sr-users August 2005

sr-users@lists.kamailio.org

220 participants
558 discussions

by Sam Lee

Try adding an alias in it. Eg. Alias=192.168.2.2 -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of han yibing Sent: Wednesday, August 03, 2005 10:08 AM To: serusers(a)lists.iptel.org Subject: [Serusers] serctl add user error hi serctl add 8800 8800 8800(a)192.168.2.2 using above command to add user ,but get following error ERROR:400;check if you use aliases in SER,in fact ,i don't use alisses in ser.cfg file. please help me. thanks ___________________________________________________________ 雅虎邮箱超强增值服务－2G超大空间、pop3收信、无限量邮件提醒 http://cn.mail.yahoo.com/mail_alert/promo1.html _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

18 years, 9 months

[Serusers] serctl add user error

by han yibing

hi serctl add 8800 8800 8800(a)192.168.2.2 using above command to add user ,but get following error ERROR:400;check if you use aliases in SER,in fact ,i don't use alisses in ser.cfg file. please help me. thanks ___________________________________________________________ 雅虎邮箱超强增值服务－2G超大空间、pop3收信、无限量邮件提醒 http://cn.mail.yahoo.com/mail_alert/promo1.html

18 years, 9 months

[Serusers] rtpproxy problem.

by Ray Van Dolson

I'm running into a problem with SER & rtpproxy trying to get a NAT scenario to work correctly. I'm still learning about SER, so bear with me... :) Here is my network layout in a nutshell: SIP ATA - 10.0.201.196 (private only, no NAT to internet) SER - eth0: 208.208.53.5/24 - eth1: 10.0.201.5/24 Asterisk - eth0: 65.234.34.5/24 *Everything* is configured to proxy through SER to Asterisk currently. My configuration file is here: http://webdev.digitalpath.net/~rayvd/voice/ser.cfg Here's the problem I'm running into: 1. SIP ATA places call (INVITE sent) 2. SER routes INVITE to Asterisk (and modifies c= in the SDP portion to point back to 208.208.53.5 as I understand it). 3. Ring, ring, other side answers. 4. Asterisk sends back a 200 OK message with c= 65.234.34.5 in its SDP message body. 5. My configuration changes this to 208.208.53.5 6. Voice packets now are outbound on SER eth1 with src ip of 208.208.53.5 and dst ip of 10.0.201.196. Obviously this will not work. >From what I understand however, rtpproxy appears to only use one IP address regardless of the direction traffic is going. If I bind rtpproxy to 10.0.201.5 on SER, then the IP in step 6 above will be correct, but the Asterisk server will be trying to send RTP traffic to 10.0.201.5 instead of 208.208.53.5. What am I doing wrong here? Can I run two instances of rtpproxy depending on the direction of the audio? I've tried different parameters to force_rtp_proxy() to no avail. Is there a function to manually change the c= line in the SDP body before it gets sent out? Thanks for any help! -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

18 years, 9 months

[Serusers] failure_route problem

by davor jovanovic

Hi all, I have a problem with sending a call to PSTN gateway when user is offline. I always get "404 Not found" what is normal because user is not in location database, but I don't understand why failure_route[1] doesn't work. Below is my ser.cfg so if someone could help. I'm using ser 0.9.3. Any help would be appreciated. Thanks __________________________________ 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; }; if (method=="REGISTER") record_route(); loose_route(); if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("srce.hr", "subscriber")) { www_challenge("srce.hr", "0"); break; }; save("location"); break; }; lookup("aliases"); # if (!uri==myself) { # append_hf("P-hint: outbound alias\r\n"); # route(1); # break; # }; if (method=="INVITE"){ t_on_failure("1"); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); route(2); break; }; }; if (!t_relay()) { sl_reply_error(); }; } ############ route[2] { # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # forward to gateway rewritehostport("161.53.2.235:5060"); t_relay_to_udp("161.53.2.235", "5060"); } failure_route[1] { revert_uri(); rewritehostport("161.53.2.235:5060"); append_branch(); t_relay_to_udp("161.53.2.235", "5060"); }

18 years, 9 months

[Serusers] lookup 2 uri

by Richard Z

Hi, I am trying to implement a feature to ring two ip phones simultaneously. For example, when I call 101(a)test.com, it rings both 101(a)test.com and 102(a)test.com. In this case, the ruri is 101(a)test.com. I can use append_branch 102(a)test.com, but is there a way to lookup the locations of both 101 and 102? Thanks, Richard

18 years, 9 months

[Users] avp questions

by Klaus Darilion

Hi! I'm playing with AVPs stored in a database and wonder about the following example found in the doc: avp_db_load("$from","i:678"); This loads all AVPs with attribute==678, regardless of the from URI. Is there a usecase for this? If not, why not use the specifier /uri as default if no specifier is selected. Also, in avp_db_load the source may be an avp-alias. Thus, how do you differ between the From: header URI "$from" and the avp-alias "$from"? regards, Klaus PS: Maybe I'm a little bit too academic ;-)

18 years, 9 months

[Users] about the next release

by Daniel-Constantin Mierla

The features we proposed (and many other) for next release are almost done. As you can see in the roadmap (http://openser.org/roadmap.php), mandatory items were achieved, a few optional are still to be worked on and other that would be good to have in next release popped up on the mailing lists. The major item which is still pending is the authentication via uac module. Another thing that should be there is the clean up of serctl + support for the other database types. Other topics discussed on mailing list: - error handling - to be done gradually - IP filters - some basic privacy support - user ACL (group membership improvements) All these are nice features, and they should be included in OpenSER as soon as possible. Anyhow, all the time will show up new good ideas, so at a moment we have to make a new release, and as OpenSER proposed to be, the releases should not make users to wait too long. Since there were many new features in development version, the proposed way to go is to start testing the development version as much as possible from now, even if we do not freeze the CVS right now. After just a few weeks freeze the CVS so we can have a fresh release somewhere in September. Meanwhile new features (some from listed above, and not only) may make their way into release, if they are fast to achieve. The more complex ones, will go probably in the second one. Details about what is new in the development version: http://openser.org/features-0.10.x.php http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/ChangeLog?view=mar… Any comment and suggestions are welcome. Let's start preparing a new release!

18 years, 9 months

[Serusers] Start and fnish call

by Anderson Alves de Albuquerque

With SER how can I know when the client SIP finish a call? I need that SER can start a script when the client SIP finish a call.

18 years, 9 months

[Serusers] URI rewrite utils

by Pavol Segeč

Hi, Ser has a few cmd which allows rewrite request uri or some part of it. Does SER have some cmd which allow rewrite caller uri or some part of it? For example, my uri is 1767(a)mydomain.com but when I'm calling outside of my VoIP world to PSTN (for example through Cisco call manager) I would like to identify myself by E.164 address format, i.e. +4214151341767(a)mydomain.com. It should be helpful for me change my URI. Thanks, palo

18 years, 9 months

[Serusers] MARK RODDY/LAKE is out of the office.

by mark.roddy＠lakecommunications.com

I will be out of the office starting 08/02/2005 and will not return until 08/18/2005. I will not be checking my mail. For Daidalos related issues please contact Stephen Butler. *********************************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify postmaster(a)lakecommunications.com This footnote also confirms that this email message has been scanned for the presence of computer viruses and other security threats. ***********************************************************************************

18 years, 9 months

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sr-users August 2005