database OK. When I try and start the main OpenSIPS daemon, I get the following error in /var/log/messages: Jun 29 11:22:51 a301-sls1-vm3 /usr/local/sbin/opensips[21062]: ERROR:core:db_check_api: module db_pgsql does not export db_use_table function Jun 29 11:22:51 a301-sls1-vm3 /usr/local/sbin/opensips[21062]: ERROR:auth_db:mod_init: unable to bind to a database driver Jun 29 11:22:51 a301-sls1-vm3 /usr/local/sbin/opensips[21062]: ERROR:core:init_mod: failed to initialize module auth_db Jun 29 11:22:51 a301-sls1-vm3 /usr/local/sbin/opensips[21062]: ERROR:core:main: error while initializing modules The error message mentions db_pgsql, yet the module name the source code all refer to db_postgres. Is this a bug in the code, or something stupid I've done ? Thanks, GTG ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DIALOG not deleted on BYE
Hi Uwe, Uwe Kastens wrote: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. yes, it sounds like. Unfort I was not able to find out what the states and the events means. You can find the meaning of each state in: modules/dialog/dlg_hash.h So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? I can try : ) could you (privately if needed) please send me the the full logs for the entire call (debug=6) - for the non working part. Thanks and regards, Bogdan BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Update = dlg_handlers vs. dlg_db_handler DIALOG not deleted on BYE
Hi Uwe, please see my previous email regarding the traces. Regards, Bogdan Uwe Kastens wrote: Hi, If I use db_mode=1 the dialog is deleted from the database (as expected) but not from the memory. I will test with a fresh installation and maybe open a bug report. BR Uwe Uwe Kastens schrieb: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. Unfort I was not able to find out what the states and the events means. So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Documentation - $dlg_var
Hi Thomas,
yes, the correct format is $dlg_var(whatever) . Where is the bogus docs
you are referring at?
Thanks and Regards,
Bogdan
Thomas Gelf wrote:
Documentation (dialog module) shows examples with $dlg_var(whatever).
In my believes this is wrong, as a quick test proved that
$dlg_var('whatever') != $dlg_var(whatever) != $dlg_var(whatever)
I guess the correct form is $dlg_var(whatever), isn't it?
Best regards,
Thomas Gelf
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] unusual RURI format
I actually do this with the dialplan module right now.. seems to be working well.. For some reason, I wsa getting a 500 Overlapping requests error when I first did this.. Changed the regex and it seems to be happy now.. We'll see.. Thanks, Brett On Sun, Jun 28, 2009 at 1:18 PM, Iñaki Baz Castillo i...@aliax.net wrote: 2009/6/26 Brett Nemeroff br...@nemeroff.com: All, I've got a customer that is sending me calls with an RURI like this: sip:1311207;npdi=yes;rn=1310...@1.2.3.4 when I use anything that parses the RURI like $rU, it shows $rU=sip:13151207;npdi=yes;rn=131 Which is exactly everything from sip: to the first @ sign. The customer isn't able to change this RURI (it's coming from their downstream) What can I do to properly have $rU parse this up? That SIP URI seems to come from a TEL conversion. However the URI username is 1311207;npdi=yes;rn=131. An user shouldn't send such an username to a server expecting that the server will drop after the ; in the username. If you want to parse it, I would use a substitution by deleting all the content after the first ; in the RURI username using AVPOPS. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] alias_db_find() and $ru
Hi Thomas, that is true - I made a fix on SVN - could you test again, please. Thank and regards, Bogdan Thomas Gelf wrote: While playing around with alias_db_find (great feature!) I discovered that the following is fine: if (alias_db_find(dbaliases, $var(vmuri), $var(vmuser), rd)) This variant: if (alias_db_find(dbaliases, $var(vmuri), $ru, rd)) passes config file check, but startup fails (showing lot of garbage and special chars): ERROR:alias_db:find_fixup: PVAR ... AVP ... VAR ERROR:lotsofstrangechars Cheers, Thomas Gelf ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] unusual RURI format
Hi Brett, Brett Nemeroff wrote: All, I've got a customer that is sending me calls with an RURI like this: sip:1311207;npdi=yes;rn=1310...@1.2.3.4 when I use anything that parses the RURI like $rU, it shows $rU=sip:13151207;npdi=yes;rn=131 Which is exactly everything from sip: to the first @ sign. and according to the RFC3261, this is correct and a SIP-URI is sip: [ userinfo ] hostport uri-parameters [ headers ] And $rU returns this userinfo (see 19.1.1 SIP and SIPS URI Components). The customer isn't able to change this RURI (it's coming from their downstream) What can I do to properly have $rU parse this up? I think you already found the answer :) - using dialplan to extract only the username part from userinfo. Regards, Bogdan Thanks, Brett ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:core:db_allocate_rows: no memory left - Please help
Hi Mani, I fixed this on 1.6 (devel) version by adding fetch support in the PUA module, so now it can add as many row as possible without any mem issues. Could you give it a try? if you confirm the fix, I will do a backport to 1.5 also. Best regards, Bogdan mani sivaraman wrote: I'm using opensips 1.5.1 for the past 1 week. Suddenly I get this error below when starting. What is going on. Jun 26 15:52:14 [2535] ERROR:core:db_allocate_rows: no memory left Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_convert_rows: no private memory left Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_convert_result: error while converting rows Jun 26 15:52:14 [2535] DBG:core:db_free_columns: freeing result columns at 0x81a9b78 Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_store_result: error while converting result Jun 26 15:52:14 [2535] ERROR:core:db_do_query: error while storing resultJun 26 15:52:14 [2535] ERROR:pua:db_restore: while querrying table Jun 26 15:52:14 [2535] ERROR:pua:mod_init: while restoring hash_table Jun 26 15:52:14 [2535] ERROR:core:init_mod: failed to initialize module pua Jun 26 15:52:14 [2535] ERROR:core:main: error while initializing modules Jun 26 15:52:14 [2535] DBG:xmpp:destroy: cleaning up... Jun 26 15:52:14 [2535] DBG:pua:destroy: destroying module ... Jun 26 15:52:14 [2535] DBG:db_mysql:db_mysql_submit_query: discon reset for 135960296 Jun 26 15:52:14 [2535] DBG:core:pool_remove: removing connection from the pool Jun 26 15:52:14 [2535] DBG:presence_xml:destroy: start Jun 26 15:52:14 [2535] NOTICE:presence:destroy: destroy module ... Jun 26 15:52:14 [2535] DBG:xlog:destroy: destroy module... Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown : start Jun 26 15:52:14 [2535] DBG:tm:unlink_timer_lists: emptying DELETE list Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: emptying hash table Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: releasing timers Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: removing semaphores Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: destroying callback lists Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown : done Jun 26 15:52:14 [2535] DBG:core:destroy_tls: entered Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy: Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy: destroying the shared memory lock ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] pua_xmpp xmpp2Sip simple presence does not work
I'm trying hard to debug the xmpp-2-SIP SIMPLE presence issue. I have opensips 1.5.1 and Jabberd2 server installed on same physical box. I have configured pua_xmpp, pua and xmpp module on used the example config given for pua_xmpp module. I'm able to send MESSAGE back and forth between sip and xmpp buddy. XMPP (Pidgin) client can detect the presence of sip buddy. But The SIP Client (opensips server ) is not getting any kind of presence status from xmpp server. I get the below error when running this setup. I'm not sure if opensips server is subscribing to the xmpp buddy first of all. When the XMPP buddy goes offline/online I can see nothing (no debug info) being printed on the console of openeips server. Please any one , please help to debug this xmpp to sip cominication issue. I really apprecaite your help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS boot Camp
Hi, Indeed the Bootcamp in San Francisco was fully booked with more than 2 weeks before - we had to turn people down. I was discussion with Flavio about the e-Bootcamp version - were you can get access via an elearning platform to the course materials for let's say 2 months and you can study and run the seminars by yourself. Included, you will have the possibility to fire questions to the teachers if you have something to clarify or if you got stuck with the labs What do you think of such approach ? Regards, Bogdan bay2x1 wrote: Hi to all! I was interested in attending the boot camp scheduled last June 15 however due to financial problems I wasn't able to do so. I also ask one of my friends who was working within the area to attend the boot camp, but unfortunately when he tried to register it was already fully booked. I just want to ask if there are available online videos of the discussions on the seminar, and if there is a downloadable copy of DVD-disc which was distributed on the seminar. I believe that the DVD would be a good reference specially the sample configuration files. I am hoping that in the near future there would be a boot camp which will be held on any South East Asian country, (e.g.; Singapore, Hong Kong, Philippines, Thailand). ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:core:db_allocate_rows: no memory left - Please help
Thanks Bogdan, I will definitely try the devel (1.6). I cannot install 1.6 on the current server I have. I have to setup another box and then try it out. I will let you know. Mean while I'm trying hard to make xmpp2simple presence status work. Any help or pointer would be appreciated. Thanks Mani On Mon, Jun 29, 2009 at 11:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Mani, I fixed this on 1.6 (devel) version by adding fetch support in the PUA module, so now it can add as many row as possible without any mem issues. Could you give it a try? if you confirm the fix, I will do a backport to 1.5 also. Best regards, Bogdan mani sivaraman wrote: I'm using opensips 1.5.1 for the past 1 week. Suddenly I get this error below when starting. What is going on. Jun 26 15:52:14 [2535] ERROR:core:db_allocate_rows: no memory left Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_convert_rows: no private memory left Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_convert_result: error while converting rows Jun 26 15:52:14 [2535] DBG:core:db_free_columns: freeing result columns at 0x81a9b78 Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_store_result: error while converting result Jun 26 15:52:14 [2535] ERROR:core:db_do_query: error while storing resultJun 26 15:52:14 [2535] ERROR:pua:db_restore: while querrying table Jun 26 15:52:14 [2535] ERROR:pua:mod_init: while restoring hash_table Jun 26 15:52:14 [2535] ERROR:core:init_mod: failed to initialize module pua Jun 26 15:52:14 [2535] ERROR:core:main: error while initializing modules Jun 26 15:52:14 [2535] DBG:xmpp:destroy: cleaning up... Jun 26 15:52:14 [2535] DBG:pua:destroy: destroying module ... Jun 26 15:52:14 [2535] DBG:db_mysql:db_mysql_submit_query: discon reset for 135960296 Jun 26 15:52:14 [2535] DBG:core:pool_remove: removing connection from the pool Jun 26 15:52:14 [2535] DBG:presence_xml:destroy: start Jun 26 15:52:14 [2535] NOTICE:presence:destroy: destroy module ... Jun 26 15:52:14 [2535] DBG:xlog:destroy: destroy module... Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown : start Jun 26 15:52:14 [2535] DBG:tm:unlink_timer_lists: emptying DELETE list Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: emptying hash table Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: releasing timers Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: removing semaphores Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: destroying callback lists Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown : done Jun 26 15:52:14 [2535] DBG:core:destroy_tls: entered Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy: Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy: destroying the shared memory lock ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SOLVED: Re: OpenSIPS 1.5.1 with Postgres
Hi Gordon, But a postgres URL should look like: postgres://opensips:opensip...@localhost/opensips This is the intended way of using the postgres DB. Regards, Bogdan Gordon Ross wrote: On 29/06/2009 15:18, Gordon Ross gr...@ucs.cam.ac.uk wrote: I'm trying to get OpenSIPS 1.5.1 running with Postgres (8.1) on SLES10 SP2 I've created the database OK. When I try and start the main OpenSIPS daemon, I get the following error in /var/log/messages: Jun 29 11:22:51 a301-sls1-vm3 /usr/local/sbin/opensips[21062]: ERROR:core:db_check_api: module db_pgsql does not export db_use_table function Jun 29 11:22:51 a301-sls1-vm3 /usr/local/sbin/opensips[21062]: ERROR:auth_db:mod_init: unable to bind to a database driver [snip] The postgres support is broken in OpenSIP. The module names are incorrect. I've posted a bug, containing the fix in the bug tracker, bug ID 2814081. GTG ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SOLVED: Re: OpenSIPS 1.5.1 with Postgres
On 29/06/2009 17:19, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: But a postgres URL should look like: postgres://opensips:opensip...@localhost/opensips This is the intended way of using the postgres DB. I'm not sure where we want to have this discussion (on this list, on another list or in the bug tracker) I've posted a comment on the tracker basically saying my hack makes the postgres configuration more consistent across OpenSIPS. The problem (as I see it) is that Postgres naming in OpenSIP is *NOT* consistent. In the scripts, it's referred to as PGSQL, yet in the C code, it's postgres. GTG ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SOLVED: Re: OpenSIPS 1.5.1 with Postgres
Hi Gordan, Gordon Ross wrote: On 29/06/2009 17:19, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: But a postgres URL should look like: postgres://opensips:opensip...@localhost/opensips This is the intended way of using the postgres DB. I'm not sure where we want to have this discussion (on this list, on another list or in the bug tracker) Let's continue here :) I've posted a comment on the tracker basically saying my hack makes the postgres configuration more consistent across OpenSIPS. The problem (as I see it) is that Postgres naming in OpenSIP is *NOT* consistent. In the scripts, it's referred to as PGSQL, yet in the C code, it's postgres. I see your point - and I agree with you ; I would rather change the scripts (from PGSQL to POSTGRES) and the C code is number one and I prefer to do the changes in the satellite tools (much easier to cope for the users). Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:core:db_allocate_rows: no memory left - Please help
Hi Mani,
if this will make the testing easier, here is the patch you can apply
against 1.5.
Regards,
Bogdan
mani sivaraman wrote:
Thanks Bogdan, I will definitely try the devel (1.6). I cannot install
1.6 on the current server I have. I have to setup another box and then
try it out. I will let you know. Mean while I'm trying hard to make
xmpp2simple presence status work. Any help or pointer would be
appreciated.
Thanks
Mani
On Mon, Jun 29, 2009 at 11:04 AM, Bogdan-Andrei Iancu
bog...@voice-system.ro mailto:bog...@voice-system.ro wrote:
Hi Mani,
I fixed this on 1.6 (devel) version by adding fetch support in the
PUA module, so now it can add as many row as possible without any
mem issues. Could you give it a try? if you confirm the fix, I
will do a backport to 1.5 also.
Best regards,
Bogdan
mani sivaraman wrote:
I'm using opensips 1.5.1 for the past 1 week. Suddenly I get
this error below when starting. What is going on.
Jun 26 15:52:14 [2535] ERROR:core:db_allocate_rows: no memory left
Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_convert_rows:
no private memory left
Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_convert_result:
error while converting rows
Jun 26 15:52:14 [2535] DBG:core:db_free_columns: freeing
result columns at 0x81a9b78
Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_store_result:
error while converting result
Jun 26 15:52:14 [2535] ERROR:core:db_do_query: error while
storing resultJun 26 15:52:14 [2535] ERROR:pua:db_restore:
while querrying table
Jun 26 15:52:14 [2535] ERROR:pua:mod_init: while restoring
hash_table
Jun 26 15:52:14 [2535] ERROR:core:init_mod: failed to
initialize module pua
Jun 26 15:52:14 [2535] ERROR:core:main: error while
initializing modules
Jun 26 15:52:14 [2535] DBG:xmpp:destroy: cleaning up...
Jun 26 15:52:14 [2535] DBG:pua:destroy: destroying module ...
Jun 26 15:52:14 [2535] DBG:db_mysql:db_mysql_submit_query:
discon reset for 135960296
Jun 26 15:52:14 [2535] DBG:core:pool_remove: removing
connection from the pool
Jun 26 15:52:14 [2535] DBG:presence_xml:destroy: start
Jun 26 15:52:14 [2535] NOTICE:presence:destroy: destroy module ...
Jun 26 15:52:14 [2535] DBG:xlog:destroy: destroy module...
Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown : start
Jun 26 15:52:14 [2535] DBG:tm:unlink_timer_lists: emptying
DELETE list
Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: emptying hash table
Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: releasing timers
Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: removing semaphores
Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: destroying callback
lists
Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown : done
Jun 26 15:52:14 [2535] DBG:core:destroy_tls: entered
Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy:
Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy: destroying
the shared memory lock
___
Users mailing list
Users@lists.opensips.org mailto:Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Index: modules/pua/pua.c
===
--- modules/pua/pua.c (revision 5804)
+++ modules/pua/pua.c (revision 5805)
@@ -338,17 +338,31 @@
LM_ERR(in use table\n);
return -1;
}
-
- if(pua_dbf.query(pua_db,0, 0, 0, result_cols,0, n_result_cols, 0,res) 0)
- {
- LM_ERR(while querrying table\n);
- if(res)
+
+ if (DB_CAPABILITY(pua_dbf, DB_CAP_FETCH)) {
+ if(pua_dbf.query(pua_db,0, 0, 0, result_cols,0, n_result_cols, 0,0) 0)
{
- pua_dbf.free_result(pua_db, res);
- res = NULL;
+ LM_ERR(while querying table\n);
+ return -1;
}
- return -1;
+ if(pua_dbf.fetch_result(pua_db, res, 500 /*rows*/)0)
+ {
+ LM_ERR(Error fetching rows\n);
+ return -1;
+ }
+ } else {
+ if(pua_dbf.query(pua_db,0, 0, 0,result_cols,0,n_result_cols,0,res) 0)
+ {
+ LM_ERR(while querrying table\n);
+ if(res)
+ {
+pua_dbf.free_result(pua_db, res);
+res = NULL;
+ }
+ return -1;
+ }
}
+
if(res== NULL)
return -1;
@@ -362,200 +376,212 @@
LM_DBG(found %d db entries\n, res-n);
- for(i =0 ; i res-n ; i++)
- {
- row = res-rows[i];
- row_vals = ROW_VALUES(row);
-
- pres_uri.s= (char*)row_vals[puri_col].val.string_val;
- pres_uri.len = strlen(pres_uri.s);
-
- LM_DBG(pres_uri= %.*s\n, pres_uri.len, pres_uri.s);
-
- memset(etag, 0, sizeof(str));
- memset(tuple_id, 0, sizeof(str));
- memset(watcher_uri, 0, sizeof(str));
- memset(call_id, 0, sizeof(str));
- memset(to_tag,
Re: [OpenSIPS-Users] Feature request: Loop detection
OK, let me see how difficult is to re-design the pike module, as so far, the way the internal data is kept is highly IP-format dependent. Regards, Bogdan Brett Nemeroff wrote: Yeah, that's a great idea actually, I could just concatenate some PVs to form a key like $si-$rU. On Mon, Jun 29, 2009 at 8:53 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Brett, This will be kind of pike but instead of using as input the source IP string, it should use a custom string you build form script, right ? this string will be a kind of key (logical one) to identify the loop. Regards, Bogdan Brett Nemeroff wrote: Hey All, I was wanting to submit a feature request for loop detection. Specifically NOT SIP loop detection, but when another technology / B2BUA is involved where max-forwards can't be used. This is for big loops. The idea is similar to the pike module. However, you bascically look at the to_did and the source IP and if you see more than X calls in Y period, begin to reject them for Z seconds. Simple enough. This has come up a dozen times for me and for now I have to handle it with kludge of memcache, and perl scripts to detect these issues in my cdr. The loops are a bit nuts and are always the results of someone doing something stupid (but hey, it does happen). The loops are like, my customer sends me a call to one of thier own DIDs (they've misrouted it to me) and I send it to my carrier, who sends it to the pstn, back to my customer, back to me, etc.. There may be a ss7 portion in there so it keeps looking like a new call on the SIP side. So without anything, this can clog up my call paths pretty quickly, the proposed feature would blacklist the source_ip to_did combination for a period of time to kill the loop. Thoughts? -Brett ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Feature request: Loop detection
Why is that? Does it provide rate-limiting for subnets of sending traffic as well? Seems like the function needs to be redone altogether with the whole tree business.. -Brett On Mon, Jun 29, 2009 at 12:02 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: OK, let me see how difficult is to re-design the pike module, as so far, the way the internal data is kept is highly IP-format dependent. Regards, Bogdan Brett Nemeroff wrote: Yeah, that's a great idea actually, I could just concatenate some PVs to form a key like $si-$rU. On Mon, Jun 29, 2009 at 8:53 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Brett, This will be kind of pike but instead of using as input the source IP string, it should use a custom string you build form script, right ? this string will be a kind of key (logical one) to identify the loop. Regards, Bogdan Brett Nemeroff wrote: Hey All, I was wanting to submit a feature request for loop detection. Specifically NOT SIP loop detection, but when another technology / B2BUA is involved where max-forwards can't be used. This is for big loops. The idea is similar to the pike module. However, you bascically look at the to_did and the source IP and if you see more than X calls in Y period, begin to reject them for Z seconds. Simple enough. This has come up a dozen times for me and for now I have to handle it with kludge of memcache, and perl scripts to detect these issues in my cdr. The loops are a bit nuts and are always the results of someone doing something stupid (but hey, it does happen). The loops are like, my customer sends me a call to one of thier own DIDs (they've misrouted it to me) and I send it to my carrier, who sends it to the pstn, back to my customer, back to me, etc.. There may be a ss7 portion in there so it keeps looking like a new call on the SIP side. So without anything, this can clog up my call paths pretty quickly, the proposed feature would blacklist the source_ip to_did combination for a period of time to kill the loop. Thoughts? -Brett ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Feature request: Loop detection
the IPs are internally kept in a tree that assumes that the data is IP-only - each node can have maximum 256 sub-nodes, but with some twists I can do it more generic, to support any kind of data.. Regards, Bogdan Brett Nemeroff wrote: Why is that? Does it provide rate-limiting for subnets of sending traffic as well? Seems like the function needs to be redone altogether with the whole tree business.. -Brett On Mon, Jun 29, 2009 at 12:02 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: OK, let me see how difficult is to re-design the pike module, as so far, the way the internal data is kept is highly IP-format dependent. Regards, Bogdan Brett Nemeroff wrote: Yeah, that's a great idea actually, I could just concatenate some PVs to form a key like $si-$rU. On Mon, Jun 29, 2009 at 8:53 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Brett, This will be kind of pike but instead of using as input the source IP string, it should use a custom string you build form script, right ? this string will be a kind of key (logical one) to identify the loop. Regards, Bogdan Brett Nemeroff wrote: Hey All, I was wanting to submit a feature request for loop detection. Specifically NOT SIP loop detection, but when another technology / B2BUA is involved where max-forwards can't be used. This is for big loops. The idea is similar to the pike module. However, you bascically look at the to_did and the source IP and if you see more than X calls in Y period, begin to reject them for Z seconds. Simple enough. This has come up a dozen times for me and for now I have to handle it with kludge of memcache, and perl scripts to detect these issues in my cdr. The loops are a bit nuts and are always the results of someone doing something stupid (but hey, it does happen). The loops are like, my customer sends me a call to one of thier own DIDs (they've misrouted it to me) and I send it to my carrier, who sends it to the pstn, back to my customer, back to me, etc.. There may be a ss7 portion in there so it keeps looking like a new call on the SIP side. So without anything, this can clog up my call paths pretty quickly, the proposed feature would blacklist the source_ip to_did combination for a period of time to kill the loop. Thoughts? -Brett ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org mailto:Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DIALOG not deleted on BYE
Hi Uwe, Thanks for the traces. Looking at the opensips logs, I say you do loose_route() twice for the ACK which looks twice for the dialog and increase the ref twice for the dialogthis is why the ref never gets back to 0 to allow the dialog to be destroyed.. Could you confirm this for me ? even if it's a script error , the dialog module should cope with it..I will look for a fix. Thanks and regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Uwe Kastens wrote: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. yes, it sounds like. Unfort I was not able to find out what the states and the events means. You can find the meaning of each state in: modules/dialog/dlg_hash.h So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? I can try : ) could you (privately if needed) please send me the the full logs for the entire call (debug=6) - for the non working part. Thanks and regards, Bogdan BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No RADIUS traffic
hi, For radius support these packages are needed. *libradius-ng -libs and devel headers*- if you want to use functionalities with radius support - authentication, accounting, group support, etc On Wed, Jun 24, 2009 at 10:14 AM, Leon Li leon...@aarnet.edu.au wrote: Hi Uwe, The file doesn't exist. :( Could you confirm my following installation is enough for OpenSIP + RADIUS? 1. FreeRADIUS 2.1.3 2. radiusclient-ng 0.5.6 3. openSIP 1.5.1 Do I need libradius-ng-dev or libradius-ng as well? My system is Red Hat 5. Regards, Leon -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: Tuesday, 23 June 2009 5:31 PM To: Leon Li Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] No RADIUS traffic Li, I was wondering about the answer from radius: WARNING: Ignoring Status-Server request due to security configuration If I try the same I will get an answer like: Received response ID 196, code 2, length = 20 Could you please check your shared secret. Also, I cannot find file /var/run/radius.seq. Is it created automatically? I should be there if radius will work - but remember your permissions. You can try one thing: set fork=no in opensips.cfg, install strace and start opensips with strace -f -e open opensips. Now start one attempt to register etc.pp. and watch the line with the seq. [pid 20680] open(/var/run/opensips/radius.seq, O_RDWR|O_CREAT|O_APPEND, 0666) = 13 BR Uwe Leon Li schrieb: Uwe, I got the following from RADIUS when issue the command you gave. rad_recv: Status-Server packet from host 127.0.0.1:39297, id=17, length=38 WARNING: Ignoring Status-Server request due to security configuration --- Walking the entire request list --- Nothing to do. Sleeping until we see a request. rad_recv: Status-Server packet from host 127.0.0.1:39297, id=17, length=38 WARNING: Ignoring Status-Server request due to security configuration --- Walking the entire request list --- So I assume that the radius server is working? Also, I cannot find file /var/run/radius.seq. Is it created automatically? Regards, Leon -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: Wednesday, 17 June 2009 6:01 PM To: Leon Li Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] No RADIUS traffic Leon, mysql.so in opensips is not needed for the radius authentication. Shared secrets for radius are correct? Anyway you should see some traffic on the radius server. Could you please test echo Message-Authenticator = 0x00 | radclient 127.0.0.1:1812 status shared secret You should see then traffic on radiusd -X If yes I would start checking permissions again BR uwe Leon Li schrieb: Hi Ashwini, I have added param for aut_radius, but no luck. L Why do I need mysql.so if the radius server will host all users credential? Regards, Leon *From:* ASHWINI NAIDU [mailto:ashwini.na...@gmail.com] *Sent:* Monday, 15 June 2009 2:52 PM *To:* Leon Li *Cc:* Uwe Kastens; users@lists.opensips.org *Subject:* Re: [OpenSIPS-Users] No RADIUS traffic On Mon, Jun 15, 2009 at 10:19 AM, ASHWINI NAIDU ashwini.na...@gmail.com mailto:ashwini.na...@gmail.com wrote: hi leon, But i do not see your openser communicating with radiusclient. modparam(auth_radius, radius_config, /etc/radiusclient-ng/radiusclient.conf) mention the path of radiusclient.conf properly. Your mysql support is also commented. *loadmodule mysql.so* On Mon, Jun 15, 2009 at 5:13 AM, Leon Li leon...@aarnet.edu.au mailto:leon...@aarnet.edu.au wrote: Here it is. ### Global Parameters # debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ debug=6 fork=no log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = /usr/local/etc/openser/tls/user/user-cert.pem #tls_private_key = /usr/local/etc/openser/tls/user/user-privkey.pem
Re: [OpenSIPS-Users] Hairpin routing and Loop avoidance
On Wed, Jun 24, 2009 at 06:01:52PM -0400, I?aki Baz Castillo wrote: El Mi?rcoles, 24 de Junio de 2009, Steven E. Ames escribi?: Hi. I have an OpenSIPS and Asterisk setup. Incoming calls come from VoIP Carrier to OpenSIPS. OpenSIPS does some dbaliases because sometimes multiple numbers are assigned to same asterisk extension. This all works great. However, when asterisk makes a call outward for a number that is actually local then I get a Loop and asterisk is unhappy. Example: I have 2 incoming DIDs (111-222- and 111-222-3334). On OpenSIPS in dbaliases I translated 111-222- to 111-222-3334 and send it to asterisk. All is fine. On asterisk it knows about 3334 but not . So if another extension on asterisk dials 111-222- it gets to OpenSIPS. OpenSIPS does know about and knows how to handle it. It converts it to 3334 and sends it back to asterisk. Voila. Loop. Now the actual behavior is what I want but I want to modify the SIP INVITE such that asterisk will accept it and not gripe about the loop. Any pointers? This is a kwnown bug of Asterisk. Does that translate as I'm out of luck pending asterisk people fixing this, or can someone point me at a workaround or the right phrase to google up some additional info on the asterisk bug? -steve ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] pua_xmpp xmpp2Sip simple presence does not work
I'm trying hard to debug the xmpp-2-SIP SIMPLE presence issue. I have opensips 1.5.1 and Jabberd2 server installed on same physical box. I have configured pua_xmpp, pua and xmpp module on used the example config given for pua_xmpp module. I'm able to send MESSAGE back and forth between sip and xmpp buddy. XMPP (Pidgin) client can detect the presence of sip buddy. But The SIP Client (opensips server ) is not getting any kind of presence status from xmpp server. I get the below error when running this setup. I'm not sure if opensips server is subscribing to the xmpp buddy first of all. When the XMPP buddy goes offline/online I can see nothing (no debug info) being printed on the console of openeips server. Please any one , please help to debug this xmpp to sip cominication issue. I really apprecaite your help. On Fri, Jun 26, 2009 at 11:03 AM, mani sivaraman mani.opens...@gmail.comwrote: The space is presence only here in the email (due to some formatting issue). The cfg file on opensips server does not have any space in between. I get presence.winfo NOTIFY. I'm missing only the presence NOTIFY. I see the repeated error like this one ERROR:presence:get_stored_info: record not found in hash_table ERROR:presence:handle_subscribe: getting stored info In the below logs, I always see for event presence.winfo, but no for event presence. This means that there are is lot of activity for watcher Info Subscribe and NOTIFY. But nothing for presence event. Also the method pua_xmpp_req_winfo sounds more like this will send subscribe/notify for winfo, not for preesnce event. IS there a seperate method to call to request subscribe for preesnce event. Or do you call presence subscribe in this same function. This could be the reason why there are no SUBSCRIBEs or presence NOTIFIEs back. Please tell me what this error is, os that I can try to debug this. Coudl you do paid support for setting up pua_xmpp module ? Or If you can't could you point any one else who can do this. I'm under pressure to complete this. Any information reg. consultation is appreciated. Here is the full debug. Listening on udp: 172.16.0.139 [172.16.0.139]:5060 Aliases: *: sips01.smithmicro.com:* Jun 26 10:41:27 [25843] NOTICE:presence:child_init: init_child [0] pid [25843] sips01:/usr/local/etc/opensips# Jun 26 10:42:47 [25845] INFO:presence:send_notify_request: NOTIFY sip:jabb1*xmpp.smithm icro.com@ sip-xmpp.smithmicro.com via sip:jabb1*xmpp.smithmicro@sip-xmpp.smithmicro.com on behalf of sip:rbox...@sip s01.smithmicro.com for event presence Jun 26 10:42:47 [25847] ERROR:pua_xmpp:Notify2Xmpp: 'To' header ALREADY PARSED:sip:jabb1*xmpp.smithmicro@sip-xmpp. smithmicro.com Jun 26 10:42:47 [25847] ERROR:pua_xmpp:Notify2Xmpp: 'From' header not parsed Jun 26 10:42:47 [25846] INFO:presence:send_notify_request: NOTIFY sip:jabb1*xmpp.smithmicro@sip-xmpp.smithmicro.com via sip:jabb1* xmpp.smithmicro@sip-xmpp.smithmicro.com on behalf of sip:msivara...@sips01.smithmicro.comsip%3amsivara...@sips01.smithmicro.comfor event presence Jun 26 10:42:47 [25844] ERROR:pua_xmpp:Notify2Xmpp: 'To' header ALREADY PARSED:sip:jabb1*xmpp.smithmicro@sip-xmpp. smithmicro.com Jun 26 10:42:47 [25844] ERROR:pua_xmpp:Notify2Xmpp: 'From' header not parsed Jun 26 10:43:03 [25847] ERROR:presence:get_stored_info: record not found in hash_table Jun 26 10:43:03 [25847] ERROR:presence:handle_subscribe: getting stored info Jun 26 10:43:03 [25846] ERROR:presence:get_stored_info: record not found in hash_table Jun 26 10:43:03 [25846] ERROR:presence:handle_subscribe: getting stored info Jun 26 10:43:03 [25847] ERROR:presence:get_stored_info: record not found in hash_table Jun 26 10:43:03 [25847] ERROR:presence:handle_subscribe: getting stored info Jun 26 10:43:03 [25844] INFO:presence:send_notify_request: NOTIFY sip:jabb1*xmpp.smithmicro@sip-xmpp.smithmicro.com via sip: 172.16.0.139:5060 on behalf of sip:jabb1*xmpp.smithmicro.com@ sip-xmpp.smithmicro.com for event presence.winfo Jun 26 10:43:03 [25844] INFO:presence:send_notify_request: NOTIFY sip:jabb2*xmpp.smithmicro@sip-xmpp.smithmicro.com via sip: 172.16.0.139:5060 on behalf of sip:jabb2*xmpp.smithmicro.com@ sip-xmpp.smithmicro.com for event presence.winfo Jun 26 10:43:04 [25845] INFO:presence:send_notify_request: NOTIFY sip:jabb2*xmpp.smithmicro@sip-xmpp.smithmicro.com via sip:jabb2*xmpp.smithmicro@sip-xmpp.smithmicro.com on behalf of sip:rbox...@sips01.smithmicro.com sip%3arbox...@sips01.smithmicro.comfor event pre sence Jun 26 10:43:04 [25844] ERROR:pua_xmpp:Notify2Xmpp: 'To' header ALREADY PARSED:sip:jabb2*xmpp.smithmicro@sip-xmpp. smithmicro.com Jun 26 10:43:04 [25844] ERROR:pua_xmpp:Notify2Xmpp: 'From' header not parsed Jun 26 10:43:04 [25846] INFO:presence:send_notify_request: NOTIFY sip:jabb2*xmpp.smithmicro@sip-xmpp.smithmicro.com via
Re: [OpenSIPS-Users] pua_xmpp sip-xmpp gateway - works one way, not other way
The space is presence only here (due to some formatting issue). The cfg file does not have any space in between. I get presence.winfo NOTIFY. I'm missing only the presence NOTIFY. I see the repeated error like this one ERROR:presence:get_stored_info: record not found in hash_table ERROR:presence:handle_subscribe: getting stored info This could be the reason why there are no SUBSCRIBEs or presence NOTIFIES back. Please tell me what this error is, os that I can try to debug this. Coudl you do paid support for setting up pua_xmpp module ? Or If you can't could you point any one else who can do this. I'm under pressure to complete this. Any information reg. consultation is appreciated. Here is the full debug. Listening on udp: 172.16.0.139 [172.16.0.139]:5060 Aliases: *: sips01.smithmicro.com:* Jun 26 10:41:27 [25841] INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) Jun 26 10:41:27 [25843] NOTICE:core:main: version: opensips 1.5.1-tls (i386/linux) Jun 26 10:41:27 [25843] INFO:core:main: using 32 Mb shared memory Jun 26 10:41:27 [25843] INFO:core:main: using 1 Mb private memory per process Jun 26 10:41:27 [25843] NOTICE:signaling:mod_init: initializing module ... Jun 26 10:41:27 [25843] INFO:sl:mod_init: Initializing StateLess engine Jun 26 10:41:27 [25843] INFO:tm:mod_init: TM - initializing... sips01:/usr/local/etc/opensips# Jun 26 10:41:27 [25843] INFO:maxfwd:mod_init: in Jun 26 10:41:27 [25843] INFO:usrloc:ul_init_locks: locks array size 512 Jun 26 10:41:27 [25843] INFO:registrar:mod_init: initializing... Jun 26 10:41:27 [25843] INFO:textops:mod_init: initializing... Jun 26 10:41:27 [25843] INFO:xlog:mod_init: initializing... Jun 26 10:41:27 [25843] INFO:acc:mod_init: initializing... Jun 26 10:41:27 [25843] INFO:auth:mod_init: initializing... Jun 26 10:41:27 [25843] INFO:auth_db:mod_init: initializing... Jun 26 10:41:27 [25843] INFO:alias_db:mod_init: initializing... Jun 26 10:41:27 [25843] NOTICE:presence:mod_init: initializing module ... Jun 26 10:41:27 [25843] INFO:core:probe_max_receive_buffer: using a UDP receive Jun 26 10:41:27 [25844] NOTICE:presence:child_init: init_child [1] pid [25844] Jun 26 10:41:27 [25845] NOTICE:presence:child_init: init_child [2] pid [25845] Jun 26 10:41:27 [25846] NOTICE:presence:child_init: init_child [3] pid [25846] Jun 26 10:41:27 [25847] NOTICE:presence:child_init: init_child [4] pid [25847] Jun 26 10:41:27 [25849] NOTICE:presence:child_init: init_child [-1] pid [25849] Jun 26 10:41:27 [25851] NOTICE:presence:child_init: init_child [-2] pid [25851] Jun 26 10:41:27 [25852] NOTICE:presence:child_init: init_child [5] pid [25852] Jun 26 10:41:27 [25853] NOTICE:presence:child_init: init_child [6] pid [25853] Jun 26 10:41:27 [25854] NOTICE:presence:child_init: init_child [7] pid [25854] Jun 26 10:41:27 [25856] NOTICE:presence:child_init: init_child [8] pid [25856] Jun 26 10:41:27 [25858] NOTICE:presence:child_init: init_child [-4] pid [25858] Jun 26 10:41:27 [25843] NOTICE:presence:child_init: init_child [0] pid [25843] sips01:/usr/local/etc/opensips# Jun 26 10:42:47 [25845] INFO:presence:send_notify_request: NOTIFY sip:jabb1*xmpp.smithm icro.com@ sip-xmpp.smithmicro.com via sip:jabb1*xmpp.smithmicro.com@ sip-xmpp.smithmicro.com on behalf of sip:rbox...@sip s01.smithmicro.com for event presence Jun 26 10:42:47 [25847] ERROR:pua_xmpp:Notify2Xmpp: 'To' header ALREADY PARSED:sip:jabb1*xmpp.smithmicro@sip-xmpp. smithmicro.com Jun 26 10:42:47 [25847] ERROR:pua_xmpp:Notify2Xmpp: 'From' header not parsed Jun 26 10:42:47 [25846] INFO:presence:send_notify_request: NOTIFY sip:jabb1* xmpp.smithmicro@sip-xmpp.smithmicro.com via sip:jabb1* xmpp.smithmicro@sip-xmpp.smithmicro.com on behalf of sip:msivara...@sips01.smithmicro.comsip%3amsivara...@sips01.smithmicro.comfor event presence Jun 26 10:42:47 [25844] ERROR:pua_xmpp:Notify2Xmpp: 'To' header ALREADY PARSED:sip:jabb1*xmpp.smithmicro@sip-xmpp. smithmicro.com Jun 26 10:42:47 [25844] ERROR:pua_xmpp:Notify2Xmpp: 'From' header not parsed Jun 26 10:43:03 [25847] ERROR:presence:get_stored_info: record not found in hash_table Jun 26 10:43:03 [25847] ERROR:presence:handle_subscribe: getting stored info Jun 26 10:43:03 [25846] ERROR:presence:get_stored_info: record not found in hash_table Jun 26 10:43:03 [25846] ERROR:presence:handle_subscribe: getting stored info Jun 26 10:43:03 [25847] ERROR:presence:get_stored_info: record not found in hash_table Jun 26 10:43:03 [25847] ERROR:presence:handle_subscribe: getting stored info Jun 26 10:43:03 [25844] INFO:presence:send_notify_request: NOTIFY sip:jabb1* xmpp.smithmicro@sip-xmpp.smithmicro.com via sip:172.16.0.139:5060 on behalf of sip:jabb1*xmpp.smithmicro@sip-xmpp.smithmicro.com for event presence.winfo Jun 26 10:43:03 [25844] INFO:presence:send_notify_request: NOTIFY sip:jabb2* xmpp.smithmicro@sip-xmpp.smithmicro.com via sip:172.16.0.139:5060 on behalf of
Re: [OpenSIPS-Users] sip-xmpp gateway - presence notification does not work
Hi Anca I do not see any error, but I see some warning in tm module. Not sure if they are of concern. I have also pasted the full config file for your review. Yes I did follow the instruction from the opensips example link. Note that I'm using kamailio 1.5.1 because I tried purple initially and then I swityched to pua_xmpp since purple is still new and needs more work. Hope it is ok to use Kamailio for pua_xmpp as well. I have one question about tls and non-tls. Is it a must that I should use only tls version if I want opensips to talk to other publisc xmpp servers like gtalk etc. I know that jabberd2 has the required tls support to talk to gtalk etc, but should the opensips be also use tls version ? Listening on udp: 172.16.0.141 [172.16.0.141]:5060 Aliases: udp: xmpp1.smithmicro.com:5060 Jun 23 08:28:23 [2675] INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) Jun 23 08:28:23 [2677] NOTICE:core:main: version: kamailio 1.5.1-tls (i386/linux) Jun 23 08:28:23 [2677] INFO:core:main: using 32 Mb shared memory Jun 23 08:28:23 [2677] INFO:core:main: using 4 Mb private memory per process Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module db_mysql Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module mi_fifo Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module sl Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module tm kmilo:/usr/local/etc/kamailio# Jun 23 08:28:23 [2677] INFO:tm:mod_init: fr_inv_timer_next value is 30 Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module rr Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module pv Jun 23 08:28:23 [2677] INFO:pv:shvar_init_locks: locks array size 16 Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module maxfwd Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module usrloc Jun 23 08:28:23 [2677] INFO:usrloc:ul_init_locks: locks array size 512 Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module registrar Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module textops Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module uri_db Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module siputils Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module xlog Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module acc Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module auth Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module auth_db Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module presence Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module presence_xml Jun 23 08:28:23 [2677] INFO:core:init_mod: initializing module pua Jun 23 08:28:24 [2677] INFO:core:init_mod: initializing module xmpp Jun 23 08:28:24 [2677] INFO:core:init_mod: initializing module pua_xmpp Jun 23 08:28:24 [2677] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Jun 23 08:28:30 [2681] INFO:presence:send_notify_request: NOTIFY sip:msivara...@172.16.0.141 sip%3amsivara...@172.16.0.141 via sip:msivara...@172.16.1.125:7489;transport=udp on behalf of sip:msivara...@172.16.0.141 sip%3amsivara...@172.16.0.141 for event presence.winfo Jun 23 08:28:30 [2678] INFO:presence:send_notify_request: NOTIFY sip:msivara...@172.16.0.141 sip%3amsivara...@172.16.0.141 via sip:msivara...@172.16.1.125:7489;transport=udp on behalf of sip:msivara...@172.16.0.141 sip%3amsivara...@172.16.0.141 for event presence.winfo Jun 23 08:28:30 [2678] INFO:presence:send_notify_request: NOTIFY sip:msivara...@172.16.0.141 sip%3amsivara...@172.16.0.141 via sip:msivara...@172.16.1.125:7489;transport=udp on behalf of sip:msivara...@172.16.0.141 sip%3amsivara...@172.16.0.141 for event presence Jun 23 08:28:30 [2678] INFO:presence:send_notify_request: NOTIFY sip:msivara...@172.16.0.141 sip%3amsivara...@172.16.0.141 via sip:msivara...@172.16.1.125:7489;transport=udp on behalf of sip:rbox...@172.16.0.141 sip%3arbox...@172.16.0.141 for event presence Jun 23 08:28:30 [2678] INFO:presence:send_notify_request: NOTIFY sip:msivara...@172.16.0.141 sip%3amsivara...@172.16.0.141 via sip:msivara...@172.16.1.125:7489;transport=udp on behalf of sip:jabb2* xmpp1.smithmicro@sip-xmpp1.smithmicro.com for event presence Jun 23 08:28:30 [2680] INFO:presence:send_notify_request: NOTIFY sip:jabb2* xmpp1.smithmicro@sip-xmpp1.smithmicro.com via sip:172.16.0.141:5060 on behalf of sip:jabb2*xmpp1.smithmicro@sip-xmpp1.smithmicro.com for event presence.winfo Jun 23 08:28:30 [2679] WARNING:tm:t_unref: script writer didn't release transaction Jun 23 08:28:31 [2680] INFO:presence:send_notify_request: NOTIFY sip:msivara...@172.16.0.141 sip%3amsivara...@172.16.0.141 via sip:msivara...@172.16.1.125:7489;transport=udp on behalf of sip:jabb1* xmpp1.smithmicro@sip-xmpp1.smithmicro.com for event presence Jun 23 08:28:31 [2681] INFO:presence:send_notify_request: NOTIFY sip:jabb1* xmpp1.smithmicro@sip-xmpp1.smithmicro.com via
Re: [OpenSIPS-Users] DIALOG not deleted on BYE
Hi Bogdan, Again, thanks a lot for your help. The loose_route() seems to cause the problem, but somehow its needed to pass byes correctly to the UA. So I need to work a little on my skript. I will try the 1.6 ASAP and let you know the result. BR Uwe Bogdan-Andrei Iancu schrieb: If you could test, a fix is available on 1.6 (trunk) version - if ok, I will do the backport. Regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Thanks for the traces. Looking at the opensips logs, I say you do loose_route() twice for the ACK which looks twice for the dialog and increase the ref twice for the dialogthis is why the ref never gets back to 0 to allow the dialog to be destroyed.. Could you confirm this for me ? even if it's a script error , the dialog module should cope with it..I will look for a fix. Thanks and regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Uwe Kastens wrote: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. yes, it sounds like. Unfort I was not able to find out what the states and the events means. You can find the meaning of each state in: modules/dialog/dlg_hash.h So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? I can try : ) could you (privately if needed) please send me the the full logs for the entire call (debug=6) - for the non working part. Thanks and regards, Bogdan BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS boot Camp
Hello Bogdan, say 2 months and you can study and run the seminars by yourself. Included, you will have the possibility to fire questions to the teachers if you have something to clarify or if you got stuck with the labs What do you think of such approach ? I would love that, since one would be able to learn on weekend or after work. For some of us its not that easy to take a week off. No question that a personal training is better. I would take 2 Accounts if you will offer it. BR Uwe BTW: What about a opensips user meeting? I would prefer North of europe:) -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DIALOG not deleted on BYE
OK - with the fix from SVN you should be able to call loose_route() as many times you want without any risk - just let me know if it works as expected. Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, Again, thanks a lot for your help. The loose_route() seems to cause the problem, but somehow its needed to pass byes correctly to the UA. So I need to work a little on my skript. I will try the 1.6 ASAP and let you know the result. BR Uwe Bogdan-Andrei Iancu schrieb: If you could test, a fix is available on 1.6 (trunk) version - if ok, I will do the backport. Regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Thanks for the traces. Looking at the opensips logs, I say you do loose_route() twice for the ACK which looks twice for the dialog and increase the ref twice for the dialogthis is why the ref never gets back to 0 to allow the dialog to be destroyed.. Could you confirm this for me ? even if it's a script error , the dialog module should cope with it..I will look for a fix. Thanks and regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Uwe Kastens wrote: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. yes, it sounds like. Unfort I was not able to find out what the states and the events means. You can find the meaning of each state in: modules/dialog/dlg_hash.h So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? I can try : ) could you (privately if needed) please send me the the full logs for the entire call (debug=6) - for the non working part. Thanks and regards, Bogdan BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list
Re: [OpenSIPS-Users] OpenSIPS boot Camp
Uwe Kastens wrote: Hello Bogdan, say 2 months and you can study and run the seminars by yourself. Included, you will have the possibility to fire questions to the teachers if you have something to clarify or if you got stuck with the labs What do you think of such approach ? I would love that, since one would be able to learn on weekend or after work. For some of us its not that easy to take a week off. No question that a personal training is better. I would take 2 Accounts if you will offer it. OK, perfect - I will discuss with Flavio to evaluate such a model and to start putting it in place! Thanks for the feedback, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS user meeting ?
I would love something like this :) Any other people willing to go for something like this ? Regards, Bogdan Uwe Kastens wrote: Hello Bogdan, BTW: What about a opensips user meeting? I would prefer North of europe:) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] 1.6 and mysql = crash Re: DIALOG not deleted on BYE
Bogdan, Sorry for bothering again. I tried the latest trunk from svn and opensips is dying after accessing the mysql db. I will attach the trace. BR Uwe Bogdan-Andrei Iancu schrieb: OK - with the fix from SVN you should be able to call loose_route() as many times you want without any risk - just let me know if it works as expected. Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, Again, thanks a lot for your help. The loose_route() seems to cause the problem, but somehow its needed to pass byes correctly to the UA. So I need to work a little on my skript. I will try the 1.6 ASAP and let you know the result. BR Uwe Bogdan-Andrei Iancu schrieb: If you could test, a fix is available on 1.6 (trunk) version - if ok, I will do the backport. Regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Thanks for the traces. Looking at the opensips logs, I say you do loose_route() twice for the ACK which looks twice for the dialog and increase the ref twice for the dialogthis is why the ref never gets back to 0 to allow the dialog to be destroyed.. Could you confirm this for me ? even if it's a script error , the dialog module should cope with it..I will look for a fix. Thanks and regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Uwe Kastens wrote: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. yes, it sounds like. Unfort I was not able to find out what the states and the events means. You can find the meaning of each state in: modules/dialog/dlg_hash.h So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? I can try : ) could you (privately if needed) please send me the the full logs for the entire call (debug=6) - for the non working part. Thanks and regards, Bogdan BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org
Re: [OpenSIPS-Users] SOLVED: Re: OpenSIPS 1.5.1 with Postgres
BTW, why the desire to use POSTGRES rather than PGSQL ? GTG ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS user meeting ?
That is a brilliant idea, specially for people like me who won't be able to travel to America or Europe to attend the boot camp, because of time and financial constraints. May I also suggest that some sort of online certification be offered for those who had taken the course. I just hope this would be available on the soonest possible time. In regards to my former query, is there any available copy of the DVD-disc(the one distributed on the seminar) online for public download. It will make studying opensips easier because I know that those configurations are working and that will serve as a reference of what I have configured. I have been studying opensips for several months and its very difficult to find sample working configurations online that I can compare with my own configuration. The only trusted reference I am using is the OpenSER book of Flavio E. Goncalves but some of the codes are obsolete or has been replaced. Bogdan-Andrei Iancu wrote: I would love something like this :) Any other people willing to go for something like this ? Regards, Bogdan Uwe Kastens wrote: Hello Bogdan, BTW: What about a opensips user meeting? I would prefer North of europe:) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- View this message in context: http://n2.nabble.com/OpenSIPS-boot-Camp-tp3171851p3177984.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Registration and Loose-Route
I am new and need an explanation. I have installed opensips 1.5 with
database support. I am trying to authenticate via the subscriber's
table. Utilizing the sample config file and uncommenting the areas to
allow authentication via database, I try to register a sip device. I
have added a user using opensipsctl. When the registration requests
comes in, it dies in the loose_route() function with the error 403
Preload Route Denied. According to the documentation on the
loose_route() function, if there is no to-tag and there is only on route
header indicating the localproxy, the function should return false. It
is returning true. I then added the sip domain to the domain table and
the error changes to 401Unauthorized. Please explain. I am including
the SIP message and the debug output.
Jun 29 01:15:03 [15473] DBG:core:parse_msg: SIP Request:
Jun 29 01:15:03 [15473] DBG:core:parse_msg: method: REGISTER
Jun 29 01:15:03 [15473] DBG:core:parse_msg: uri:
sip:kwesi.chicagosip1.akan.us.com
Jun 29 01:15:03 [15473] DBG:core:parse_msg: version: SIP/2.0
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=2
Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type 232,
branch = z9hG4bK728627284; state=6
Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type 235,
rport = n/a; state=17
Jun 29 01:15:03 [15473] DBG:core:parse_via: end of header reached, state=5
Jun 29 01:15:03 [15473] DBG:core:parse_headers: via found, flags=2
Jun 29 01:15:03 [15473] DBG:core:parse_headers: this is the first via
Jun 29 01:15:03 [15473] DBG:core:receive_msg: After parse_msg...
Jun 29 01:15:03 [15473] DBG:core:receive_msg: preparing to run routing
scripts...
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=100
Jun 29 01:15:03 [15473] DBG:core:parse_to: end of header reached, state=10
Jun 29 01:15:03 [15473] DBG:core:parse_to: display={},
ruri={sip:3124530...@kwesi.chicagosip1.akan.us.com}
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: To [48];
uri=[sip:3124530...@kwesi.chicagosip1.akan.us.com]
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: to body
[sip:3124530...@kwesi.chicagosip1.akan.us.com
]
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: cseq CSeq: 6493
REGISTER
Jun 29 01:15:03 [15473] DBG:maxfwd:is_maxfwd_present: value = 70
Starting to process request
Jun 29 01:15:03 [15473] DBG:uri:has_totag: no totag
we are about to check for cancel
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=78
Jun 29 01:15:03 [15473] DBG:tm:t_lookup_request: start searching:
hash=15692, isACK=0
Jun 29 01:15:03 [15473] DBG:tm:matching_3261: RFC3261 transaction
matching failed
Jun 29 01:15:03 [15473] DBG:tm:t_lookup_request: no transaction found
we are about to check registration and multidomain
we are about to check for loose route
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=200
Jun 29 01:15:03 [15473] DBG:rr:is_preloaded: is_preloaded: Yes
Jun 29 01:15:03 [15473] DBG:core:grep_sock_info: checking if host==us:
29==14 [kwesi.chicagosip1.akan.us.com] == [209.252.110.37]
Jun 29 01:15:03 [15473] DBG:core:grep_sock_info: checking if port 5060
matches port 5060
Jun 29 01:15:03 [15473] DBG:core:check_self: host != me
Jun 29 01:15:03 [15473] DBG:rr:after_loose: Topmost URI is NOT myself
Jun 29 01:15:03 [15473] DBG:rr:after_loose: URI to be processed:
'sip:kwesi.chicagosip1.akan.us.com:5060;lr'
Jun 29 01:15:03 [15473] DBG:rr:after_loose: Next URI is a loose router
Jun 29 01:15:03 [15473] DBG:core:parse_to_param: tag=1590215359
Jun 29 01:15:03 [15473] DBG:core:parse_to: end of header reached, state=29
Jun 29 01:15:03 [15473] DBG:core:parse_to: display={},
ruri={sip:3124530...@kwesi.chicagosip1.akan.us.com}
Attempt to route with preloaded Route's
[sip:3124530...@kwesi.chicagosip1.akan.us.com/sip:3124530...@kwesi.chicagosip1.akan.us.com/sip:kwesi.chicagosip1.akan.us.com/1069016662-606...@98.122.86.123]jun
29
01:15:03 [15473] DBG:core:parse_headers: flags=
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: content_length=0
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: found end of header
Jun 29 01:15:03 [15473] DBG:core:check_via_address: params
98.122.86.123, 98.122.86.123, 0
Jun 29 01:15:03 [15473] DBG:core:destroy_avp_list: destroying list 0
Jun 29 01:15:03 [15473] DBG:core:receive_msg: cleaning up
#
U 2009/06/29 01:35:01.608581 98.122.86.123:6062 - 209.252.110.37:5060
REGISTER sip:kwesi.chicagosip1.akan.us.com SIP/2.0.
Via: SIP/2.0/UDP 98.122.86.123:6062;branch=z9hG4bK1362945809;rport.
Route: sip:kwesi.chicagosip1.akan.us.com:5060;lr.
From: sip:3124530...@kwesi.chicagosip1.akan.us.com;tag=1590215359.
To: sip:3124530...@kwesi.chicagosip1.akan.us.com.
Call-ID: 1069016662-606...@98.122.86.123.
CSeq: 6494 REGISTER.
Contact:
sip:3124530...@98.122.86.123:6062;reg-id=2;+sip.instance=urn:uuid:--1000-8000-000B821473A2.
Max-Forwards: 70.
User-Agent: Grandstream GXW-4004 V1.3A 1.0.1.15.
Supported: path.
Re: [OpenSIPS-Users] LDAP Authentication
Bogdan, I'm not an LDAP expert either, but I will try to explain the scenario better. As you said, the LDAP bind is static - done once in the beginning and sourced from the ldap.cfg file. Unfortunately, we have a filter on our LDAP server that prevents ordinary users from seeing the password field in the LDAP entry. The way we verify authentication in our environment is by dynamically substituting the LDAP bind DN with the client's uid (and password) and making a simple LDAP query using that uid. If that bind is successful, then we know that the password is correct. It doesn't seem like there is anyway to configure opensips in that manner. The aim, with LDAP, was to have a single-signon environment for our LAN and SIP accounts. This doesn't seem possible, unless you or anyone else on the list has any further suggestions. We could use kerberos/AD authentication from the client if that is a possibility. Regards, Alan Rubin -Original Message- From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Sent: Monday, 29 June 2009 10:13 PM To: Alan Rubin Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] LDAP Authentication Hi Alan, I'm not an LDAP expert to get into details about how ldap should be configured or soWhat I can tell is that the bind is static (only once done at the beginning at that's it)Can you send me a link or something to read more about what this dynamic bind means in LDAP ? Thanks and regards, Bogdan Alan Rubin wrote: Bogdan, Apparently the email administrator had a regex on the SMTP gateway to reject messages with pass (and) word (combined) because of previous users succumbing to phishing exercises. It may work now, but I will continue to check the archives. Oh well. Regarding: Now, going to the actual issue, the problem is related to password - about how the client and server (ldap) are keeping the password - do they both keep it same format (like plain text) ? Regards, Bogdan I think I've figured out the issue, although I don't believe there is a solution. Hopefully you can verify, either way. The bind user in the ldap.cfg file does not have the privilege to retrieve the pass word field from our LDAP directory. The only way our LDAP setup is supposed to work is by binding using the user-to-be-authenticated directly with the LDAP directory server. It is my understanding, and this is where you can verify or correct me, that opensips and the LDAP module can not change the bind user dynamically. Regards, Alan Rubin ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Registration and Loose-Route
Are you sure that the message are been processed by a register block of your
configuration? can you attach your configuration file? you can use log
function in the differents blocks of your configuration , in order to
clarify your debug.
best regards.
Pancho.
On Mon, Jun 29, 2009 at 9:06 PM, Nathaniel L Keeling
keel...@akan-tech.comwrote:
I am new and need an explanation. I have installed opensips 1.5 with
database support. I am trying to authenticate via the subscriber's
table. Utilizing the sample config file and uncommenting the areas to
allow authentication via database, I try to register a sip device. I
have added a user using opensipsctl. When the registration requests
comes in, it dies in the loose_route() function with the error 403
Preload Route Denied. According to the documentation on the
loose_route() function, if there is no to-tag and there is only on route
header indicating the localproxy, the function should return false. It
is returning true. I then added the sip domain to the domain table and
the error changes to 401Unauthorized. Please explain. I am including
the SIP message and the debug output.
Jun 29 01:15:03 [15473] DBG:core:parse_msg: SIP Request:
Jun 29 01:15:03 [15473] DBG:core:parse_msg: method: REGISTER
Jun 29 01:15:03 [15473] DBG:core:parse_msg: uri:
sip:kwesi.chicagosip1.akan.us.com
Jun 29 01:15:03 [15473] DBG:core:parse_msg: version: SIP/2.0
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=2
Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type 232,
branch = z9hG4bK728627284; state=6
Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type 235,
rport = n/a; state=17
Jun 29 01:15:03 [15473] DBG:core:parse_via: end of header reached, state=5
Jun 29 01:15:03 [15473] DBG:core:parse_headers: via found, flags=2
Jun 29 01:15:03 [15473] DBG:core:parse_headers: this is the first via
Jun 29 01:15:03 [15473] DBG:core:receive_msg: After parse_msg...
Jun 29 01:15:03 [15473] DBG:core:receive_msg: preparing to run routing
scripts...
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=100
Jun 29 01:15:03 [15473] DBG:core:parse_to: end of header reached, state=10
Jun 29 01:15:03 [15473] DBG:core:parse_to: display={},
ruri={sip:3124530...@kwesi.chicagosip1.akan.us.comsip%3a3124530...@kwesi.chicagosip1.akan.us.com
}
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: To [48];
uri=[sip:3124530...@kwesi.chicagosip1.akan.us.comsip%3a3124530...@kwesi.chicagosip1.akan.us.com
]
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: to body
[sip:3124530...@kwesi.chicagosip1.akan.us.comsip%3a3124530...@kwesi.chicagosip1.akan.us.com
]
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: cseq CSeq: 6493
REGISTER
Jun 29 01:15:03 [15473] DBG:maxfwd:is_maxfwd_present: value = 70
Starting to process request
Jun 29 01:15:03 [15473] DBG:uri:has_totag: no totag
we are about to check for cancel
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=78
Jun 29 01:15:03 [15473] DBG:tm:t_lookup_request: start searching:
hash=15692, isACK=0
Jun 29 01:15:03 [15473] DBG:tm:matching_3261: RFC3261 transaction
matching failed
Jun 29 01:15:03 [15473] DBG:tm:t_lookup_request: no transaction found
we are about to check registration and multidomain
we are about to check for loose route
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=200
Jun 29 01:15:03 [15473] DBG:rr:is_preloaded: is_preloaded: Yes
Jun 29 01:15:03 [15473] DBG:core:grep_sock_info: checking if host==us:
29==14 [kwesi.chicagosip1.akan.us.com] == [209.252.110.37]
Jun 29 01:15:03 [15473] DBG:core:grep_sock_info: checking if port 5060
matches port 5060
Jun 29 01:15:03 [15473] DBG:core:check_self: host != me
Jun 29 01:15:03 [15473] DBG:rr:after_loose: Topmost URI is NOT myself
Jun 29 01:15:03 [15473] DBG:rr:after_loose: URI to be processed:
'sip:kwesi.chicagosip1.akan.us.com:5060;lr'
Jun 29 01:15:03 [15473] DBG:rr:after_loose: Next URI is a loose router
Jun 29 01:15:03 [15473] DBG:core:parse_to_param: tag=1590215359
Jun 29 01:15:03 [15473] DBG:core:parse_to: end of header reached, state=29
Jun 29 01:15:03 [15473] DBG:core:parse_to: display={},
ruri={sip:3124530...@kwesi.chicagosip1.akan.us.comsip%3a3124530...@kwesi.chicagosip1.akan.us.com
}
Attempt to route with preloaded Route's
[
sip:3124530...@kwesi.chicagosip1.akan.us.com/sip:3124530...@kwesi.chicagosip1.akan.us.com/sip:kwesi.chicagosip1.akan.us.com/1069016662-606...@98.122.86.123]jun
29
01:15:03 [15473] DBG:core:parse_headers: flags=
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: content_length=0
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: found end of header
Jun 29 01:15:03 [15473] DBG:core:check_via_address: params
98.122.86.123, 98.122.86.123, 0
Jun 29 01:15:03 [15473] DBG:core:destroy_avp_list: destroying list 0
Jun 29 01:15:03 [15473] DBG:core:receive_msg: cleaning up
#
U 2009/06/29 01:35:01.608581
Re: [OpenSIPS-Users] Registration and Loose-Route
If there is no entry in the domain table, the it will error in the
loose_route() function and the error message that I get is 403 Preload
Route denied. When I add an entry to the domain table, it passes the
loose_route() function and then error while authenticating. I have
placed an xlog statement within the register block of the config file
and right before the loose_route() function block is executed. I have
included my config file.
thanks
Nathaniel
Eduardo Panciera wrote:
Are you sure that the message are been processed by a register block
of your configuration? can you attach your configuration file? you can
use log function in the differents blocks of your configuration , in
order to clarify your debug.
best regards.
Pancho.
On Mon, Jun 29, 2009 at 9:06 PM, Nathaniel L Keeling
keel...@akan-tech.com mailto:keel...@akan-tech.com wrote:
I am new and need an explanation. I have installed opensips 1.5 with
database support. I am trying to authenticate via the subscriber's
table. Utilizing the sample config file and uncommenting the areas to
allow authentication via database, I try to register a sip device. I
have added a user using opensipsctl. When the registration requests
comes in, it dies in the loose_route() function with the error 403
Preload Route Denied. According to the documentation on the
loose_route() function, if there is no to-tag and there is only on
route
header indicating the localproxy, the function should return false. It
is returning true. I then added the sip domain to the domain table and
the error changes to 401Unauthorized. Please explain. I am including
the SIP message and the debug output.
Jun 29 01:15:03 [15473] DBG:core:parse_msg: SIP Request:
Jun 29 01:15:03 [15473] DBG:core:parse_msg: method: REGISTER
Jun 29 01:15:03 [15473] DBG:core:parse_msg: uri:
sip:kwesi.chicagosip1.akan.us.com
http://kwesi.chicagosip1.akan.us.com/
Jun 29 01:15:03 [15473] DBG:core:parse_msg: version: SIP/2.0
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=2
Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type
232,
branch = z9hG4bK728627284; state=6
Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type
235,
rport = n/a; state=17
Jun 29 01:15:03 [15473] DBG:core:parse_via: end of header reached,
state=5
Jun 29 01:15:03 [15473] DBG:core:parse_headers: via found, flags=2
Jun 29 01:15:03 [15473] DBG:core:parse_headers: this is the first via
Jun 29 01:15:03 [15473] DBG:core:receive_msg: After parse_msg...
Jun 29 01:15:03 [15473] DBG:core:receive_msg: preparing to run routing
scripts...
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=100
Jun 29 01:15:03 [15473] DBG:core:parse_to: end of header reached,
state=10
Jun 29 01:15:03 [15473] DBG:core:parse_to: display={},
ruri={sip:3124530...@kwesi.chicagosip1.akan.us.com
mailto:sip%3a3124530...@kwesi.chicagosip1.akan.us.com}
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: To [48];
uri=[sip:3124530...@kwesi.chicagosip1.akan.us.com
mailto:sip%3a3124530...@kwesi.chicagosip1.akan.us.com]
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: to body
[sip:3124530...@kwesi.chicagosip1.akan.us.com
mailto:sip%3a3124530...@kwesi.chicagosip1.akan.us.com
]
Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: cseq CSeq: 6493
REGISTER
Jun 29 01:15:03 [15473] DBG:maxfwd:is_maxfwd_present: value = 70
Starting to process request
Jun 29 01:15:03 [15473] DBG:uri:has_totag: no totag
we are about to check for cancel
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=78
Jun 29 01:15:03 [15473] DBG:tm:t_lookup_request: start searching:
hash=15692, isACK=0
Jun 29 01:15:03 [15473] DBG:tm:matching_3261: RFC3261 transaction
matching failed
Jun 29 01:15:03 [15473] DBG:tm:t_lookup_request: no transaction found
we are about to check registration and multidomain
we are about to check for loose route
Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=200
Jun 29 01:15:03 [15473] DBG:rr:is_preloaded: is_preloaded: Yes
Jun 29 01:15:03 [15473] DBG:core:grep_sock_info: checking if host==us:
29==14 [kwesi.chicagosip1.akan.us.com
http://kwesi.chicagosip1.akan.us.com/] == [209.252.110.37]
Jun 29 01:15:03 [15473] DBG:core:grep_sock_info: checking if port 5060
matches port 5060
Jun 29 01:15:03 [15473] DBG:core:check_self: host != me
Jun 29 01:15:03 [15473] DBG:rr:after_loose: Topmost URI is NOT myself
Jun 29 01:15:03 [15473] DBG:rr:after_loose: URI to be processed:
'sip:kwesi.chicagosip1.akan.us.com:5060;lr'
Jun 29 01:15:03 [15473] DBG:rr:after_loose: Next URI is a loose router
Jun 29 01:15:03 [15473] DBG:core:parse_to_param: tag=1590215359
Jun 29 01:15:03 [15473] DBG:core:parse_to: end of
