sr-users November 2004

sr-users@lists.kamailio.org

203 participants
386 discussions

[Serusers] please help serweb email confirmation
by rosita sita 05 Nov '04

05 Nov '04

Dear all, i have problems in using SERWEB. i get the latest serweb in berlios. i'm running ser on redhat 9. my version ser is 8.12. mysql 3.x.x , sendmail 8.12 and php 4.x.x. my ser and my serweb are running well but when someone want to subscribe via serweb the e-mail confirmation doesn't arrive to user who subscribe and serweb say: register is successfully. i have check php.ini, httpd.conf, and sendmail configuration, they all seems OK. i tried to solve this problems in a week but the result is ZERO. please help me . Regards, Sita, Politeknik Caltex Riau Indonesia Sita Rosita ___________________________________________________ Create your own safe chat rooms. http://www.homemaster.net - Homemaster. Come Together. Online.

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[Serusers] problem registering with serweb, no email confirmation to user.
by rosita sita 05 Nov '04

05 Nov '04

dear all, im using ser 8.12,redhat 9.0,php.4.x,mysql.3.x,sendmail.8.x. and serweb i download the latest version on berlios. all of my ser service is running ok, but when some one access serweb to subscribe as new users. in my serweb it's say okay and you are sucessfully and the email confirmation will be sent shortly. but when i check in users email, the email isn't arrived. how comes. i just try this in a intranet. are anyone have idea and hint??? regard's sita (politeknik caltex riau) indonesia Sita Rosita ___________________________________________________ Meet other people who share your interests. http://www.homemaster.net - Homemaster. Come Together. Online.

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[Serusers] Bad ports in a Messenger Client comunication
by RAQUEL LOPEZ DIAZ 05 Nov '04

05 Nov '04

Hi!! I have one promblem in a comunication with two Client. I use sip express router 0.8.14 with Nat and rtpproxy. I captured packets: U 81.35.36.166:19168 -> 194.179.25.52:5060 INVITE sip:raquel@wmserver.hi.inet SIP/2.0..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <sip:chano@wmserver.hi.inet>;tag=fe05c1ce-52 c4-4814-9689-c635dcf48e35..To: <sip:raquel@wmserver.hi.inet>..Call- ID: 0bfff467-3c8c-4be1-ba2b-eba9b1106cff@81.35.36.166..CSeq: 1 INVITE..C ontact: <sip:81.35.36.166:19124>..User-Agent: Windows RTC/1.0..Content-Type: application/sdp..Content-Length: 531....v=0..o=sayuritra 0 0 I N IP4 81.35.36.166..s=session..c=IN IP4 81.35.36.166..b=CT:1000..t=0 0..m=audio 19183 RTP/AVP 97 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8 000..a=rtpmap:111 SIREN/16000..a=fmtp:111 bitrate=16000..a=rtpmap:112 G7221/16000..a=fmtp:112 bitrate=24000..a=rtpmap:6 DVI4/16000..a=rtpma p:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a= fmtp:101 0-16..m=video 52292 RTP/AVP 34 31..a=rtpmap:34 H263/90000..a=rtpmap:31 H261/90000.. # U 194.179.25.52:5060 -> 81.35.36.166:19124 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <sip:chano@wmserver.hi.inet>;tag=fe0 5c1ce-52c4-4814-9689-c635dcf48e35..To: <sip:raquel@wmserver.hi.inet>..Call-ID: 0bfff467-3c8c-4be1-ba2b- eba9b1106cff@81.35.36.166..CSeq: 1 I NVITE..Server: Sip EXpress router (0.8.14 (i386/linux))..Content- Length: 0..Warning: 392 194.179.25.52:5060 "Noisy feedback tells: pid=884 8 req_src_ip=81.35.36.166 req_src_port=19168 in_uri=sip:raquel@wmserver.hi.inet out_uri=sip:80.32.97.245:24761 via_cnt==1".... # U 194.179.25.52:5060 -> 80.32.97.245:24761 INVITE sip:80.32.97.245:24761 SIP/2.0..Max-Forwards: 10..Record- Route: <sip:raquel@194.179.25.52;ftag=fe05c1ce-52c4-4814-9689- c635dcf48e35; lr=on>..Via: SIP/2.0/UDP 194.179.25.52;branch=z9hG4bKd76e.f9f1d9c2.0..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <sip:chano@wmserve r.hi.inet>;tag=fe05c1ce-52c4-4814-9689-c635dcf48e35..To: <sip:raquel@wmserver.hi.inet>..Call-ID: 0bfff467-3c8c-4be1-ba2b- eba9b1106cff(a)81.35 .36.166..CSeq: 1 INVITE..Contact: <sip:81.35.36.166:19124>..User- Agent: Windows RTC/1.0..Content-Type: application/sdp..Content-Length: 550 ..P-Behind-NAT: Yes....v=0..o=sayuritra 0 0 IN IP4 81.35.36.166..s=session..c=IN IP4 194.179.25.52..b=CT:1000..t=0 0..m=audio 35016 RTP/AVP 97 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8000..a=rtpmap:111 SIREN/16000..a=fmtp:111 bitrate=16000..a=rtpmap:112 G7221/16000..a=fmtp:112 bitrate=24000..a=rtpmap:6 DVI4/16000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:3 G SM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..m=video 52292 RTP/AVP 34 31..a=rtpmap:34 H263/90000..a=rtpmap:31 H261/90000..a =nortpproxy:yes.. # U 80.32.97.245:24761 -> 194.179.25.52:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 194.179.25.52;branch=z9hG4bKd76e.f9f1d9c2.0..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <sip:c hano(a)wmserver.hi.inet>;tag=fe05c1ce-52c4-4814-9689-c635dcf48e35..To: <sip:raquel@wmserver.hi.inet>;tag=183dfa8b-0a7b-43b4-9fd4-01dfd3d6dbf3 ..Call-ID: 0bfff467-3c8c-4be1-ba2b-eba9b1106cff@81.35.36.166..CSeq: 1 INVITE..User-Agent: Windows RTC/1.0..Content-Length: 0.... # U 80.32.97.245:24761 -> 194.179.25.52:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 194.179.25.52;branch=z9hG4bKd76e.f9f1d9c2.0..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <sip: chano(a)wmserver.hi.inet>;tag=fe05c1ce-52c4-4814-9689- c635dcf48e35..To: <sip:raquel@wmserver.hi.inet>;tag=183dfa8b-0a7b-43b4- 9fd4-01dfd3d6dbf 3..Call-ID: 0bfff467-3c8c-4be1-ba2b-eba9b1106cff@81.35.36.166..CSeq: 1 INVITE..Record-Route: <sip:raquel@194.179.25.52;ftag=fe05c1ce-52c4-4 814-9689-c635dcf48e35;lr=on>..User-Agent: Windows RTC/1.0..Content- Length: 0.... # U 194.179.25.52:5060 -> 81.35.36.166:19124 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <sip:chano@wmserver.hi.inet>;tag=fe05c1ce-52c4-4814-9689-c635dcf48e 35..To: <sip:raquel@wmserver.hi.inet>;tag=183dfa8b-0a7b-43b4-9fd4- 01dfd3d6dbf3..Call-ID: 0bfff467-3c8c-4be1-ba2b- eba9b1106cff(a)81.35.36.166. .CSeq: 1 INVITE..Record-Route: <sip:raquel@194.179.25.52;ftag=fe05c1ce-52c4-4814-9689- c635dcf48e35;lr=on>..User-Agent: Windows RTC/1.0..Con tent-Length: 0.... # U 80.32.97.245:24761 -> 194.179.25.52:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 194.179.25.52;branch=z9hG4bKd76e.f9f1d9c2.0..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <sip:chano @wmserver.hi.inet>;tag=fe05c1ce-52c4-4814-9689-c635dcf48e35..To: <sip:raquel@wmserver.hi.inet>;tag=183dfa8b-0a7b-43b4-9fd4- 01dfd3d6dbf3..Ca ll-ID: 0bfff467-3c8c-4be1-ba2b-eba9b1106cff@81.35.36.166..CSeq: 1 INVITE..Record-Route: <sip:raquel@194.179.25.52;ftag=fe05c1ce-52c4- 4814-9 689-c635dcf48e35;lr=on>..Contact: <sip:192.168.1.34:9466>..User- Agent: Windows RTC/1.0..Content-Type: application/sdp..Content-Length: 528. ...v=0..o=hobbes 0 0 IN IP4 192.168.1.34..s=session..c=IN IP4 192.168.1.34..b=CT:1000..t=0 0..m=audio 17318 RTP/AVP 97 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8000..a=rtpmap:111 SIREN/16000..a=fmtp:111 bitrate=16000..a=rtpmap:112 G7221/16000..a=fmtp:112 bitrate=24000..a=rtpmap :6 DVI4/16000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 te lephone-event/8000..a=fmtp:101 0-16..m=video 20872 RTP/AVP 34 31..a=rtpmap:34 H263/90000..a=rtpmap:31 H261/90000.. # U 194.179.25.52:5060 -> 81.35.36.166:19124 SIP/2.0 200 OK..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <sip:chano@wmserver.hi.inet>;tag=fe05c1ce-52c4-4814-9689- c635dcf48e35..T o: <sip:raquel@wmserver.hi.inet>;tag=183dfa8b-0a7b-43b4-9fd4- 01dfd3d6dbf3..Call-ID: 0bfff467-3c8c-4be1-ba2b- eba9b1106cff@81.35.36.166..CSeq : 1 INVITE..Record-Route: <sip:raquel@194.179.25.52;ftag=fe05c1ce- 52c4-4814-9689-c635dcf48e35;lr=on>..Contact: <sip:80.32.97.245:24761>..Us er-Agent: Windows RTC/1.0..Content-Type: application/sdp..Content- Length: 547....v=0..o=hobbes 0 0 IN IP4 192.168.1.34..s=session..c=IN IP4 194.179.25.52..b=CT:1000..t=0 0..m=audio 35018 RTP/AVP 97 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8000..a=rtpmap:111 SIREN/16000..a=fmtp: 111 bitrate=16000..a=rtpmap:112 G7221/16000..a=fmtp:112 bitrate=24000..a=rtpmap:6 DVI4/16000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a =rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..m=video 20872 RTP/AVP 3 4 31..a=rtpmap:34 H263/90000..a=rtpmap:31 H261/90000..a=nortpproxy:yes.. # U 81.35.36.166:19168 -> 194.179.25.52:5060 ACK sip:raquel@194.179.25.52;ftag=fe05c1ce-52c4-4814-9689- c635dcf48e35;lr=on SIP/2.0..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <s ip:chano@wmserver.hi.inet>;tag=fe05c1ce-52c4-4814-9689- c635dcf48e35..To: <sip:raquel@wmserver.hi.inet>;tag=183dfa8b-0a7b-43b4- 9fd4-01dfd3d6 dbf3..Call-ID: 0bfff467-3c8c-4be1-ba2b- eba9b1106cff@81.35.36.166..CSeq: 1 ACK..Route: <sip:80.32.97.245:24761>..Contact: <sip:81.35.36.166: 19124>..User-Agent: Windows RTC/1.0..Content-Length: 0.... # U 194.179.25.52:5060 -> 80.32.97.245:24761 ACK sip:80.32.97.245:24761 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:raquel@194.179.25.52;ftag=fe05c1ce-52c4-4814-9689-c635dcf48e35;lr= on>..Via: SIP/2.0/UDP 194.179.25.52;branch=0..Via: SIP/2.0/UDP 81.35.36.166:19124..From: "chano" <sip:chano@wmserver.hi.inet>;tag=fe05c1ce- 52c4-4814-9689-c635dcf48e35..To: <sip:raquel@wmserver.hi.inet>;tag=183dfa8b-0a7b-43b4-9fd4- 01dfd3d6dbf3..Call-ID: 0bfff467-3c8c-4be1-ba2b-e ba9b1106cff@81.35.36.166..CSeq: 1 ACK..Contact: <sip:81.35.36.166:19124>..User-Agent: Windows RTC/1.0..Content-Length: 0..P-hint: rr-enforc ed.... For video, the ports used are: Client 1 : IP 81.35.36.166 port 52292 SER: IP 194.179.25.52 ports 20872 52292 Client 2: IP 80.32.97.245 port 20872 But, rtpproxy use: For client 1 port 20872 For client 2 port 52292 what´s happened?? I`m sorry my bad english. Thanks Raquel

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RE: [Serusers] sems IVR plugin bug???
by Java Rockx 05 Nov '04

05 Nov '04

Great. One thing I was thinking is that if most people need this value to be set smaller then someone should commit a change to CVS this this at 8 to 10 rather than 20. Thanks, Paul --- Richard <richard(a)o-matrix.org> wrote: > Hi Java, > > That value is to prevent double detection, i.e. the same key is detected > twice. Too large, it runs the risk of not detecting the right key. To small, > it increases the chance of double detection. > > I use 7 or 8 in my system. > > Richard > > > -----Original Message----- > > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On > > Behalf Of Java Rockx > > Sent: Thursday, November 04, 2004 5:44 PM > > To: Java Rockx; ser users > > Subject: Re: [Serusers] sems IVR plugin bug??? > > > > Hi All. > > > > Perhaps I just fixed this, but will my change cause other problems? The > > change I made was in > > IvrDtmfDetector.h > > > > I changed > > > > DTMF_INTERVAL = 20 > > > > to > > > > DTMF_INTERVAL = 10 > > > > And now dialing DTMF digits is ok. Will reducing DTMF_INTERVAL to 10 cause > > other issues? > > > > Regards, > > Paul > > > > --- Java Rockx <javarockx(a)yahoo.com> wrote: > > > > > Hello All. > > > > > > I'm using sems which I updated from berlios CVS tonight (2004-11-04). > > > > > > There appears to be a bug in the DTMF detection in the IVR plugin. > > > > > > Using the DTMF-detection.pl test script that ships with sems, DTMF > > detection works fine for > > > non-repeating digits. However, it always drops a digit if you repeat a > > number. > > > > > > For example: > > > > > > GOOD ===> Press 123456 and the IVR plugin will receive 123456 > > > BAD ===> Press 001122 the IVR plugin will receive 012 > > > > > > I've tested this with my grandstream BT100 using DTMF mode as > "in-audio", > > "via RTP", and "via > > > SIP > > > INFO". The "in-audio" and "via RTP" both produce this problem while the > > "via SIP INFO" digits > > > are > > > never detected (which is understandable). > > > > > > I searched the mailing list but found no reference to this problem. > > > > > > So the question is this; Does anyone have a patch to fix this? > > > > > > Regards, > > > Paul > > > > > > > > > > > > > > > > > > __________________________________ > > > Do you Yahoo!? > > > Check out the new Yahoo! Front Page. > > > www.yahoo.com > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > __________________________________ > > Do you Yahoo!? > > Check out the new Yahoo! Front Page. > > www.yahoo.com > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > __________________________________ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com

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[Serusers] Registering SER to Proxy.
by Tom Gaudasinski 05 Nov '04

05 Nov '04

Greetings, I have been reading a lot about SER lately, trying to figure out the intricacies, but cannot find out how to achieve one thing. I want to be able to get SER to register to another SIP Proxy(another SER?) to receive a given prefix. I'm trying to achieve a combination of Asterisk and SER, but the box I have SER running on must also be connected to another Proxy which is not under my control. I read that SER is a SIP Proxy which can also act as a registrar. It would seem logical that It is possible to get my SER box to register to the other SIP Proxy to tell it to forward the given prefix to me so that I can pass it off elsewhere. But how do I achieve this? Is it possible? All help is greatly appreciated. Thank you. Tom Gaudasinski

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[Serusers] problems about ser server's jabber and location
by C Y 05 Nov '04

05 Nov '04

hi everyone : there are some problems about ser server's jabber module and database location. 1.when i configure the jabber module.there are some errors about database sip_jab.However I cannot find the database sip_jab in the server.The configuration is modparam("jabber", "db_url", "sql://ser:helso@127.0.0.1/sip_jab) 2.there are two tables in databse ser:"server_monitoring" and "server_monitoring_agg".But when the server is running.There is no datas in the tables.Do the monitor infomations write into them? 3.In the database ser,there is a table location.However when a user log in.The field "domain" is still null.How can I write the domain infomation into the table "location"? How can i solve the problem? These are some informations related to my setting and problem: 1、operating system:Linux 7.0 2、SER distribution: ser-0.8.14_linux_i386.tar.gz 3、SER build: version: 0.8.14 (i386/linux) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.168.4.3 2004/06/28 15:41:21 andrei Exp $ main.c compiled on 12:28:01 Jul 27 2004 with gcc 2.95 4、SER configuration file : debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) check_via=yes # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=yes # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" alias="voipv6.edu.cn" "210.25.130.252" "localhost" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/jabber.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/vm.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc","db_url","sql://ser:heslo@localhost/ser") # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # -- auth params -- modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("jabber", "db_url", "sql://ser:helso@voipv6.edu.cn/sip_jab) modparam("jabber", "jaddress", "voipv6.edu.cn") modparam("jabber", "jport", 5222) modparam("jabber", "jdomain", "voipv6.edu.cn=*") modparam("jabber", "aliases", "1;msn.x.com=%") # ------------------------- 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=~"voipv6.edu.cn") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voipv6.edu.cn", "subscriber")) { www_challenge("voipv6.edu.cn", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } _________________________________________________________________ 免费下载 MSN Explorer: http://explorer.msn.com/lccn

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[Serusers] RE: Transfer and Conferencing - Please help!
by Bob Carlson 05 Nov '04

05 Nov '04

I sent this earlier and got no responses. Perhaps this is not the right forum to ask this question. Can any one suggest a better place to go for this information? Thanks, Bob -----Original Message----- From: Bob Carlson Sent: Wednesday, November 03, 2004 3:22 PM To: 'SerUsers' Subject: Transfer and Conferencing Let me apologize in advance for my question, which is a little rudimentary. We are just starting a project that will use SER and I am being forced to document right now how transfer and conferencing will be handled. I have spent a lot of time looking for definitive information on the subject with no luck. Well, maybe too much luck. There seem to be many proposals and models and so on, but it is not clear to me what is actually being done in practice. I have downloaded all the RFCs and proposal papers on the subject. I am still reviewing them, but I think the folks on this forum can help me a lot. I need to know the SIP message sequences for performing a call transfer and a blind call transfer and for constructing a conference. I have found information in proposals, but I need to know what actual, available SIP phones can do. We have some phones that we will test, but I do not know what they do when you press their transfer and conference buttons. Pardon me again for my impatience in asking before I have tried this out. The Transfer models are straightforward, but conferencing is more complicated. We must construct a simple conferencing model where the conferencing is performed by a central server, a SIP IPX. Only conferences of 3 participants need to be supported. We want it to look exactly like 3-way calling on your home phone. During a call, put the call on hold with a conference button, call another phone, hit conference button, the two calls are joined in a 3-way conference. The document draft-ietf-sipping-service-examples-07.txt seems to be very helpful on the subject, but all examples are in the form of 3 or more UAs and do not address any examples from the point of view of a PBX. I can see how to extend the examples to a PBX case, except for one aspect. If the IP-PBX is to perform the action as a proxy, what does the phone send the IP-PBX to indicate the steps in the process. Put more plainly, what happens when the user hits the Transfer or Conference button on the phone? What message is sent to the IP-PBX? Can anyone tell me where else I should be looking? Is the service examples draft the best base document to work from? Thanks in advance, Bob Carlson

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[Serusers] sems IVR plugin bug???
by Java Rockx 05 Nov '04

05 Nov '04

Hello All. I'm using sems which I updated from berlios CVS tonight (2004-11-04). There appears to be a bug in the DTMF detection in the IVR plugin. Using the DTMF-detection.pl test script that ships with sems, DTMF detection works fine for non-repeating digits. However, it always drops a digit if you repeat a number. For example: GOOD ===> Press 123456 and the IVR plugin will receive 123456 BAD ===> Press 001122 the IVR plugin will receive 012 I've tested this with my grandstream BT100 using DTMF mode as "in-audio", "via RTP", and "via SIP INFO". The "in-audio" and "via RTP" both produce this problem while the "via SIP INFO" digits are never detected (which is understandable). I searched the mailing list but found no reference to this problem. So the question is this; Does anyone have a patch to fix this? Regards, Paul __________________________________ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com

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Re: Re: Re: [Serusers] asterisk can not hangup .user Wildcard X100P
by dev2003 05 Nov '04

05 Nov '04

Flynn modules being loaded properly. -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' Nov 5 09:06:43 NOTICE[1118463168]: chan_zap.c:5055 ss_thread: Got event 2 (Ring/Answered)... Nov 5 09:06:45 NOTICE[1118463168]: chan_zap.c:5055 ss_thread: Got event 2 (Ring/Answered)... Nov 5 09:06:46 NOTICE[1118463168]: chan_zap.c:5055 ss_thread: Got event 2 (Ring/Answered)... -- Executing Goto("Zap/1-1", "default|12345|1") in new stack -- Goto (default,12345,1) -- Executing Ringing("Zap/1-1", "") in new stack -- Executing Answer("Zap/1-1", "") in new stack -- Executing AGI("Zap/1-1", "xml.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/xml.agi -- Playing 'agent-pass' (language 'en') -- xml.agi: The Input data is #,queuename is group2 -- xml.agi: User Input : # -- xml.agi: get in GetAgent group2 -- xml.agi: no agent is avail -- xml.agi: no agent -- Started music on hold, class 'default', on Zap/1-1 -- xml.agi: get in GetAgent group2 -- xml.agi: no agent is avail -- xml.agi: get in GetAgent group2 -- xml.agi: no agent is avail -- xml.agi: get in GetAgent group2 -- xml.agi: no agent is avail -- xml.agi: get in GetAgent group2 -- xml.agi: no agent is avail ........................................ --soft hangup zap/1-1 Requested Hangup on channel 'Zap/1-1' -- Stopped music on hold on Zap/1-1 == Spawn extension (default, 12345, 3) exited non-zero on 'Zap/1-1' -- Executing Hangup("Zap/1-1", "") in new stack == Spawn extension (default, h, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' // I have found from google it is common problem http://www.marko.net/asterisk/archives/0208/0329.html This has been covered in many other messages, but in order for the X100P to detect hangup, you must have disconnect supervision on the phone line. If the line you are providing is from another PBX, then it is highly unlikely that it supplies the disconnect supervision. Most telephone switches do support disconnect supervision, but it's not always on by default. You can tell by using a lighted keypad which receives its power only from the phone line, and then calling it and hanging up on it. If the lighted keypad blinks off for a moment then your line has the disconnect supervision, otherwise it doesn't. http://www.marko.net/asterisk/archives/0206/0198.html "ks" is the right one. Have you confirmed that your line supports disconnect supervision? Here is an easy test: Get a phone that has a lighted keypad, where the lighted keypad is powered ONLY by the phone line's power. Call the phone and then hang up on it. If the lighted keypad goes out for a brief time, then you do have disconnect supervision and we have to figure out how to tune the driver to see it. >why don't you try using the stable version of asterisk? right now i'm >using version 1.0RC2 (which is the version just before the actual >1.0.0) and it's been fantastic. > >What you're describing could be a whole bunch of things -- IRQ sharing >problems, faulty configuration files, etc. Why don't you attach a bit >more information, perhaps the zapata.conf file. > >are the appropriate modules being loaded properly? Try watching the >output of "dmesg" to see if there are any problems with the digium >card. > >Flynn > > >On Thu, 4 Nov 2004 10:08:34 +0800, dev2003 <dev2003(a)mail.ustc.edu.cn> wrote: >> Flynn, >> >> this sort of behaviour happenning all the time. >> such as when call 11,then I hangup. >> but when I recall 11,then it is busy. >> os redhat 9. >> /usr/sbin/asterisk -r >> Asterisk CVS-HEAD-09/10/04-21:34:12, Copyright (C) 1999-2004 Digium. >> Written by Mark Spencer <markster(a)digium.com> >> >> zaptel-0.9.0 >> >> >> >> >> >can you give more details about your asterisk settings? when is this >> >sort of behaviour happenning -- all the time? or only when a specific >> >condition occurs? >> > >> >flynn >> > >> >p/s including your config files wouldn't be a bad idea as well >> > >> > >> >On Wed, 3 Nov 2004 18:33:28 +0800, dev2003 <dev2003(a)mail.ustc.edu.cn> wrote: >> >> serusers，您好！ >> >> >> >> asterisk can not hangup .user Wildcard X100P. >> >> when using phone call,asterisk can not hangup. >> >> I should tpye: >> >> soft hangup zap/1-1 >> >> then can hangup. >> >> >> >> dev2003 >> >> dev2003(a)mail.ustc.edu.cn >> >> 2004-11-03 >> >> >> >> _______________________________________________ >> >> Serusers mailing list >> >> Serusers(a)iptel.org >> >> http://mail.iptel.org/mailman/listinfo/serusers >> >> >> >> >> >> >> >> = = = = = = = = = = = = = = = = = = = = >> >> 致 >> 礼！ >> >> >> dev2003 >> dev2003(a)mail.ustc.edu.cn >> 2004-11-04 >> >> >> = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　dev2003 　　　　　　　　dev2003(a)mail.ustc.edu.cn 　　　　　　　　　　2004-11-05

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[Serusers] serweb with PHP 5.0
by JOHN J. BARTON 04 Nov '04

04 Nov '04

FYI: I tried to run seweb with PHP 5.0.2. It sort of works, but at least one function that serweb relies upon has changed in v5: Warning: array_merge() [function.array-merge]: Argument #1 is not an array in /home/sipa/var/serweb/html/user_interface/my_account.php on line 286 More info on: http://www.php.net/manual/en/function.array-merge.php I switched to php 4.3 and all is well. John

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sr-users November 2004