sr-users

sr-users@lists.kamailio.org

37 participants
37547 discussions

[Users] connect xmpp component to jabber xcp?
by Matt Winter 26 Apr '07

26 Apr '07

Hello, I'm having a problem connecting my openser xmpp module(in component mode) to a jabber xcp server. Has anyone done this? Here's my openser.cfg file. Thanks. # more /etc/openser/openser.cfg # # $Id: openser-xmpp.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script for XMPP GW # # make sure in your main SIP server that you send # only the adequate SIP MESSAGES to XMPP GW # # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 listen=udp:10.1.1.234:5060 alias=xmpp.jabbersrv.corp.com # ------------------ module loading ---------------------------------- mpath="/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # XMPP loadmodule "xmpp.so" modparam("xmpp", "domain_separator", "*") modparam("xmpp", "gateway_domain", "xmpp.jabbersrv.corp.com") modparam("xmpp", "xmpp_domain", "xmpp.jabbersrv.corp.com") modparam("xmpp", "xmpp_host", "jabbersrv.corp.com") modparam("xmpp", "backend", "component") modparam("xmpp", "xmpp_password", "BCD11234DC") # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo_xmpp") # -- usrloc params -- # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; ### absorb retransmissions ### if (!t_newtran()) { sl_reply_error(); return; } if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } log("*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; }

2 4

[Users] sending NOTIFY messages
by Mark Price 26 Apr '07

26 Apr '07

Hi, Are there any generally used methods for sending SIP NOTIFY messages to a phone that is registered through openser? I.e. for sending events to sip phones? For example, I imagine a script that would register to the mydomain.org openser server as an administrative user and send a NOTIFY packet to user(a)mydomain.org. I'm hoping not to reinvent the wheel here. Mark Price

2 3

[Serusers] t_replicate "REGISTER" causes retransmission?
by Chahn Kim 26 Apr '07

26 Apr '07

Hi, I am testing simple(vs clean, per Jiri) t_replicate(server 2) on "REGISTER" from server 1 to another ser server 2. Server 2 logs "ERROR:registrar:update_contacts: invalid cseq for aor...". ngrep shows that server 1 send "REGISTER" twice. May be the re-transmitted request causes the incalid cseq? Can anyone tell me how to fix this problem? John Chahn Kim

1 0

[Users] Strange behaviour
by Ronnie Flink 26 Apr '07

26 Apr '07

Hi, We have openser (currently 1.1.1) deployed in our iptel-solution and almost everything seems to work ok but one strange thing has come up. A user has a few Dlink DVG-2001S (behind a net-hide firewall) that tries to register but doesn't succeed and my guess is that openser sends the reply to the incorrect port and there firewall blocks this. A small ngrep-dump of the session: U 2007/04/24 11:40:12.620330 xx.xx.243.11:3710 -> xx.xxx.248.10:5060 REGISTER sip:sip.domain.tld SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:5060;branch=z9hG4bK8294fce7242e1605. Max-Forwards: 70. From: xx55551904 <sip:xx55551904@sip.domain.tld>;tag=1c2bcb8b48c89388. To: xx55551904 <sip:xx55551904@sip.domain.tld>. Call-ID: fec625fb04d0f004(a)192.168.0.164. CSeq: 1 REGISTER. Contact: *. Expires: 0. Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS, INFO. Content-Length: 0. . U 2007/04/24 11:40:12.620555 xx.xxx.248.10:5060 -> xx.xxx.243.11:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.0.164:5060;branch=z9hG4bK8294fce7242e1605;received=xx.xxx.243.11. From: xx55551904 <sip:xx55551904@sip.domain.tld>;tag=1c2bcb8b48c89388. To: xx55551904 <sip:xx55551904@sip.domain.tld>. Call-ID: fec625fb04d0f004(a)192.168.0.164. CSeq: 1 REGISTER. Server: OpenSer (1.1.1-tls (i386/linux)). Content-Length: 0. Warning: 392 xx.xxx.248.10:5060 "Noisy feedback tells: pid=23636 req_src_ip=xx.xxx.243.11 req_src_port=3710 in_uri=sip:sip.domain.tld out_uri=sip :sip.domain.tld via_cnt==1". Does anyone have a guess what might be causing openser to send the reply to the wrong port ? The openser-box has a live internet-ip and and isn't behind any type of firewall or nat-device. I think all client nat-stuff is correct since it seems to work with other customers that are behind net-hide firewalls (broadband-routers). /Thanks in advance - Ronnie Flink

2 1

RE: [Serusers] global contact list
by Barta János 26 Apr '07

26 Apr '07

Hello! Thanks for your answer again, and I'm sorry for the delayed answer, but I was out of office. My biggest problem is, that I don't know, what I should see :-(. I have an eyeBeam window, that contains the "Calls & Conntats" parameters. In this window I can see my (user) contact list, but I didn't see the global one. For example (the PC is my user contact list): -PC -105 -12345 -barta and it is working fine, but I can not see here (in this window) the global one: sip:~# cat /var/www/xcap-root/rls-services/global/index <?xml version="1.0" encoding="UTF-8"?> <rls-services> <service uri="sip:bartaj-list@test.intra"> <list name="Human resources"> <entry uri="sip:test1@test.intra"> <display-name>bartaj</display-name> </entry> <entry uri="sip:test2@test.intra"> <display-name>12345</display-name> </entry> </list> <packages> <package>presence</package> </packages> </service> <service uri="sip:hr2@test.intra"> <list name="Human resources"> <entry uri="sip:test1@test.intra"> <display-name>bartaj</display-name> </entry> <entry uri="sip:test2@test.intra"> <display-name>12345</display-name> </entry> </list> <packages> <package>presence</package> </packages> </service> </rls-services> I think in that case I should see the next tree: -PC -105 -12345 -barta -Human resources -bartaj -12345 As You asked, I made a capture between the SER and XCAP. It was a little difficult, because I didn't see interesting traffic between the SER and XCAP, but a few minutes later I realised, that the SER for the communication (in that case) uses the loopback (lo) interface. You can see here the capture, that I made on the lo interface: 1 0.000000 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [SYN] Seq=0 Len=0 MSS=16396 TSV=3808162073 TSER=0 WS=0 2 0.000056 127.0.0.1 -> 127.0.0.1 TCP www > 37512 [SYN, ACK] Seq=0 Ack=1 Win=32767 Len=0 MSS=16396 TSV=3808162073 TSER=3808162073 WS=0 3 0.000065 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=1 Ack=1 Win=32767 Len=0 TSV=3808162073 TSER=3808162073 4 0.000072 127.0.0.1 -> 127.0.0.1 HTTP GET /xcap/pres-rules/users/12345/presence-rules.xml HTTP/1.1 5 0.000079 127.0.0.1 -> 127.0.0.1 TCP www > 37512 [ACK] Seq=1 Ack=113 Win=32767 Len=0 TSV=3808162073 TSER=3808162073 6 0.000545 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK 7 0.000559 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=113 Ack=262 Win=32767 Len=0 TSV=3808162074 TSER=3808162074 8 0.000571 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP traffic 9 0.000578 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=113 Ack=870 Win=32767 Len=0 TSV=3808162074 TSER=3808162074 10 0.005535 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [SYN] Seq=0 Len=0 MSS=16396 TSV=3808162079 TSER=0 WS=0 11 0.005556 127.0.0.1 -> 127.0.0.1 TCP www > 37513 [SYN, ACK] Seq=0 Ack=1 Win=32767 Len=0 MSS=16396 TSV=3808162079 TSER=3808162079 WS=0 12 0.005565 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=1 Ack=1 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 13 0.005616 127.0.0.1 -> 127.0.0.1 HTTP GET /xcap/pres-rules/users/bartaj/presence-rules.xml HTTP/1.1 14 0.005623 127.0.0.1 -> 127.0.0.1 TCP www > 37513 [ACK] Seq=1 Ack=114 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 15 0.005989 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK 16 0.006000 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=114 Ack=262 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 17 0.006011 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP traffic 18 0.006017 127.0.0.1 -> 127.0.0.1 TCP 37513 > www [ACK] Seq=114 Ack=870 Win=32767 Len=0 TSV=3808162079 TSER=3808162079 19 2.979491 127.0.0.1 -> 127.0.0.1 HTTP GET /xcap/rls-services/global/index HTTP/1.1 20 2.979722 127.0.0.1 -> 127.0.0.1 HTTP HTTP/1.1 200 OK 21 2.979734 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=209 Ack=1126 Win=32767 Len=0 TSV=3808165053 TSER=3808165053 22 2.979745 127.0.0.1 -> 127.0.0.1 HTTP Continuation or non-HTTP traffic 23 2.979751 127.0.0.1 -> 127.0.0.1 TCP 37512 > www [ACK] Seq=209 Ack=1887 Win=32767 Len=0 TSV=3808165053 TSER=3808165053 You can see, that in the message number 19 the SER downloaded the index file. This file contains correct information: sip:~# wget 127.0.0.1/xcap/rls-services/global/index --17:22:43-- http://127.0.0.1/xcap/rls-services/global/index => `index' Connecting to 127.0.0.1:80... connected. HTTP request sent, awaiting response... 200 OK Length: 761 [text/plain] 100%[====================================>] 761 --.--K/s 17:22:43 (7.26 MB/s) - `index' saved [761/761] sip:~# sip:~# cat index <?xml version="1.0" encoding="UTF-8"?> <rls-services> <service uri="sip:bartaj-list@test.intra"> <list name="Human resources"> <entry uri="sip:test1@test.intra"> <display-name>bartaj</display-name> </entry> <entry uri="sip:test2@test.intra"> <display-name>12345</display-name> </entry> </list> <packages> <package>presence</package> </packages> </service> <service uri="sip:hr2@test.intra"> <list name="Human resources"> <entry uri="sip:test1@test.intra"> <display-name>bartaj</display-name> </entry> <entry uri="sip:test2@test.intra"> <display-name>12345</display-name> </entry> </list> <packages> <package>presence</package> </packages> </service> </rls-services> sip:~# And at last, how can I add query_rls_services and query_resource_list log informations into the config file? Thx, Jani -----Original Message----- From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] Sent: Tuesday, April 03, 2007 11:18 AM To: Barta János Cc: serusers(a)iptel.org Subject: Re: [Serusers] global contact list Hi, this part of config file is not sufficient for me, but I guess that RLS related things stayed as in the original config you mentioned... In which way it doesn't work? (For example it returns non OK responses?) Please could you try to capture HTTP traffic between XCAP server and SER? (If SER really queries the right resource list document.) You can add more log information around important config parts (query_rls_services and query_resource_list functions are most important here). BTW the second message you sent is not subscription to resource list or user, it is subscription to event package 'sip-profile' (content of Event header field) which is not supported by SER. It is possible to use registration status as source for presence - in this case you have to have PA module on your server handling registrations and PA module parameter 'use_callbacks' must be set to 1. Vaclav <br><br>_____________________________________________________________________<br> Tudod már, hol nyaralsz? Úti beszámolókat, utazási ajánlatokat találsz a <a href="http://ad.adverticum.net/b/cl,1,6022,159388,223268/click.prm" target="new">travelline.hu-n!</a><br> Ha Te is megírod élményeidet, megnyerhetsz egy párizsi utat vagy egy kalandtúrát az Alpokba!

1 0

[Serusers] REGISTERED(TRANSACTION) BUT NO DIALOG
by Adeyeye Michael 25 Apr '07

25 Apr '07

Hi, My clients get registered but no call is successfully transfered when I used UCTIMSCLIENT or Linphone. Besides I can't get any output or trace to know what is wrong with SER. micadeyeye@micadeyeye:/opt/OpenIMSCore$ sudo ngrep port 5060 -d eth0 Password: interface: eth0 (10.128.0.0/255.255.0.0) filter: (ip or ip6) and ( port 5060 ) PLS HELP. MIKE

1 0

[Serusers] No trace!!!
by Adeyeye Michael 25 Apr '07

25 Apr '07

Can anyone help out? I can see any output or trace. micadeyeye@micadeyeye:/opt/OpenIMSCore$ sudo ngrep port 5060 -d eth0 Password: interface: eth0 (10.128.0.0/255.255.0.0) filter: (ip or ip6) and ( port 5060 ) MIke.

1 0

[Users] WeSIP setup problem
by ventilo35＠free.fr 25 Apr '07

25 Apr '07

Hello I'm having problems running OpenSER with WeSIP AS. I have installed OpenSER 1.2 stable release, and also tried the latest version using SVN, and I use WeSIP 0.1.4. The same problem always arises: I get either "no route" if a servlet was invoked, or "481" if no servlet was run. In both cases, it seems WeSIP is not able to send the INVITE back to OpenSER. Here are the traces: <First trace: a servlet named neos is run; it just proxies the message> SeasMessageEvent [SeasRouter] - New message received, from 192.168.1.17:5060 to 192.168.1.17:5060 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 45959,label=642523441 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 192.168.1.17/5060 ExpressMessageChannel [ExpressMChannel[0]] - About to process INVITE sip:5002@192.168.1.11 SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned ExpressMessageChannel [ExpressMChannel[0]] - Done processing INVITE sip:5002@192.168.1.11 SIP/2.0 SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:5002@192.168.1.11 SIP/2.0 Call-ID: 20011e80-296f38df-c0a814d5(a)192.168.1.11 From: "5001" <sip:5001@192.168.1.11>;tag=10005f00-296f38e8 To: <sip:5002@192.168.1.11> CSeq: 100 INVITE Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc0a814d51000bf80296f38e9 Contact: "5001" <sip:5001@192.168.1.17:5060> Max-Forwards: 69 User-Agent: LR SIP Phone Allow: INFO,UPDATE,NOTIFY,REFER Content-Type: application/sdp Content-Length: 406 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:45959 label:642523441 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bKc0a814d51000bf80296f38e9] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 192.168.1.11 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, trying match against Rules ConvergedHost [SipProcessor[4]] - Mapped request to:/neos ConvergedContextValve [SipProcessor[4]] - new session created :20011e80-296f38df-c0a814d5@192.168.1.11 doInvite within neos servlet: requestURI is sip:5002@192.168.1.11 StandardProxy [SipProcessor[4]] - sip:5002@192.168.1.11 Added to proxying targets. StandardProxy [SipProcessor[4]] - doProxy: proxying to sip:5002@192.168.1.11 StandardProxy [SipProcessor[4]] - ========== Request Sent ============== INVITE sip:5002@192.168.1.11 SIP/2.0 Call-ID: 20011e80-296f38df-c0a814d5(a)192.168.1.11 From: "5001" <sip:5001@192.168.1.11>;tag=10005f00-296f38e8 To: <sip:5002@192.168.1.11> CSeq: 100 INVITE Via: SIP/2.0/UDP 192.168.1.11:5060,SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc0a814d51000bf80296f38e9 Contact: "5001" <sip:5001@192.168.1.17:5060> Max-Forwards: 68 User-Agent: LR SIP Phone Allow: INFO,UPDATE,NOTIFY,REFER Content-Type: application/sdp Content-Length: 406 StandardProxy [SipProcessor[4]] - javax.sip.TransactionUnavailableException: no route! SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] <Second trace: the servlet is deactivated> SeasMessageEvent [SeasRouter] - New message received, from 192.168.1.17:5060 to 192.168.1.17:5060 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 32078,label=1509443196 flags=0 ExpressMessageChannel [ExpressMChannel[2]] - peerAddress = 192.168.1.17/5060 ExpressMessageChannel [ExpressMChannel[2]] - About to process INVITE sip:5002@192.168.1.11 SIP/2.0 SipConnector [ExpressMChannel[2]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[2]] - An incoming message is being assigned ExpressMessageChannel [ExpressMChannel[2]] - Done processing INVITE sip:5002@192.168.1.11 SIP/2.0 SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:5002@192.168.1.11 SIP/2.0 Call-ID: 20011b80-29725a1f-c0a814d5(a)192.168.1.11 From: "5001" <sip:5001@192.168.1.11>;tag=10005c00-29725a28 To: <sip:5002@192.168.1.11> CSeq: 100 INVITE Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc0a814d51000bf8029725a29 Contact: "5001" <sip:5001@192.168.1.17:5060> Max-Forwards: 69 User-Agent: LR SIP Phone Allow: INFO,UPDATE,NOTIFY,REFER Content-Type: application/sdp Content-Length: 408 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:32078 label:1509443196 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bKc0a814d51000bf8029725a29] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 192.168.1.11 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, trying match against Rules ConvergedHost [SipProcessor[4]] - Mapped request to:/neos ConvergedHostValve [SipProcessor[4]] - No servlet-mapping matched this Request. SipResponse [SipProcessor[4]] - >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 481 Call Leg/Transaction Does Not Exist Call-ID: 20011b80-29725a1f-c0a814d5(a)192.168.1.11 From: "5001" <sip:5001@192.168.1.11>;tag=10005c00-29725a28 To: <sip:5002@192.168.1.11> CSeq: 100 INVITE Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc0a814d51000bf8029725a29 Max-Forwards: 69 Content-Length: 0 I have thoroughly reviewed my configuration against the rules explained in the WeSIP website, everything seems in order (address OpenSER receives on is 192.168.1.11:5060, address for SEAS and WeSIP is 127.0.0.1:5080). Could you please tell me if this is something you already encountered and solved ? Thanks in advance Francois

2 1

[Users] XCAP Lite 0.9.1 released
by Adrian Georgescu 25 Apr '07

25 Apr '07

Change log xcap-lite (0.9.1) unstable; urgency=low * Added commit/rollback capability (use MDB2 database driver instead of DB) * Added inserted_time in watcher sync (diff provided by Klaus Darilion) The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-0.9.1.tar.gz Regards, Adrian

1 0

[Serusers] How to handle Cisco Phone registrations withouth DNS SRV records
by Mahatma Sarasvati 25 Apr '07

25 Apr '07

Hello All, I want to be able to handle registrations from Cisco Phones on the public internet to my SER server. I don't run my own public DNS servers. I use register.com for DNS and they don't support DNS SRV records as far as I know. So, my Cisco phone config has to have "voice.mydomain.com" in order for the REGISTER message to reach the SER server. However, I want my user accounts to be "jsmith(a)mydomain.com." The Cisco phone config is limiting because it appends the name of the proxy to the registration. Instead of 'username(a)mydomain.com' its trying to register with 'username(a)voice.mydomain.com' How can this be handled in the SER config? I use DNS SRV records on the LAN and it works great. What is the best method to handle this scenario? Thanks, Mahatma __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users