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37517 discussions

[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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[Users] Re: Conference is Not Working.... with OpenSER And, Asterisk
by Paul Smith 25 Oct '06

25 Oct '06

This is an Asterisk config problem rather than an OpenSer one. The conference applications in Asterisk need a timing source. If you install a Digium or Sangoma card then Asterisk will use it as a timing source. As you are not using such a card you need to compile and modprobe a module called ztdummy to create a pseudo timing device.

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[Serusers] How do I allow gateway on ser to not ask for any authentication
by Lokesh Kumar 25 Oct '06

25 Oct '06

Hello , How do I configure my ser and radius so that it won’t ask for credentials from PSTN gateway, Please help Thanks very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 - Release Date: 10/24/2006

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[Serusers] Re: Asterisk Manager
by Maps 25 Oct '06

25 Oct '06

Dear Friends and Supporters! I try to write a php application to monitor the asterisk, but when I open the .php to access to asterisk to retrieve the information about the queues status, sip show peers, realtime mysql status etc... However, It just return to me "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and start asterisk with user is asterisk. HTTPD is run under asterisk user. My manager.conf has an entry. [admin] secret = password deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user However, my php still unable to retrieve the information for asterisk. Did I miss somethings? Your help would be very appreciated! Regards, Lan

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[Serusers] MySQL has gone away
by Mark Price 25 Oct '06

25 Oct '06

Hi, I'm getting the MySQL has gone away error. I have set the following in ser.cfg: modparam("mysql", "auto_reconnect", 1) modparam("mysql", "ping_interval", 60) Restart fixes this, but that is of course not sufficient. I suppose I can detect every time that error happens with swatch or some such thing and restart the service. This seems to be a common error. Is there a common solution? Mark

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[Serusers] MySQL confusion
by Chris Parkinson 25 Oct '06

25 Oct '06

Hello, I'm a complete newbie to SER and I'm having a problem following the MySQL section of the "getting started" guide. Whenever I try to use serctl to add a user I get access denied; LabServ1:/home/chris# export SIP_DOMAIN=mylab.com LabServ1:/home/chris# serctl add test1 test1pass test1(a)mylab.com MySql password: ERROR 1045: Access denied for user: 'serro@localhost' (Using password: YES) error: SER/FIFO not accessible: 2 LabServ1:/home/chris# I think this may be down to the MySQL install. I've installed MySQL and tested it, however I can't find any documentation anywhere to show me what users, tables and databases to setup for SER. I'm using SER on Debian. It's quite possible I'm missing something obvious. I'd appreciate a pointer in the right direction. Chris

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[Users] X-lite is not Working, But SipEyeMedia Softphones is Working Fine.
by sunkara 25 Oct '06

25 Oct '06

Hello Users, I don't Know What Happening in this Issues... Help me in this . When Testing the X-lite Softphones, is not working. When I use the Sip EyeMedia Softphone it Working Fine. Is Problem in X-lite Configuration or Router(Firewall Gateway Keepers ) is blocking it... 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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[Serusers] Using different domains
by Nick Hoffman 25 Oct '06

25 Oct '06

Hi guys. I just noticed that when user agents register, the domain in their SIP URI does not have to equal their domain in the subscribers database table. For instance, I have 2 domains in the domain table: sip.altcall.com sip.voxpak.com The username "foobar" in my subscribers table has the domain "sip.altcall.com", but can register and make calls as foobar(a)sip.voxpak.com . Why is this? Cheers, -- Nick E: nick.hoffman(a)voxpak.com P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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