September 2007 - sr-users

[OpenSER-Users] Installing OpenSER in multiprocessor machine
by John Barry 13 Sep '07

13 Sep '07

Hello, I would like to know if there are any especial considerations when compiling, configuring, and running OpenSER v1.2.2 in a multiprocessor machine. I am using a quad-processor machine (Intel Xeon @ 3.00GHz) with Red Hat Enterprise Linux AS release 4 (2.6.9-42.0.3.ELsmp #1 SMP x86_64 x86_64 x86_64 GNU/Linux). I am also using MySQL DB for authentication in the same machine. I will appreciate any advice or information. Bests, JB _________________________________________________________________ Check out some new online services at Windows Live Ideas—so new they haven’t even been officially released yet. http://ideas.live.com

2 1

[Serusers] problem in DNS lookup from enum
by geethanjali.csg＠wipro.com 13 Sep '07

13 Sep '07

Hi, I am configuring ser to use enum module. I am getting this error when dns lookup is done from ser. 0(27798) get_record: lookup(8.7.6.e164.arpa, 35) failed 0(27798) enum_query(): No NAPTR record found for 8.7.6.e164.arpa. This is my zone file: $TTL 3D @ IN SOA e164.arpa. root( 4 ; serial (d. adams) 3H ; refresh 15M ; retry 1W ; expiry 1D ) ; minimum IN NS localhost IN NS e164.arpa. localhost A 127.0.0.1 e164.arpa. A 10.200.220.200 $ORIGIN 1.2.3.4.e164.arpa. IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:abcd@e164.arpa!" . $ORIGIN 8.7.6.e164.arpa. IN NAPTR 20 100 "u" "E2U+sip" "!^.*$!sip:xyz@e164.arpa!" . named.conf file contains, zone "e164.arpa" IN { type master; file "/var/named/named.enum"; }; This is my enum code in ser.cfg. loadmodule "/usr/local/lib/ser/modules/enum.so" modparam("enum", "domain_suffix", "e164.arpa.") if (method == "INVITE" && uri =~ "sip:\+[0-9]*+@.") { if (enum_query("e164.arpa")) { route(1); }; }; Its possible to ping the domain e164.arpa from the m/c where ser is installed. Any help in this regard is appreciated. Regards Geethanjali The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

Re: [Serusers] WARNING: xl_mod_init: more IP 10.0.0.21 not used and ERROR: init_tcpudp_sock: bind: Address already in use [98]
by Jiri Kuthan 13 Sep '07

13 Sep '07

At 11:55 12/09/2007, Frank Wang wrote: >I downloaded ser 2.0.0-rc1 source code from <http://www.iptel.org/ser_2_0_0_release_candidate_1>http://www.iptel.org/ser_2_0_0_release_candidate_1 and install on my Linux step by step according the tutorial on the WEB. > >When I running SER, I got follow messages: > >[root@sipmcu ser]# ../../sbin/ser -f ser.cfg >Listening on > udp: <http://10.0.0.21>10.0.0.21 [<http://10.0.0.21>10.0.0.21]:5060 > tcp: <http://10.0.0.21>10.0.0.21 [<http://10.0.0.21>10.0.0.21]:5060 >Aliases: > tcp: <http://sipmcu.leadtek.com.cn:5060>sipmcu.leadtek.com.cn:5060 > udp: <http://sipmcu.leadtek.com.cn:5060>sipmcu.leadtek.com.cn:5060 > *: mytestser.org:* > >WARNING: no fork mode > 0(3078) init_tcp: using epoll_lt as the io watch method (auto detected) > 0(3078) Maxfwd module- initializing > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used unimportant. (I checked the code and I was a bit puzzled by it, but it really shold not matter much.) > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) ERROR: init_tcpudp_sock: bind: Address already in use [98] This is the problem. Maybe you have a SER instance running from the last time and you are starting the script again, for which reason SER can't be restarted -- it can't claim the same port number twice. > 0(3078) ERROR: ctl: mod_init: init ctrl. sockets failed > 0(3078) init_mod(): Error while initializing module ctl >ERROR: error while initializing modules This could have same reason. > >could anybody tell me how about it? Thanks a lot. > >Regards, >Frank >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

2 2

Re: [OpenSER-Users] WeSIP with openser 1.2.0 again
by Elias Baixas 13 Sep '07

13 Sep '07

Hi Gregorio, posting the SIP message that caused the problem would help me reproduce it, also some OpenSER logs would help. It seems to be related to the binary-encoding of the message. try out sipSpy to capture messages if you like. remove any field which you find confidential (lest Max-Forwards...) from the SIP message send it to me privately if you want. regards, Elias Gregorio wrote: > > Hi everyone, > > > > Hi! > > Just installed WeSIP beta-0.1.4 and openser 1.2.0 in the same machine, > configuration looks alright because some SIP Servlets receive > messages. Now I am getting an exception in the SEAS module when I > receive MESSAGE, I think this problem is similar to a previous one in > the mailing list (Sat Apr 28 16:07:34 CEST 2007, [WeSIP with openser > 1.2.0 again]). Does someone know anything about it? > > > > This is the exception: > > > > Thanks for helping. > > > > java.lang.ArrayIndexOutOfBoundsException: 5942 > > at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) > > at > com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) > > at > com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) > > at > com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) > > at > com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) > > at > com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) > > at > com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) > > at > com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:357) > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > Exception in thread "ExpressMChannel[1]" java.lang.NullPointerException > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > java.lang.ArrayIndexOutOfBoundsException: 5942 > > at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) > > at > com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) > > at > com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) > > at > com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) > > at > com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) > > at > com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) > > at > com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) > > at > com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:357) > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[OpenSER-Users] redirect server and handling 302 messages
by erik.linder＠accenture.com 12 Sep '07

12 Sep '07

Hello, I'm trying to make openser process 302 messages though I have problems(part of .cfg file below). I'll briefly explain the call flow I'd like to have: UA1 ---INVITE---> Session border controller ---INVITE--->SIP PROXY1 SIP PROXY1---INVITE--->REDIRECT SERVER---302 MOVED--->SIP PROXY1 SIP PROXY1---INVITE--->SIP PROXY2---INVITE---> Session border controller---INVITE---> UA2 This is how I want it to work. However managed to get a weird call flow were the 302 is forwarded back to the session border controller by the proxy1 which then trigger an INVITE sent to the SIP proxy2. But that is not what I want. My 302 message looks like this 20:12:34.314705 IP 192.168.200.101.sip > 192.168.200.102.sip: SIP, length: 488 E.....@.@.%....e...f........SIP/2.0 302 Moved temporarily Via: SIP/2.0/UDP 192.168.200.102;branch=z9hG4bK1658.2543f3d6.0 Via: SIP/2.0/UDP 192.168.200.10:5060;branch=z9hG4bKpis9m130dgj04b08v7k1.1 To: "22222"<sip:22222@btsip.bt.com>;tag=cc0e227ef05fff2dd833db745f3905a6.25f 2 From: "111"<sip:11111@btsip.bt.com>;tag=SDvto0501-df614a6e Call-ID: SDvto0501-515b91532ff7927a8345f39add52762f-06ahc50 CSeq: 1 INVITE Contact: , sip:22222@mydomain.com;maddr=192.168.200.103:5060;side=term> Content-Length: 0 192.168.200.101 is redirect server 192.168.200.102 is the proxy1 192.168.200.103 is proxy2 192.168.200.10 is session border controller 1. I would like to have proxy1 process the 302 message but there seems (from other mails on the mailing list) that openser does not allow routing depending on maddr in the Contact field. Sic :-( <- how can this routing decision been taken , using rewritehostport after get_redirect("*") is kind of ignored. The only workaround I have is to change the mydomain.com to the IP of proxy2 to get the automatic redirect 2. How can I block the redirect to the session border controller and send an invite right away to the proxy2? Here are the relevant part of the logic I use in proxy1 to handle the 302 I'm using this to trigger the forward to the redirect server: lookup("aliases"); #dest is remote if (!lookup("location")) { t_on_failure("3"); xlog("L_INFO","redirect invite to SLS\n"); #change orig to term in request URI $avp(s:request_uri_to_use)=$ru; avp_subst("$avp(s:request_uri_to_use)", "/orig/term/"); avp_pushto("$ru","$avp(s:request_uri_to_use)"); #forward("udp:192.168.200.101:5060"); #exit; rewritehostport("192.168.200.101:5060"); if(!t_relay()) { sl_reply_error(); } return; and this to try to handle the 302 that is received but it does not behave like I want: failure_route[3] { xlog("L_WARN","redirect FAILUREROUTE: time_t=$Ts messageid=$mi remark=here we are $bR\n"); if(t_check_status("3[0-9][0-9]")) { get_redirects("*"); xlog("L_WARN","^method=$rm ^ruri=$ru ^messageid=$mi^remark=2status is 3xx\n--$br--\nbR((($bR)))\n"); #rewritehost("$(ru{uri.maddr})"); serialize_branches(0); } xlog("L_WARN","nextbranch\n"); #next_branches(); #t_on_failure("3"); #t_relay(); return; #t_reply("404", "No branches left"); } Help would be appreciated Cheers erik This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited.

2 2

[OpenSER-Users] authorization problems with multiple servers
by Ben Smithurst 12 Sep '07

12 Sep '07

Hi We have a problem with clients authorizing (whether for INVITE or REGISTER) against a pair of openser servers, served via either round robin A records, or multiple equal preference SRV records. The problem in a nutshell is client sends INVITE to server 1 server 1 replies with authorization challenge client sends INVITE with auth to server 2 server 2 doesn't like the 'nonce' value Now, this only happens via our nat proxy, as far as we can tell, so I think the problem lies there (and I have contacted the vendor of that system). However I wonder if there is any way to make openser work in such a setup, in case other client do the same thing (which IMO is quite a stupid thing to do) ... would the valid 'nonce' values need to be stored in the DB? Incidentally we are using mysql authentication and we are using usrloc in db_mode 3, with both openser machines pointed at the same mysql server. We're using a pretty recent openser SVN revision (dev15). Any tips would be appreciated. thanks -ben -- Ben Smithurst http://www.bensmithurst.com/

2 3

[OpenSER-Users] WeSIP with openser 1.2.0 again
by Gregorio 12 Sep '07

12 Sep '07

Hi everyone, Hi! Just installed WeSIP beta-0.1.4 and openser 1.2.0 in the same machine, configuration looks alright because some SIP Servlets receive messages. Now I am getting an exception in the SEAS module when I receive MESSAGE, I think this problem is similar to a previous one in the mailing list (Sat Apr 28 16:07:34 CEST 2007, [WeSIP with openser 1.2.0 again]). Does someone know anything about it? This is the exception: Thanks for helping. java.lang.ArrayIndexOutOfBoundsException: 5942 at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) at com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266 ) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser .java:357) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[1]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) java.lang.ArrayIndexOutOfBoundsException: 5942 at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) at com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266 ) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser .java:357) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595)

1 0

[OpenSER-Users] Wanted: VoIP Engineer for Warsaw !
by laurent schweizer 12 Sep '07

12 Sep '07

Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates. Peoplefone is a certified partner of Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid…>and AVM/FRITZ!Box <http://www.fritz-shop.ch/>. Due to our rapid growth, for our new Polish subsidiary we are currently seeking for: VOIP SPECIALIST Place of work: Warsaw *Requirements:* - Graduation from college or university with a Bachelor's degree (preferably IT) - Experience with PHP - Practical knowledge of C and C++ - Practical knowledge of Mysql and Postgresql - Linux experience - Knowledge of IP Networks, UDP, TCP - Experience with tools for Network analysis like Ethereal - VoIP basic knowledge, VoIP servers, configuration of devices - Fluency in English - Ability to interact with individuals and groups at all levels - Detail oriented and analytical - Strong verbal and written communication skills Knowledge of Perl, Java, Asterisk / SER / OPENSER, ability to configure routers, Cisco or Patton gateways, knowledge of SIP and STUN protocol, knowledge of NAT problems, of outbandProxy, knowledge of monitoring tools like Cactus, Nagios, MRTG or high availability tools like DRBD, Hearthbeat would be an additional asset. Interested individuals are requested to send their resume to: ewa.ciesla(a)peoplefone.com In your application please put the following statement: I hereby agree for processing the following personal information strictly for recruitment purposes in accordance with the regulation regarding the protection of personal data passed on the following date: 29.08.97r. Dz.U nr 133 poz. 883.

1 0

[OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 12 Sep '07

12 Sep '07

Hello, I have problems calling from my phones connected to openser to grandstream phones on another university to ser. Problem is, that my openser generates a "CANCEL" SIP packet sent to grandstream phone, but no CANCEL received from my phone. Wireshark shows something like this: 1189587144.843962 158.195.15.100 -> 158.197.16.77 SIP/SDP Status: 200 OK, with session description 1189587144.845528 158.197.16.77 -> 158.195.15.100 SIP Request: CANCEL sip:+421xxxx44903@domain.sk 1189587144.853654 158.195.15.100 -> 158.197.16.77 SIP Status: 200 canceling Communication before is OK. 158.197.16.77 is my openser server and 158.195.15.100 is peers phone. Any idea, why openser can generate this packet? Can openser generate a SIP packet not received from client? Calling from my phone to another phone on same university works well. Thank you. SAL

3 4

[Serusers] WARNING: xl_mod_init: more IP 10.0.0.21 not used and ERROR: init_tcpudp_sock: bind: Address already in use [98]
by Frank Wang 12 Sep '07

12 Sep '07

I downloaded ser 2.0.0-rc1 source code from http://www.iptel.org/ser_2_0_0_release_candidate_1 and install on my Linux step by step according the tutorial on the WEB. When I running SER, I got follow messages: [root@sipmcu ser]# ../../sbin/ser -f ser.cfg Listening on udp: 10.0.0.21 [10.0.0.21]:5060 tcp: 10.0.0.21 [10.0.0.21]:5060 Aliases: tcp: sipmcu.leadtek.com.cn:5060 udp: sipmcu.leadtek.com.cn:5060 *: mytestser.org:* WARNING: no fork mode 0(3078) init_tcp: using epoll_lt as the io watch method (auto detected) 0(3078) Maxfwd module- initializing 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) ERROR: init_tcpudp_sock: bind: Address already in use [98] 0(3078) ERROR: ctl: mod_init: init ctrl. sockets failed 0(3078) init_mod(): Error while initializing module ctl ERROR: error while initializing modules could anybody tell me how about it? Thanks a lot. Regards, Frank

1 0

Fwd: [OpenSER-Users] Transaction Deletion
by Brandon Armstead 12 Sep '07

12 Sep '07

---------- Forwarded message ---------- From: Brandon Armstead <brandon.armstead(a)gmail.com> Date: Sep 7, 2007 12:06 PM Subject: Re: [OpenSER-Users] Transaction Deletion To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Hello Bodgan, There is no noticable delay however the only thing I see is that one of the RR'ed proxy's responds sends out 200 OK rather than the original. You can see an example sipcap of this at this url: http://pastebin.ca/685897 . If you could let me know if you see anything that looks rather, odd, towards the end you can see the last few transactions and where there is no response to 216.82.224.203's 200 OK, where it flips from 216.82.224.202 to 216.82.224.203 from the RR. Also then later the PSTN follows with a BYE, due to no response, once again thanks for your help! Regards, Brandon On 9/7/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > > Hi Brandon, > > Is there any noticeable delay between the 200OK and ACK ? > > regards, > bogdan > > Brandon Armstead wrote: > > Hello Users, > > > > I have a weird scenario where the PSTN is responding with 200 OK, > > however Openser Never forwards on the ACK, after several > > retransmissions of 200OK to Openser, PSTN responds with BYE, anyone > > know what might be going on? I've played with the timers to no avail, > > any help is appreciated, thank you! The log below is as follows: > > > > 375 Sep 6 12:18:10 openser /usr/local/sbin/openser[17385]: DEBUG: > > timer routine:1,tl=0x2b78e4e73588 next=(nil), timeout=261 > > 376 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > timer routine:2,tl=0x2b78e4e733b8 next=(nil), timeout=262 > > 377 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > wait_handler : removing 0x2b78e4e73338 from table > > 378 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > delete transaction 0x2b78e4e73338 > > 379 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > wait_handler : done > > 380 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: > > DEBUG:parse_to:end of header reached, state=10 > > 381 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: > > get_hdr_field: <To> [26]; uri=[sip:openser] > > 382 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: to > > body [<sip:openser> ] > > 383 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: > > get_hdr_body : content_length=0 > > 384 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: > > DEBUG:destroy_avp_list: destroying list (nil) > > 385 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: > > DEBUG:forward_reply: found module tm, passing reply to it > > 386 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_check: start=0xffffffffffffffff > > 387 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > add_param: tag=gK0288a919 > > 388 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: > > DEBUG:parse_to:end of header reached, state=29 > > 389 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > get_hdr_field: <To> [53]; uri=[ sip:17144828306@openser] > > 390 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: to > > body [<sip:17144828306@openser>] > > 391 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: hash 53903 label 1526431020 branch 0 > > 392 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: no matching transaction exists > > 393 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: failure to match a transaction > > 394 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_check: end=(nil) > > > > It seems like its not matching the transaction, because a timer is > > removing the transaction? Any ideas? Thanks! > > -- > > Brandon Armstead > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > -- Brandon Armstead -- Brandon Armstead

2 1

Re: [OpenSER-Users] 513 Massage too big
by Bogdan-Andrei Iancu 12 Sep '07

12 Sep '07

Arya, it means you have a routing error in the script - you need to debug your script to see where the routing logic is broken - try using xlog() messages to see the execution path of your script. I cannot help you with more than this. regards, bogdan Arya wrote: > any idea how I can fix the problem? > > On 9/11/07, *Arya* <arya6000(a)gmail.com <mailto:arya6000@gmail.com>> > wrote: > > I used ngrep but it just shows that the massage keeps bouncing > server to client, and eventually it says massage too big > > > On 9/11/07, * Bogdan-Andrei Iancu* < bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi, > > use ngrep or tcpdump to get a network trace from the proxy > server, on > the loopback interface. > > Regards, > Bogdan > > Arya wrote: > > Can you tell me how I can trace it? could this be an issue > with my > > router? its the Linksys Wr54G and its set on DMZ to the server > > > > On 9/10/07, *Bogdan-Andrei Iancu * <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro> > > <mailto: bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>> wrote: > > > > Hi Arya, > > > > I suspect you have a routing problem that cause the > request to loop on > > openser (via loopback interface) - openser keeps sending the > > request to > > itself. That's why you get a "Message to big" or "too > many hops".... > > > > run a trace on lo to see if this is what's happening. > > > > regards, > > bogdan > > > > Arya wrote: > > > Hello everyone > > > > > > I have a problem connecting to OpenSER from my public IP. > > > > > > I can connect and make calls to OpenSER inside my LAN > but when I > > > connect from outside LAN I get the 513 Massage too big > massage. > > I than > > > changed the openSER default config's section that > checks a message > > > size to a bigger number and I got the I get the "483 - > too many > > Hops" > > > error. > > > > > > I changed the config back to default and tested it with > ngrep and I > > > tried connecting to the server from a server in another > state. > > > > > > The SIP server is on 192.168.1.109 > <http://192.168.1.109> <http://192.168.1.109 > <http://192.168.1.109>> > > <http://192.168.1.109> and the > > > computer trying to connect is on 192.168.1.100 > <http://192.168.1.100> > > < http://192.168.1.100> <http://192.168.1.100 < > http://192.168.1.100>> > > > > > > And ngrep gave me these massages: >

1 0

[OpenSER-Users] 'while' statement in config file
by Daniel-Constantin Mierla 12 Sep '07

12 Sep '07

Hello, although is more part of devel, the decision will affect users of future stable releases, so I posted on both lists. Maybe some of you noticed that in the ongoing process of improving the pseudo-variables system, the AVPs and HDRs can be accessed with dynamic index. That means next lines are correct in the config file. $var(i) = 1; xlog("$(avp(i:20)[$var(i)])"); if($(avp(i:20)[$var(i)]) == 3) { } In the past, the index was static, as a number. The question I want to address is whether makes sense to add 'while' statements to iterate through avps or hdrs. More or less, the cycles can be done via recursive routes. The block: $var(i) = 0; while($(avp(i:20)[$var(i)]) != null) { ..... $var(i) = $var(i) +1; } is equivalent with: $var(i) = 0; route(1); .... route[1] { if(($(avp(i:20)[$var(i)]) == null) return; ... $var(i) = $var(i) +1; route(1); } 'while' may look more attractive, but could open the pandora box, people asking for more and more, that will turn config language is something very complex (not saying that is simple now :-) ). The drawback with recursive routes is the fixed number of routes that can be defined with the standard openser: 40 (number can be increased if openser is recompiled), because for each cycle another route has to be used. Waiting for your opinions... Cheers, Daniel

3 2

[OpenSER-Users] add/remove headers and append_branch
by Madalina Pitis 12 Sep '07

12 Sep '07

Hello, I have a problem trying to remove a header using remove_hf() after an append_branch(). I am using 1.2 version of OpenSer. The scenario with the problem is as described in the folowing lines. When I receive an INVITE I try a parallel fork (one leg towards PSTN, the other towards IP), at which time I append a "Redirection" header. The PSTN leg will be answered with a 486 response, due to the presence of the "Redirection" header. When the IP leg expires, I receive a 408 response, at which time I do a serial fork, and resend the message towards the PSTN destination, this time without the "Redirection" header. The problem is that I cannot seem to be able to remove (or find) the header. I tried everywhere (before appending the new branch, after, in branch_route, before relaying), as you will see below. I tried searching the mailing lists but nothing turned up. I would greatly appreciate any help, for I am running out of ideas. Here is a simplified version of my configuration script, containing only the significant bits of the routing blocks: #--------------------------------------------------- # main routing logic #--------------------------------------------------- route { # check maxfwd and message lenght # do loose routing if (method=="INVITE") { # rewrite ruri rewriteuri(<PSTN destination>); # signal forking append_hf("Redirection: count=5\r\n"); append_branch(); revert_uri(); t_on_failure("1"); if (!t_relay()){ xlog("acc_info", " ERROR"); sl_reply_error(); } } else if (method=="ACK" || method=="BYE" || method=="CANCEL") { # default message handler } else { # forbidden message handler } } #--------------------------------------------------- # Failure route relay #--------------------------------------------------- route[1] { xlog("acc_info", " Failure route relay"); # try removing it before t_relay if(is_present_hf("Redirection")){ xlog("acc_info", " Remove before t_relay"); remove_hf("Redirection"); } if (!t_relay()){ xlog("acc_info", " ERROR"); sl_reply_error(); } } #--------------------------------------------------- # Failure route handling #--------------------------------------------------- failure_route[1] { xlog("acc_info", " failure route handler"); if(t_check_status("408")) { t_on_branch("1"); rewriteuri(<PSTN destination>); # try removing it before append branch if(is_present_hf("Redirection")){ xlog("acc_info", " Remove before append branch"); remove_hf("Redirection"); } append_branch(); # try removing it after append branch if(is_present_hf("Redirection")){ xlog("acc_info", " Remove after append branch"); remove_hf("Redirection"); } route(1); } }

2 2

[OpenSER-Users] append_hf() and failure_route issue
by Alex Hermann 11 Sep '07

11 Sep '07

Hello all, I'm having some trouble with failure routes and custom SIP headers. In FAILURE_ROUTE I set the ruri to a new destination. This destination needs a custom SIP header. When I add a SIP header with append_hf() (from within the failure_route), this header is correctly sent to the first IP of the destination on t_relay(). However, when this destination fails with 503 or timeout, the built-in DNS based failover doesn't add the custom SIP header to the new requests for subsequent IP's. When the SIP header is added from REQUEST_ROUTE the header is sent to all destinations by the DNS based failover mechanism. How do I get the DNS based failover to send my SIP headers all the time? (The same problem is present with uac_auth()) -- Greetings, Alex Hermann

1 0

Re: [OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by Iñaki Baz Castillo 11 Sep '07

11 Sep '07

First of all sorry for this post not being in the thread it belongs to (I've not the initial mail). > But, I don't want to use the Asterisk dtmf features, > I want to use the SIP INVITE/REFER features from the > UAC (actually polycom and snom phones). > > Can someboy shed me some light on this? You receive a call from Asterisk UAC and want to transfer (blind or attender) to other UACregistered in OpenSer. Of course you don't need Asterisk native transfer, SIP allows blind and attended transfer. The only you need is that your SIP devices allow both of them (unfortunatelly many SIP devices just allow blind transfer, but in most cases reading the device manual you'll find how to do a attended transfer). The point here is that Asterisk allows receiving a REFER, so don't worry for that (I use your same escenary with no problem and SIP transfers). Maybe you find useful this links: http://tech-invite.com/Ti-sip-service-4.html http://tech-invite.com/Ti-sip-service-5.html http://dev.openwengo.com/trac/openwengo/trac.cgi/wiki/SipTransfer Regards. -- Iñaki Baz Castillo

1 0

Re: [OpenSER-Users] 513 Massage too big
by Bogdan-Andrei Iancu 11 Sep '07

11 Sep '07

Hi, use ngrep or tcpdump to get a network trace from the proxy server, on the loopback interface. Regards, Bogdan Arya wrote: > Can you tell me how I can trace it? could this be an issue with my > router? its the Linksys Wr54G and its set on DMZ to the server > > On 9/10/07, *Bogdan-Andrei Iancu * <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi Arya, > > I suspect you have a routing problem that cause the request to loop on > openser (via loopback interface) - openser keeps sending the > request to > itself. That's why you get a "Message to big" or "too many hops".... > > run a trace on lo to see if this is what's happening. > > regards, > bogdan > > Arya wrote: > > Hello everyone > > > > I have a problem connecting to OpenSER from my public IP. > > > > I can connect and make calls to OpenSER inside my LAN but when I > > connect from outside LAN I get the 513 Massage too big massage. > I than > > changed the openSER default config's section that checks a message > > size to a bigger number and I got the I get the "483 - too many > Hops" > > error. > > > > I changed the config back to default and tested it with ngrep and I > > tried connecting to the server from a server in another state. > > > > The SIP server is on 192.168.1.109 <http://192.168.1.109> > <http://192.168.1.109> and the > > computer trying to connect is on 192.168.1.100 > <http://192.168.1.100> <http://192.168.1.100 <http://192.168.1.100>> > > > > And ngrep gave me these massages: > > > > > > > > > -- > Thank You

1 0

[OpenSER-Users] Openser and Voice master
by Dan 11 Sep '07

11 Sep '07

Hi, Has anybody able to successfully able to route calls to voicemaster from openser. I tried all method but none worked. From username/password to IP based authentication to prefix based authentication. Regards Dan

1 0

[OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by perl＠libkvm.org 11 Sep '07

11 Sep '07

Hello, I am configuring a platform unig OpenSER 1.2 with Asterisk 1.2. I have all the SIP phones registered on OpenSER. I use the realtime setup on Asterisk using a view on MySQL. I need to be able to process the attended transfer between two UAC registered on OpenSER, having the first received a call from the Asterisk side, so that the second UAC gets the call transferred. I also need to do this as a blind transfer. But, I don't want to use the Asterisk dtmf features, I want to use the SIP INVITE/REFER features from the UAC (actually polycom and snom phones). Can someboy shed me some light on this? I need to evaluate if this is possible, or if it needs lots of sources modifications. Thank you very much.

1 0

[Serusers] ERROR: warning_builder: buffer size exceeded
by Kostas Marneris 10 Sep '07

10 Sep '07

Hello all, I have occasionally the following warning mesgs at my SER log : Sep 10 14:34:27 serhost ser[17957]: ERROR: warning_builder: buffer size exceeded Sep 10 14:34:27 serhost ser[17957]: WARNING: warning skipped -- too big I noticed that the R-URI in that case is : R-URI=<sip:XXXXXXXXXX@mydomain.com;user=phone;transport=UDP> when usually the R-URI is : <sip:XXXXXXXXXX@mydomain.com;transport=UDP> <sip:XXXXXXXXXXXXX@mydomain.com> Anyone seen those mesgs before ? Can this explained by the extra field 'user=phone' which the SIP UA adds at the R-URI of the INVITE mesg ? thanks Kostas

3 2

[OpenSER-Users] Problem with rtpproxy
by Jeferson Prevedello 10 Sep '07

10 Sep '07

Hi Guys, When I originate a call for an user behind of a NAT the call is without audio in the two ways. I believe that the problem is related to rtpproxy, however did not find nothing of wrong in my configuration. Some tip? openser.cfg: # ------------- !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # Nathelper with RTPproxy service run: ps -aux | grep rtp root 26098 0.0 0.0 2532 428 ? Ss 10:40 0:00 /usr/bin/rtpproxy root 11143 0.0 0.0 2536 352 ? Ss 11:30 0:00 /usr/bin/rtpproxy -l xxx.xxx.xxx.xxx -s udp:xxx.xxx.xxx.xxx 7890 logs: Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11859]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11860]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11861]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11862]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Regards Jeferson

2 2

[Serusers] Need SIP proxy to translate UDP<->TCP
by Thermal Wetland 10 Sep '07

10 Sep '07

Hello, I would like to hire someone to setup a SIP UDP<->TCP proxy using SER. SER will sit between an Asterisk box and Microsoft Exchange 2007. Asterisk speaks UDP & the Exchange box speaks TCP. I will make a fresh fully patched CentOS 5 box available for the work. Payment could be through PayPal or Rent a Coder or similar. I would expect an SER expert to be able to do this in less than 2 hours. Please let me know if you are interested - thermalwetland at gmail dot com. Thanks, Thermal

2 1

Re: [OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Hi Bogdan I got it fixed. The problem is on this line: if(!t_relay("0x05")) if I change to if(!t_relay()) everything is ok. Don't know what is 0x05 mean. Regards, Halomoan On 9/10/07, Halomoan Chow <halomoan(a)gmail.com> wrote: > > Hi > > My Openser version is openser-1.2.0-notls and the Mediaproxy is > mediaproxy-1.9.0 > I was using the openser.cfg that was generated from > http://www.sipwise.com/wizard/ for Openser + Mediaproxy only. > > Just only simple call from A to B cause segmentation fault. > > I don't know how to locate the core file and get the backtrace :( > > Regards, > > Halomoan > > On 9/10/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > > > > Hi Halomoan, > > > > provide full details about your openser version - version, how did you > > get it (source, binaries, etc). > > also locate the core file and get the backtrace. > > > > Regards, > > Bogdan > > > > Halomoan Chow wrote: > > > Dear All > > > > > > I'm running into segmentation fault error on Openser 1.2.x with > > > Mediaproxy. > > > What is the first thing I should take a look into to find out which > > > line or which module causing this segmentation fault? > > > Thank you. > > > > > > Regards, > > > > > > Halomoan > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

3 3

Re: [Serusers] [Serdev] Release of SER 2.0 (and future 2.1)
by Greger V. Teigre 10 Sep '07

10 Sep '07

Ok, I conclude then that final release of SER 2.0 is aimed at October 15. That should allow us to align SER, serctl, SERWeb, rtpproxy, and SEMS. SER, SERWeb, and SEMS will have independent releases, and they will be aligned to be interoperable. SER, rtpproxy, and serctl will have a common release. The release will include the new buildsystem that was tested in release candidate 1. The config file and features available work with sercmd, serctl, as well as with SERWeb. SEMS support is planned, but pending some patches. All components will be combined in a one-file download with sources for all components, the SIP Express Bundle. I urge everybody with patches, bug reports, or tasks related to this release to prioritize those now! g-) Greger V. Teigre wrote: > Guys, > My last post concerned documentation. I suggested decoupling > documentation from the CVS to allow us to work independently with code > and docs. > > I suggest we move forward on the decoupling and set a date for SER 2.0 > release. My suggestion is that we target October 15. This will allow > people to allocate time to do the last clean-ups, bug-fixes, and patches. > > I also suggest that we set a release date for 2.1. AFAIU, there is > already quite a lot of code that has gone into the trunk and as 2.1 is a > minor release and we strive for ser.cfg compatibility with 2.0, we > shouldn't really wait too long. > > I suggest mid-January 2008. How does that work for you? > g-) > _______________________________________________ > Serdev mailing list > Serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev > > >

1 0

[OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Dear All I'm running into segmentation fault error on Openser 1.2.x with Mediaproxy. What is the first thing I should take a look into to find out which line or which module causing this segmentation fault? Thank you. Regards, Halomoan

2 1

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 September 2007