sr-users January 2006

sr-users@lists.kamailio.org

134 participants
258 discussions

Start a nNew thread

[Serusers] windows messenger 5.1 and IM
by Yi Zheng 01 Feb '06

01 Feb '06

Note: forwarded message attached.

2 1

[Serusers] SER and Cisco ATA 186
by R2 01 Feb '06

01 Feb '06

Hi all, How I can supress of uri user=phone ? This ATA receive call from PSTN, but, when I send calls to PSTN, i don't obtain response. The integration to PSTN is on Asterisk. Thanks R2

2 1

[Serusers] Help on SER with Mediaproxy
by Muhammad Asif Ali 01 Feb '06

01 Feb '06

Hi All I have a scenario in which I want to deploy SER with mediaproxy & allow everyone/anyone connecting with my SER to talk to each other. I am using ONSIP gettingStarted document. In order to allow everyone to connect to my SER without authentication I commented the following portion in ser.cfg: in Route[2] #if (!www_authorize("","subscriber")) { # www_challenge("","0"); # break; #}; #if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # break; #}; # consume_credentials(); and in Route[3] # if (!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # break; #} else if (!check_from()) { # sl_send_reply("403", "Use From=ID"); # break; #}; #consume_credentials(); Now when I try to connect 2 UA (Simpleopal) from behind a same NAT, they do get registered, & records also appear in location table of DataBase. But when I place call from UA1 to UA2 like "call sip:UA2@aa.bb.cc.dd", where aa.bb.cc.dd is the public IP of my SER machine (mediaproxy is also running on same machine), it ends up showing transport error. On SER side: A) SER logs say: on lookup() "UA2 not found in usrloc" and on is_local() "Realm 'ab.cc.dd.ee' is not local". the is_lcoal appears again saying "Realm 192.168.0.241 is not local" where 192.168.0.241 is UA2's local IP. B)Mediaproxy log says: command request c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1 203.82.51.9:5000:audio 203.82.51.9 aa.bb.cc.dd remote 192.168.0.241 remote OPAL/2.0 info= from:UA1@aa.bb.cc.dd,to:UA2@aa.bb.cc.dd ,fromtag:b09168c6-fbf4-1810-80b4-0050badb4a03,totag: session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: started. listening on aa.bb.cc.dd:35000 command execution time: 9.35 ms command request c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1 203.82.51.9:5000:audio 203.82.51.9 aa.bb.cc.dd remote 192.168.0.241 remote OPAL/2.0 info= from:UA1@aa.bb.cc.dd,to:UA2@aa.bb.cc.dd ,fromtag:b09168c6-fbf4-1810-80b4-0050badb4a03,totag: command execution time: 0.60 ms session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: ended (did timeout). (203.82.51.9) is my LAN's public IP aa.bb.cc.dd is the public IP of SER machine C) using ngrep I found out that it sends an "SIP/2.0 408 Request Timeout" cant figure out whats going on, anybody having any idea please help. Thanx in advance Regards Muhammad Asif Ali

2 4

[Serusers] any info reg adding DTLS using OpenSSL
by Pjothi 01 Feb '06

01 Feb '06

Dear all, I am a student and am trying to setup a demonstration with TLS and DTLS support between a SIP client and a Proxy. Has anyone some information regarding adding DTLS support for a SIP client/Proxy or a more generic one using OpenSSL. I just need it for demonstration purposes and so error handling's if any need not be complete. Just a simple support and a demonstration of both would do fine. If anyone has any information regarding this, may be a how to start or any introductory article would be of great help, kindly let me know. This would be of great help. Any suggestions regarding the same are most welcome. kindly let me know and thank you very much. regards, Pjothi

2 1

[Serusers] STUN and symmetric nat again
by Istvan De 01 Feb '06

01 Feb '06

Hi! I've read about the above topic in the arcives and now I've got the same problem. User A is behind a symmetric NAT -> a new port is bound for every different (external ip, external port) pair. User B has a connection to the net without any NAT or firewall. User A wants to have a SIP+RTP session with user B. How? I guess basically what is needed is forming of a SIP message like "send the your reply to the same ip and same port from where you have received this" and user A can send this to user B. For the RTP session, a pretty similar thing would be needed like, A says to B: "I send you the RTP stream on your x port, send your rtp stream back to the same ip and port from where you receive mine". How to setup such a SIP packet? Or maybe there is a standard solution for this already? (STUN wont help in this case I guess, and TURN is not an option for me.) I've read about "symmetric SIP + symmetric RTP" maybe that is the same as I've described above? Thanks Istvan

3 2

[Serusers] adress book
by Joao Pereira 01 Feb '06

01 Feb '06

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

4 4

Re: [Serusers] Monitoring
by hgaillac-sip＠yahoo.fr 01 Feb '06

01 Feb '06

Do you Know HSRP from Cisco System ? http://www.cisco.com/univercd/cc/td/doc/cisintwk/ics/cs009.htm Harry --- Hendrik Scholz <hendrik.scholz(a)freenet-ag.de> a écrit : > Hi! > > hgaillac-sip(a)yahoo.fr wrote: > > What about VRRP for failover ? > > > > SER1 (master) > > sip agents == || === Asterisk Farm==sip/pstn > gateway > > SER2 (slave) > > What do you mean by that? > > Somewhat confused, > Hendrik > > -- > freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 > Kiel, Germany > Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 > Internet: http://www.freenet.de, eMail: > hendrik.scholz(a)freenet-ag.de > ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

2 1

[Serusers] windows messenger 5.1 and IM
by Yi Zheng 31 Jan '06

31 Jan '06

I have two window messenger 5.1 clients on the same subnet as the SER server w/ pa module. They subscribe/notify to each other ok. HOwever they cannot IM each other. The callee, after receiveing INVITE, sent back a 100 trying, then 200 OK, and then immediately BYE. I searched thru archives and looks like this is a known issue with wm. What is the solution or workaround to this? Here is the messages I captured: U 10.1.104.251:1172 -> 10.1.104.254:5060 INVITE sip:104pc1@10.1.104.254 SIP/2.0..Via: SIP/2.0/UDP 10.1.104.251:8053 ..Max-Forwards: 70..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>; tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1 .104.254>..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Cont act: <sip:10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Type: applicati on/sdp..Content-Length: 119....v=0..o=- 0 0 IN IP4 10.1.104.251..s=session ..c=IN IP4 10.1.104.251..t=0 0..m=message 5060 sip sip:104pc2@10.1.104.254 .. # U 10.1.104.254:5060 -> 10.1.104.251:8053 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.1. 104.251:8053..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>;tag=d4 6c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1.104.2 54>..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Server: Si p EXpress router (0.10.99-dev30-tm-timers-pa-3 (i386/linux))..Content-Leng th: 0..Warning: 392 10.1.104.254:5060 "Noisy feedback tells: pid=23866 re q_src_ip=10.1.104.251 req_src_port=1172 in_uri=sip:104pc1@10.1.104.254 out _uri=sip:10.1.104.248:13018 via_cnt==1".... # U 10.1.104.254:5060 -> 10.1.104.248:13018 INVITE sip:10.1.104.248:13018 SIP/2.0..Record-Route: <sip:10.1.104.254;fta g=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on>..Via: SIP/2.0/UDP 10.1.104.254;b ranch=z9hG4bK2216.fea17c01.0..Via: SIP/2.0/UDP 10.1.104.251:8053..Max-Forw ards: 16..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>;tag=d46c61 cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1.104.254>. .Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Contact: <sip: 10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Type: application/sdp..Co ntent-Length: 119..P-hint: usrloc applied....v=0..o=- 0 0 IN IP4 10.1.104. 251..s=session..c=IN IP4 10.1.104.251..t=0 0..m=message 5060 sip sip:104pc 2(a)10.1.104.254.. # U 10.1.104.248:1334 -> 10.1.104.254:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.1.104.254;branch=z9hG4bK2216.fea17 c01.0..Via: SIP/2.0/UDP 10.1.104.251:8053..From: "104pc2(a)10.1.104.254" <si p:104pc2@10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d136 2..To: <sip:104pc1@10.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..Cal l-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..User-Agent: RTC/1. 3..Content-Length: 0.... # U 10.1.104.248:1334 -> 10.1.104.254:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.1.104.254;branch=z9hG4bK2216.fea17c01. 0..Via: SIP/2.0/UDP 10.1.104.251:8053..From: "104pc2(a)10.1.104.254" <sip:10 ;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..T">4pc2(a)10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..T o: <sip:104pc1@10.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..Call-ID : 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Record-Route: <sip:10. 1.104.254;ftag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on>..Contact: <sip:10.1 .104.248:13018>..User-Agent: RTC/1.3..Content-Type: application/sdp..Conte nt-Length: 119....v=0..o=- 0 0 IN IP4 10.1.104.248..s=session..c=IN IP4 10 .1.104.248..t=0 0..m=message 5060 sip sip:104pc1@10.1.104.254.. # U 10.1.104.248:1334 -> 10.1.104.254:5060 BYE sip:10.1.104.254;ftag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on SIP/2.0.. Via: SIP/2.0/UDP 10.1.104.248:13018..Max-Forwards: 70..From: <sip:104pc1@1 0.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..To: "104pc2(a)10.1.104.25 4" <sip:104pc2@10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=692 00d1362..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 BYE..Route: <s ip:10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Length: 0.... ##

1 0

[Users] Parsing SIP responses
by Ovidiu Sas 31 Jan '06

31 Jan '06

Hi, I configured openser to handle redirect responses and re-route calls based on the Contact header from the 3xx response using the uac_redirect module. In addition to that, I would like to transfer some headers from the 3xx response into the redirected INVITE (see diagram): UAC openser UAS1 UAS2 | | | | |---INVITE--->| | | | |---INVITE--->| | | |<--3XX-------| | | | with Diversion header | | |---ACK------>| | | |---------------INVITE------->| | | | with Diversion header | | | from 3xx response from UAS1 Is there an API that will allow me to search for headers inside a response and save them in to an AVP? Thx, -ovi

2 3

Re: [Serusers] no SRV query after rewriting R-URI?
by alan chiu 31 Jan '06

31 Jan '06

great, thanks for the clarification...so simply, rewrite("host") would trigger SRV lookup. --- Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > Hi Alan! > > That's the correct behaviour. Please read RFC 3263. > If the URI contains a > port, e.g. inside.acme.com:5060 there must not be an > SRV lookup. Because > the SRV lookup leads to a port, which is already > defined. > > regards > klaus > > On Tue, January 31, 2006 20:28, alan chiu said: > > Server: Sip EXpress router (0.8.12 > (i86pc/solaris)) > > > > when endpoint sends 18005551212(a)sip.acme.com, SER > > (without rewriting of R-URI, looks it up with SRV > > *************** > > > > 386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP > > Request: INVITE sip:18005551212@sip.acme.com, with > > session description > > 386.657307 10.100.0.111 -> 192.168.1.246 SIP > Status: > > 100 trying -- your call is important to us > > 88.811658 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query SRV _sip._udp.sip.acme.com > > 98.825208 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response, No such name > > 98.825496 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query A sip.acme.com > > 108.845136 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response, No such name > > 108.845468 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query AAAA sip.acme.com > > 118.865215 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response, No such name > > 416.711408 10.100.0.111 -> 192.168.1.246 SIP > Status: > > 478 Unresolveable destination (478/TM) > > 407.921364 192.168.1.246 -> 10.100.0.111 SIP > Request: > > ACK sip:18005551212@sip.acme.com > > > > > *********************************************************************************** > > > > > > ************* > > after rewritehostport("inside.acme.com:5060"), SER > > only does an A lookup > > *************** > > > > 424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP > > Request: INVITE sip:18005551212@nat.foo.com, with > > session descripti > > on > > 415.971953 10.100.0.111 -> 192.168.1.246 SIP > Status: > > 100 trying -- your call is important to us > > 127.006508 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query A inside.foo.com > > 127.007557 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response A 192.168.1.211 > > 424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP > > Request: INVITE > sip:18005551212@inside.foo.com:5060, > > with session descriptio > > n > > 415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP > > Request: INVITE > sip:18005551212@inside.foo.com:5060, > > with session descriptio > > n > > > > > ****************************************************************** > > > > How can I get SER to consistently do SRV lookup > after > > rewriting of R-URI? > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

1 0

← Newer
1
2
3
4
5
6
7
...
26
Older →

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 January 2006