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37554 discussions

[Users] uri or to_uri
by Gregoire 26 Oct '06

26 Oct '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi the list! I am just wondering the difference between the variables "uri" and "to_uri". I saw that "to_uri" refers to the value of the field "To", but I can not see at which field refers "uri". Does someone know something about that? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFQHp6I8gmGeMTr0sRAgbuAKCmISnqxdAvCQeWTsJjxtNoqoCY8gCeJA8c kW4+/zwvtIJXofKjTTVwrGA= =j0qD -----END PGP SIGNATURE-----

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[Users] BAD REQUEST
by Jose Gil Navarrete 26 Oct '06

26 Oct '06

Hi. My problem is the next. When my sip server send the message INVITE to the phone, the phone always respond with 400 Bad Request, I think is something wrong in the message, in the UDP message the Checksum always is corrupted by the sip server, is it the problem? or is something else in the sip message? How I can correct this? this is the trace: User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 7191 (7191) Source port: 5060 (5060) Destination port: 7191 (7191) Length: 1057 Checksum: 0x879f [incorrect, should be 0xb0cd] Session Initiation Protocol Request-Line: INVITE sip:192.168.1.8:7191 SIP/2.0 Method: INVITE Resent Packet: True Suspected resend of frame: 9 Message Header Record-Route: <sip:192.168.1.20;lr=on;ftag=db79a2e2> Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bKdfe7.2eeeaa64.0 Via: SIP/2.0/UDP 192.168.1.7:7191;branch=z9hG4bKN2IVzrueyM Max-Forwards: 69 Content-Type: application/sdp To: <sip:2109075@192.168.1.20> From: "4422109087" <sip:4422109087@192.168.1.20>;tag=db79a2e2 Call-ID: eea4130a-277bd7f8(a)192.168.1.7 CSeq: 1503273304 INVITE Allow: INVITE, ACK, CANCEL, MESSAGE, OPTIONS, BYE, UPDATE, REFER, SUBSCRIBE, NOTIFY, SERVICE User-Agent: PHONE-UserAgent Contact: "4422109087" <sip:4422109087@192.168.1.7:7191> P-hint: usrloc applied Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): ntzum 717537891 1011557901 IN IP4 192.168.1.7 Session Name (s): - Connection Information (c): IN IP4 192.168.1.7 Bandwidth Information (b): CT:29 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 49648 RTP/AVP 4 96 97 98 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:96 AMR/8000 Media Attribute (a): rtpmap:97 X-G723-7200/8000 Media Attribute (a): rtpmap:98 telephone-event/8000 Media Attribute (a): fmtp:98 0-15 Media Description, name and address (m): video 49650 RTP/AVP 99 100 34 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:99 H264/90000 Media Attribute (a): fmtp:99 packetization-mode=1 Media Attribute (a): rtpmap:100 H263-1998/90000 Media Attribute (a): rtpmap:34 H263/90000 Regards! --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Users] Mysql table version problem in cvs version
by Pletli Antal 26 Oct '06

26 Oct '06

Hi, I have a problem with openser cvs version. Before the installing the cvs version i dropped the previous openser database. I have installed the new version, and after make dbinstall the "version" table contains the following record for version silo: | silo | 5 | but the debug log shows this problem: 0(0) table_version(): Invalid type (2) or nul (0) version columns for silo 0(0) MSILO:mod_init: Wrong version v-1 for table <silo>, need v5 0(0) init_mod(): Error while initializing module msilo What did I wrong? Thanks in advance, Antal

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[Users] pbxnsip and openser
by Nick De Cristofaro 26 Oct '06

26 Oct '06

Anyone here using pbxnsip as their BB user agent with OpenSER. I'm wondering what steps would be required to make it work. Is it a complicated process? Thanks Nick

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[Users] Replication ACK problem
by Dioris Moreno 26 Oct '06

26 Oct '06

Hi everybody I have a working installation of OpenSER (Proxy A). In this installation, the ACK messages are delivery without any problem through loose_route() and t_relay(). Then, I installed a second OpenSER proxy (Proxy B) and configure both servers to replicate REGISTER messages with t_replicate(), and it works fine. I made a lot of captures to check the replication process was working fine and also to check the NAT pings; each proxy takes care of the NATed clients it receives sending OPTIONS message periodically. I'm using mediaproxy to support NATed clients; to avoid each proxy starting a media session when it find a NATed client I used the following rule: a proxy receiving messages from another proxy never starts a media session, the proxy that sends the message takes care of that. Having two registered clients (Client A in Proxy A and Client B in Proxy B), I start making calls from Client A to Client B, but I found something in the captures. All the messages flow correctly, except the final ACK message sent by Client A to Client B. Proxy A receives this message and pass it to Proxy B, but Proxy B never deliver it to Client B. Of course, the media session never starts. I was searching info about this situation and I found an email respond from Bogdan about using t_newtran() to process ACK messages, I tried, but it didn't work. Has anybody ever had the same problem? Is there a special way to deal with ACK messages when we're working with replicated proxies? I'll appreciate any help.

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[Users] Conference is Not Working.... with OpenSER And Asterisk
by sunkara 26 Oct '06

26 Oct '06

Hello Users, Good Morning, I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql modules. And I'm not Using the Zapptel Cards. 9001 ----------> dial 19001(conference Users)-------openSER ---------> Asterisk ------------------------------------------------------------------------ *In Extension.conf * [from-sip] exten => 19001,1,Playback(conf-hasentered) ;exten => 19001,2,Answer exten => 19001,2,Wait(2) exten => 19001,3,CBMysql() exten => 19002,1,Playback(conf-hasentered) ;exten => 19002,,Answer exten => 19002,2,Wait(2) exten => 19002,3,MeetMe(1234|pm) exten => 19003,1,Playback(conf-hasentered) ;exten => 19003,2,Answer exten => 19003,2,Wait(2) exten => 19003,3,MeetMe(1234|pm) ------------------------------------------------------------------------ * In cbmysql.conf* [global] hostname=localhost dbname=conference password= user=root port=3306 sock=/var/lib/mysql/mysql.sock DBOpts=yes ;OptsAdm=asdp ;OptsUsr=sdp ConfApp=MeetMe ConfAppCount=MeetMeCount ; Choose one of the following to modify early join behaviour earlyalert=300 ; Tell the participant if they are too early (seconds) ;fuzzystart= ; Allow participants to join early (seconds) ------------------------------------------------------------------------ * In sip.conf *[19001] type=friend username=9001 secret=august context=from-sip host=192.168.2.75 fromdomain=192.168.2.76 realm=192.168.2.75 ;mailbox=9001@from-sip insecure=very callerid="Ravi" <9001009> disallow=all allow=ulaw allow=gsm nat=yes ------------------------------------------------------------------------ 9001,9002,and 9003 is register from openSER, When They dial 19001 to enter the conference. Following are Showing the Errors and Warning .... *Executing Playback("SIP/9001-08f8d7e0", "conf-hasentered") in new stack -- Playing 'conf-hasentered' (language 'en') -- Executing Wait("SIP/9001-08f8d7e0", "2") in new stack -- Executing CBMySQL("SIP/9001-08f8d7e0", "") in new stack -- Playing 'conf-getconfno' (language 'en') Oct 25 18:15:47 NOTICE[12281]: app_cbmysql.c:373 cb_exec: getConf: 1 -- Playing 'agent-pass' (language 'en') Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags: Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:146 passQuery: CBMySQL: Invalid room or pass Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY: -- Playing 'auth-incorrect' (language 'en') Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags: Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:146 passQuery: CBMySQL: Invalid room or pass Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY: -- Playing 'auth-incorrect' (language 'en') Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags: Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY: Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:376 cb_exec: getPass: 1 == Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:97 checkMax: Currentusers: 0 Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:390 cb_exec: checkMax: 1 Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:276 enterConf: Roomtype: 1234|| == Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 WARNING[12281]: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Oct 25 18:16:13 ERROR[12281]: chan_zap.c:7396 chandup: Unable to dup channel: No such file or directory Oct 25 18:16:13 WARNING[12281]: app_meetme.c:460 build_conf: Unable to open pseudo channel - trying device Oct 25 18:16:13 WARNING[12281]: app_meetme.c:463 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') Oct 25 18:16:17 NOTICE[12281]: app_cbmysql.c:393 cb_exec: enterConf: -1 == Spawn extension (from-sip, 19001, 3) exited non-zero on 'SIP/9001-08f8d7e0' * Please Help me..... -- Thanks & Regards, *Ravi Prakash Sunkara* *M*:+91 9985077535* O*:+91 40 23114549* F*:+91 40 40208727 /*ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com */

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[Users] Changing call ID?
by Olle E Johansson 26 Oct '06

26 Oct '06

I got a bug report in our bug report tracker for Asterisk that claims that SER changes the call ID by adding a :port after the IP address. From reading the RFC, I don't think it's a valid change. For me, it's a new call ID. Questions: - Is changing a Call ID permitted within a dialog? - Does SER/OpenSER actually do this? http://bugs.digium.com/view.php?id=8220 See you at the Summit in Berlin! regards, /Olle

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[Users] Cvs snapshot Invalid table version
by Pletli Antal 26 Oct '06

26 Oct '06

Hi, How can I fix the "version" table to work with openser cvs snapshot? I have problem with grp,silo and location tables. I have installed the DB from the cvs too. +-----------------+---------------+ | table_name | table_version | +-----------------+---------------+ | subscriber | 5 | | missed_calls | 3 | | location | 1003 | | aliases | 1003 | | grp | 2 | | re_grp | 1 | | acc | 3 | | silo | 5 | | domain | 1 | | uri | 1 | | trusted | 3 | | usr_preferences | 2 | | speed_dial | 2 | | dbaliases | 1 | | gw | 3 | | gw_grp | 1 | | lcr | 2 | | sip_trace | 1 | | address | 2 | | domainpolicy | 2 | | presentity | 1 | | presence | 1 | +-----------------+---------------+ Antal

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[Users] Registration of Polycom SoundPointIP phone with OpenSER
by jeevan ravula 26 Oct '06

26 Oct '06

Hi all, I am using Polycom SoundPointIP phone as User Agent.I want to register Polycom phone with OpenSER(with TLS support) server.Can anybody help me out in this regard? I have generated my rootCA and given to polycom phone.The polycom phone does not accept certificate from openser server side.It shows bad certificate. anybody who has used polycom phone earlier can help me out in this matter.Ishall be greatful to them Regards, Jeevan.

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Re: [Serusers] Re: How to change Caller ID in FROM field ?
by lawrence k.y. lin 26 Oct '06

26 Oct '06

Hi there, This is an update for the Caller ID with Telco Carrier. The reason that Caller ID did not work on Verizon's mobile phone is because I only change the Display Name in the From Field. After I changed the From-URI (the field after the Displan Name which is <sip: XXXXXXXXXX@ip>), then Caller ID is displayed on Verizons mobile phone without any problem. Thanks for all that replied to this question. Best Regards, Larry >lawrence k.y. lin wrote: >Steve, > > Thanks for your email and your solutions. > > I tried the first method as follws: > >replace ("^From:(.*)<", "From: \"XXXXXXXXXX\" <") > > before I sent it to another SIP device and PSTN gateway. I did see the >new Caller ID XXXXXXXXXX in another SIP device but the Caller ID to Verison >Mobile phone showed "Unavailable No". I will check with Telco carrier to >see if we did send the Caller ID to them or not. > Also I will try your second method and update to the list later. > > Thanks again. > >Larry > >>From: Steve Blair <blairs(a)isc.upenn.edu> >>To: "lawrence k.y. lin" <lawlin888(a)hotmail.com> >>CC: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Re: How to change Caller ID in FROM field ? >>Date: Tue, 24 Oct 2006 18:57:42 -0400 >> >> >>You can do this several ways. I use to just completely replace the From >>header but this screwed up call accounting (for callers with caller ID >>blocking enabled) which we do in our Cisco gateways. Now I append an >>Remote-Party-ID header leaving the Fromheader intact for accounting >>purposes. Try one of the following: >> >>1) replace("^From:(.*)>" , "From: \"Anonymous\" >><sip:anonymous@invalid.nowhere>"); >> >> Where the second string contains whatever you want the caller ID info >>to be. This example is for caller ID blocking. >> >>2) >> >>if (append_rpid_hf("Anonymous >><sip:","@upenn.edu>;party=calling;id-type=subscriber;screen=yes")) { >>} else { >> log(1, "RPID not found\n"); >>}; >> >>Of course this requires other configuration items. You may also want to >>add the following: >> >>modparam("auth", "rpid_prefix", "<sip:") >>modparam("auth", "rpid_suffix", >>"@upenn.edu;user=phone>;party=calling;screen=no;id-type=subscriber;privacy=off") >> >># >> >>-Steve >> >>lawrence k.y. lin wrote: >> >>>Hi all, >>> >>> Appreciate if someone can give me a suggestion or even an answer. >>> Since the call from SIP device to PSTN phone did not have Caller ID >>>and our Telco carriers request for the Caller ID, we have to set a number >>>before the calls are routed to PSTN gateway. >>> I did it in Asterisk before with the function calls SetCallerID(CLID) >>>or Set(CALLERID(number)=CLID) (depends on the Asterisk version) and these >>>function calls worked. But because of the scalabliity issue, I changed to >>>SER and would like to do the similar thing as in Asterisk. >>> I read the SER document and awared SIP_HF_FROM has the data for the >>>entired FROM field. Is there any other variables for Caller ID only ? >>> I searched the mailing list for SER and did not find any related >>>topic (only found one with title: caller-id with raius using sip-rpid, >>>but it is irrelevant). >>> >>> Thanks in advance. >>> >>>Larry _________________________________________________________________ Try the next generation of search with Windows Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&source=…

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