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Re: [Serusers] Looking for SER + Asterisk-as-voicemail HOWTO
by Ashling O'Driscoll 29 Jan '05

29 Jan '05

Thanks for the reply Girish. I would just like to verify the following: I currently have the following very simple config: if (method == "INVITE") { //other code t_on_failure("1"); }; failure_route[1]{ revert_uri(); rewritehostport("xxx.xxx.xxx.xxx:5062"); append_branch(); t_relay_on_udp("xxx.xxx.xxx.xxx","5062"); break; } However when I restart SER it tells me I have an error in my ser.cfg and from the log file it says I am missing a loadmodule: The following are the load modules I have inclued. Am I missing one? sl.so tm.so rr.so maxfwd.so usrloc.so registrar.so nathelper.so textops.so auth.so authdb.so Thanks, Aisling. ---- Original Message ---- From: gr_sh2003(a)yahoo.com To: ashling.odriscoll(a)cit.ie, serusers(a)iptel.org Subject: Re: [Serusers] Looking for SER + Asterisk-as-voicemail HOWTO Date: Sat, 29 Jan 2005 00:39:19 -0800 (PST) >Ashling, > >See comments inline. > >--- Ashling O'Driscoll <ashling.odriscoll(a)cit.ie> wrote: >> >> I implemented the changes in ser.cfg that Giri suggested below(will >> this cover no answer and busy?). Asterisk and ser are on the same >> machine so I now forward to 5062: see code below. Hoever how do I >now >> modify Asterisk to handle this?...Like I said Ive read lots on this >> but it gets complicated with perl scripts etc. Apologies if I am a >> bit slow picking up on this! > >Go thru the README.variables file that comes with the Asterisk >distribution. It describes about >the extension logic and how to handle different variables. Also, >learn about pattern matching for >Asterisk extensions. Now add appropriate entries for voicemail in >voicemail.conf. Have something >like this in your extensions.conf when the vm calls from SER reach >Asterisk. > >exten => _X.,1,VoiceMail2(u${EXTEN}) > >> >> ser.cfg; >> if(!lookup("location"){ >> rouet(2); >> break; >> }; > > >The above code will not serve your busy/unanswered needs. Add a >failure route and handle it there. >like: > >t_on_failure("1"); >t_relay (); >break; > >failure_route[1] { > route(2); > break; >} > >> route [2] >> { >> rewritehostport("xxx.xxx.xxx.xxx:5062"); >> t_relay_to_udp("xxx.xxx.xxx.xxx:5062"); >> break; >> }; >> >> Aisling > >Also, learning about using t_check_status will also help you handle >specific error conditions. > >Best, > >===== >Girish Gopinath <gr_sh2003(a)yahoo.com> > >__________________________________________________ >Do You Yahoo!? >Tired of spam? Yahoo! Mail has the best spam protection around >http://mail.yahoo.com > >-------------------Legal >Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. > -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 1

[Serusers] SERWeb latest version: DB tables resp. fields not found....
by Alexander Philipp Lintenhofer 29 Jan '05

29 Jan '05

I installed ser09 with mysql.so and acc.so and tried to implement serweb (latest version fetched by CVS this night). Unfortunately, the generated SQL statements do not fit to the actual implementation of the database: log errors for example: Unknown column 't1.caller_deleted' in 'where clause' Table 'ser09.uuidaliases' doesn't exist I use Slackware 10 with kernel 2.6.10, PHP 4.3.9 (5.0.3 did not work with serweb) and MySQL 4.1.8 and I know that the DB-model of ser09 differs from the model of ser0814. Where can I therefore get the latest version of the DB-model ? kind regards and thank you for your help! Philipp

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[Serusers] SERWEB problem - user_interface
by Patrick Baker 29 Jan '05

29 Jan '05

Hello all, I'm using serweb CVS rel_0_8_12 and SER ser-0.8.14. SER works fine and serweb admin works - but if I try to click account in the admin panel it hangs - or if I try to use the user_interface it will hang on login. Has anyone had a similar problem? Reg Globals is On ect... .followed the inst in the INSTALL file. Thanks, Patrick

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FW: Re: [Serusers] SER and NAT and RTProxy
by Ashling O'Driscoll 29 Jan '05

29 Jan '05

After doing a small bit of testing I have a bit more information which may help identify the problem....Basically I think the problem comes down to nat. I still get the rtp errors mentioned below in the var/log/messages file whether voice works or not(strangely enough)...but here are the scenarios that work and dont work: I have ser and asterisk on a private natted network. They are reached via a router which has a public address and does port forwarding. One my clients(a bt100 hardphone is also on this lan but registers with ser throught the public address anyway...so my understanding is that the bt100's sip register message goes out onto the internet and back in again. Now if the bt100 rings a client (e.g. xlite or windows messenger)or vice versa that are also on this lan...the call works and voice is transmitted (even though they again register through the public address by routing their sip messages onto the internet and back in again). However if the bt100 tries to ring a client not on the lan (on another network which may or may not be behind nat)...np voice is transmitted....Surely this is a nat problem even through the rtp errors are being displayed??.... If anybody has any idea how to fix this or if any more information is required in order to troubleshoot please let me know. Like I mentioned before this must be demoed next week...This natting scenarion worked when clients registered direct to asterisk so i presume its possible with ser. Thanks in advance, Aisling. ---- Original Message ---- From: ashling.odriscoll(a)cit.ie To: serusers(a)iptel.org Subject: FW: Re: [Serusers] SER and NAT and RTProxy Date: Sat, 29 Jan 2005 13:58:14 -0000 Unfortunately this still hasnt worked. I changed the modparam line in ser.cfg and ran rtpproxy again as follows: cd /root/Desktop/rtpproxy ./rtpproxy -s 127.0.0.1 Then I restarted SER. However now voice isnt being transmitted AT ALL...and i cant make it work even by running: cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsrot/ser co rtpproxy Whats so weird is that I tested all this on wed evening and everything worked fine, now I repeatedly get the rtp errors documented in my first email on the /var/log/messages file. Does anyone have any more ideas...Im supposed to be demonstrating this sytem as prototype next week and it wont look very impressive if voice wont transmit.... Very stumped, Aisling. ---- Original Message ---- From: info(a)marikar.com To: serusers(a)iptel.org Subject: Re: [Serusers] SER and NAT and RTProxy Date: Sat, 29 Jan 2005 02:30:43 +0100 >-----BEGIN PGP SIGNED MESSAGE----- >Hash: SHA1 > >Hello, > >I got the same error-messages. The rtpproxy works fine if I change >the line >modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") >to modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1") and start >the >rtpproxy with the option -s 127.0.0.1 > >I hope this will help, > >Achim > >Am Donnerstag, 27. Januar 2005 22:01 schrieb Ashling O'Driscoll: >> Hi all, >> >> I have a strange problem with the audio with some calls. I have >setup >> RTPProxy and nathelper modules. I came across an error saying the >RTP >> proxy was disabled but solved it by searching the archives and >> executing: >> >> cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co >rtpproxy. >> >> This worked fine and my audio was transmitted. However every now >and >> again (apparently at random) my audio doesnt work.When I look at >the >> error logs in /var/log/messages, I see the following: >> >> ERROR: send rtpp_command: cant read reply from a rtp proxy >> WARNING: rtpp_test: cant get version of the RTP proxy >> WARNING: rtpp_test: support for the rtp proxy has been temporarily >> disabled >> ERROR: force_rtp_proxy2: support for porxy disabled. >> >> The CVS command fixes it temporarily. Does this mean I just have to >> run the command randonly every so often?...Is there a way to >> permanently fix this? >> >> Thanks, >> Aisling. >> >> >> -------------------Legal >Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >only >> for the person to whom it is addressed. Its contents may be >protected by >> legal and/or professional privilege. Should it be received by you >in error >> please contact the sender at the above quoted email address. Any >> unauthorised form of reproduction of this message is strictly >prohibited. >> The Institute does not guarantee the security of any information >> electronically transmitted and is not liable if the information >contained >> in this communication is not a proper and complete record of the >message as >> transmitted by the sender nor for any delay in its receipt. >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)iptel.org >> http://mail.iptel.org/mailman/listinfo/serusers >-----BEGIN PGP SIGNATURE----- >Version: GnuPG v1.2.4 (GNU/Linux) > >iQEVAwUBQfrnScahenr7DhiKAQLpsQf8D8Nc9CGnvjHqfBRLqV984qgH+kE7wcPL >rN2IYNHDiA7NQRrbymJqeyiiJzgVywozKsEeDjBH+ssSNoTMISMPXFAjHdqoR6kR >W0hZB68viCl0HhxSU/fJy2+kE91+JVicNyfLZDQ9sV2muTanBo/qnpx0sev4WPey >lXEyNtwfyNP035IXUOEZg1mZQ7bWRx9EI9LBIyaQnTWqgbfPIv5S5R6XeXpcdNHb >ly0jNUWel8vr3ByzFsbbIqyxaScXWcfD027lqg78XoVvlD6JJseSIFhKgEKqOjpL >7CfngqEkSaxqampbU4BDpoU4gHNqCr2MyYoajyyD5/Ud/idC5wmJGQ== >=IKFr >-----END PGP SIGNATURE----- > >-------------------Legal >Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. > -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 1

Re: [Serusers] SER and NAT and RTProxy
by Ashling O'Driscoll 29 Jan '05

29 Jan '05

Unfortunately this still hasnt worked. I changed the modparam line in ser.cfg and ran rtpproxy again as follows: cd /root/Desktop/rtpproxy ./rtpproxy -s 127.0.0.1 Then I restarted SER. However now voice isnt being transmitted AT ALL...and i cant make it work even by running: cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsrot/ser co rtpproxy Whats so weird is that I tested all this on wed evening and everything worked fine, now I repeatedly get the rtp errors documented in my first email on the /var/log/messages file. Does anyone have any more ideas...Im supposed to be demonstrating this sytem as prototype next week and it wont look very impressive if voice wont transmit.... Very stumped, Aisling. ---- Original Message ---- From: info(a)marikar.com To: serusers(a)iptel.org Subject: Re: [Serusers] SER and NAT and RTProxy Date: Sat, 29 Jan 2005 02:30:43 +0100 >-----BEGIN PGP SIGNED MESSAGE----- >Hash: SHA1 > >Hello, > >I got the same error-messages. The rtpproxy works fine if I change >the line >modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") >to modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1") and start >the >rtpproxy with the option -s 127.0.0.1 > >I hope this will help, > >Achim > >Am Donnerstag, 27. Januar 2005 22:01 schrieb Ashling O'Driscoll: >> Hi all, >> >> I have a strange problem with the audio with some calls. I have >setup >> RTPProxy and nathelper modules. I came across an error saying the >RTP >> proxy was disabled but solved it by searching the archives and >> executing: >> >> cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co >rtpproxy. >> >> This worked fine and my audio was transmitted. However every now >and >> again (apparently at random) my audio doesnt work.When I look at >the >> error logs in /var/log/messages, I see the following: >> >> ERROR: send rtpp_command: cant read reply from a rtp proxy >> WARNING: rtpp_test: cant get version of the RTP proxy >> WARNING: rtpp_test: support for the rtp proxy has been temporarily >> disabled >> ERROR: force_rtp_proxy2: support for porxy disabled. >> >> The CVS command fixes it temporarily. Does this mean I just have to >> run the command randonly every so often?...Is there a way to >> permanently fix this? >> >> Thanks, >> Aisling. >> >> >> -------------------Legal >Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >only >> for the person to whom it is addressed. Its contents may be >protected by >> legal and/or professional privilege. Should it be received by you >in error >> please contact the sender at the above quoted email address. Any >> unauthorised form of reproduction of this message is strictly >prohibited. >> The Institute does not guarantee the security of any information >> electronically transmitted and is not liable if the information >contained >> in this communication is not a proper and complete record of the >message as >> transmitted by the sender nor for any delay in its receipt. >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)iptel.org >> http://mail.iptel.org/mailman/listinfo/serusers >-----BEGIN PGP SIGNATURE----- >Version: GnuPG v1.2.4 (GNU/Linux) > >iQEVAwUBQfrnScahenr7DhiKAQLpsQf8D8Nc9CGnvjHqfBRLqV984qgH+kE7wcPL >rN2IYNHDiA7NQRrbymJqeyiiJzgVywozKsEeDjBH+ssSNoTMISMPXFAjHdqoR6kR >W0hZB68viCl0HhxSU/fJy2+kE91+JVicNyfLZDQ9sV2muTanBo/qnpx0sev4WPey >lXEyNtwfyNP035IXUOEZg1mZQ7bWRx9EI9LBIyaQnTWqgbfPIv5S5R6XeXpcdNHb >ly0jNUWel8vr3ByzFsbbIqyxaScXWcfD027lqg78XoVvlD6JJseSIFhKgEKqOjpL >7CfngqEkSaxqampbU4BDpoU4gHNqCr2MyYoajyyD5/Ud/idC5wmJGQ== >=IKFr >-----END PGP SIGNATURE----- > >-------------------Legal >Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. > -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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[Serusers] Looking for SER + Asterisk-as-voicemail HOWTO
by Ashling O'Driscoll 29 Jan '05

29 Jan '05

I am having similar problems to those described below. I have scoured the internet with google and looked at the following link aswell: http://mail.iptel.org/pipermail/serusers/2004-October/012192.html Howvere I am getting confused because most of these configs refer to the pstn aswell. I would just like to know how SER can use asterisk for voicemail. I implemented the changes in ser.cfg that Giri suggested below(will this cover no answer and busy?). Asterisk and ser are on the same machine so I now forward to 5062: see code below. Hoever how do I now modify Asterisk to handle this?...Like I said Ive read lots on this but it gets complicated with perl scripts etc. Apologies if I am a bit slow picking up on this! ser.cfg; if(!lookup("location"){ rouet(2); break; }; route [2] { rewritehostport("xxx.xxx.xxx.xxx:5062"); t_relay_to_udp("xxx.xxx.xxx.xxx:5062"); break; }; Aisling Date: Thu, 27 Jan 2005 22:51:40 -0800 (PST) From: Girish <gr_sh2003(a)yahoo.com> Subject: Re: [Serusers] Looking for SER + Asterisk-as-voicemail HOWTO To: Felipe Martins <fmartins(a)mundivox.com>, serusers(a)iptel.org, hankipanky(a)gmail.com Message-ID: <20050128065140.14086.qmail(a)web54005.mail.yahoo.com> Content-Type: text/plain; charset=us-ascii Hello, You can run both SER and Asterisk in the same machine with different ports. Add a failure route in your ser.cfg and handle calls to voicemail in that. A simple one is given below. You might want to add necessary conditions there. See the SER documentation for more details. http://www.iptel.org/ser/doc/seruser/seruser.html route [2] { rewritehostport ("Asterisk IP: Port"); t_relay (); break; } Cheers! --- Felipe Martins <fmartins(a)mundivox.com> wrote: > Hi Hank, > > Among all the other services I want it to do, one of them is voicemail. I've tried to find some > easy howto at the web, but find nothing. I'm trying to install it by the howto I've found at > www.voip-info.org. I'm still trying to make it work. > Any advance, please let me know, cause I'm gonna do the same . :o) > > Best Regards. > Felipe Martins > > > On Thu, 27 Jan 2005 12:00:14 +0100 > Henning Verbeek <hankipanky(a)gmail.com> wrote: > > > Hi all, > > > > I am sure that this has been talked about many times, i just can't > > seem to find a good Howto on this via Google. > > > > I am using SER 0.8.14 as Registrar / Proxy and would like to use > > Asterisk as a pure Voicemail service. > > - Does this make sense? Or should i use the new VM code in 0.9.0 > > (which would require SEMS, right)? > > - Can I run them on the same machine (e.g. SER on :5060, Asterisk on > > :5080 and a forward() in between)? > > - How do I forward from SER to Asterisk? Forward? or rewrite the URI? > > > > Any help is much appreciated! > > Cheers, Hank ===== Girish Gopinath <gr_sh2003(a)yahoo.com> -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 1

[Serusers] No response from client
by support 29 Jan '05

29 Jan '05

Hi everyone, I am running ser-0.8.14 with nathelper and rtpproxy on the same box. When I try to dial a number from a client, I need to wait for a long time (20s) before I cannot hear the dial tone. When I try to using "serctl ping" to ping the client, it also takes more than 20s to wait for client response. What is the cause of this problem? Thomas

1 0

[Serusers] SER and NAT and RTProxy
by Ashling O'Driscoll 29 Jan '05

29 Jan '05

Hi all, I have a strange problem with the audio with some calls. I have setup RTPProxy and nathelper modules. I came across an error saying the RTP proxy was disabled but solved it by searching the archives and executing: cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co rtpproxy. This worked fine and my audio was transmitted. However every now and again (apparently at random) my audio doesnt work.When I look at the error logs in /var/log/messages, I see the following: ERROR: send rtpp_command: cant read reply from a rtp proxy WARNING: rtpp_test: cant get version of the RTP proxy WARNING: rtpp_test: support for the rtp proxy has been temporarily disabled ERROR: force_rtp_proxy2: support for porxy disabled. The CVS command fixes it temporarily. Does this mean I just have to run the command randonly every so often?...Is there a way to permanently fix this? Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 1

[Serusers] AddPac-no media
by Jain, Hitesh 28 Jan '05

28 Jan '05

Hi All, I have an AddPac ATA in Public calling a Cisco phone which is also in public domain. The call goes through but there is media only in one direction (from Cisco to AddPac). No media in reverse direction. Actually no packets come out from the AddPac ATA. But when I make a call from Cisco to AddPac the call goes through and there is media in both direction. This time packets do come out of the AddPac box. Is there any configuration parameter in AddPac which need to be set. If anyone has any clue plz let me know. Thanks, Hitesh.

1 0

[Serusers] SER Gateway Authentication
by Tom Gaudasinski 28 Jan '05

28 Jan '05

Greetings, I'm looking for a way to authenticate SER to an Asterisk box securely, digest would be the best but whatever is available I'll take. I cannot seem to find anything about this at voip-info, and I've had a solid google through the mailing archives. First of all, is it at all possible to authenticate with a gateway that wants to place a call through SER and vice-versa for that matter? If so, how could I achieve this? Eventually, I'd like to have hardware gateways such as cisco to coexist with SER, is it possible to authenticate those? Can SER initiate the authentication or does the Asterisk/Gateway have to initiate it? If it is also possible I would truly appreciate some examples of both how to achieve this in the ser.conf as well as in asterisk configs. Thankyou.

1 0

[Serusers] Compiling Source From 8.14
by Mr B 28 Jan '05

28 Jan '05

I have tried to compile the source from 8.14 using gmake on a fedora core 3 .. However I get a errors of files not found . I corrected this just the path issue .. However now the size of the ser executable and some of the modules are different sizes then the actual release . I don't understand that .. Could someone help me out ? When I do it is compiled and I run it the ser.cfg is comes up in gedit for debugging as apposed in the command shell like it did before. I am trying to recompile so as to include the mysql support in the acc module for accounting. Your help would be greatly appreciated. thanks

1 0

[Serusers] Straight MySQL or text CDR?
by Doug Eubanks 28 Jan '05

28 Jan '05

Is it possible to do straight CDR without using radius? I have gotten everything going that I need so far, except CDR. I am having a heck of a time getting FreeRadius to play nicely with SER, and I have been using FreeRadius for over a year and half. I get stop packets fine, and some start packets, however I always get 0 for the session time. Also, the more I think about it, FreeRadius is just another point of failure since I am not using RADIUS based auth. I am using SER 0.9.0 Can someone point me in the direction I need to go? Doug

1 0

[Serusers] Recommended rtpproxy version
by Luca Corti 28 Jan '05

28 Jan '05

Hello, I'd like to know which version of rtpproxy is rencommended for use with ser 0.8.14 and where to get it. thanks

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[Serusers] Looking for SER + Asterisk-as-voicemail HOWTO
by Henning Verbeek 28 Jan '05

28 Jan '05

Hi all, I am sure that this has been talked about many times, i just can't seem to find a good Howto on this via Google. I am using SER 0.8.14 as Registrar / Proxy and would like to use Asterisk as a pure Voicemail service. - Does this make sense? Or should i use the new VM code in 0.9.0 (which would require SEMS, right)? - Can I run them on the same machine (e.g. SER on :5060, Asterisk on :5080 and a forward() in between)? - How do I forward from SER to Asterisk? Forward? or rewrite the URI? Any help is much appreciated! Cheers, Hank -- My other signature is a regular expression. http://www.pray4snow.de

3 2

[Serusers] help!!!!
by zhang_zhaoxin＠pact518.hit.edu.cn 28 Jan '05

28 Jan '05

hi: I'm so glad to write to you. I have some question on using ser. the ua that I use is sjphone. Everyone can use the ser but unregistered. Ser don't confirm the user'username and password. I have configured ser.cfg like this loadmodule "/usr/local/lib/ser/modules/dbtext.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") I don't know why it no work well! Would you like help me. Thank you very much! 姓名：张兆心 email: zhang_zhaoxin(a)pact518.hit.edu.cn

1 0

[Serusers] ser 0.9 config
by Steve Blair 27 Jan '05

27 Jan '05

Does anyone have a working example of a config file for ser 0.9 that they can share. My 0.8.14 config keeps failing. The line numbers in the debug output indicate the problem is on a line that is blank. The NEWS file is no help. Thanks -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu

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[Serusers] Comipiling problems 8.14.0
by Mr B 27 Jan '05

27 Jan '05

I am trying to recompile from the src using the 8.14 src code located in the released version of 8.14, but am having a lot of problems with files not found so it won't compile properly. Has anyone else had problems with this ? I have also tried to compile the one from the cvs this one however even gives me greater errors. I would like to compile it with option for logging the acc transactions into mysql database . which is not turned compiled by default ( why this would be I don't know . should be one with mysql and one with the radius support) . but any way if someone could shed some light this would be nice or even nicer yet a compiled version for Fedora Core 3 The link below mentions the problem that someone else was also having with similar problems but no resolution that i could find. http://lists.iptel.org/pipermail/serusers/2005-January/014970.html Thanks Mr. B

1 0

[Serusers] Email Test not sure if it bouncing
by Mr B 27 Jan '05

27 Jan '05

Sorry for this post I just need to see if my emails are getting to the group .. I see other folks after I sent mine. Thanks for the patience Mr. B

1 0

[Serusers] control SER with D.J. Bernstein's daemontools
by u2 27 Jan '05

27 Jan '05

<http://cr.yp.to/djb.html>Hello, did somebody success to control a SER proxy with D.J. Bernstein's daemontools? As soon as SER is run, it puts itself to the shell background. As far as I found out, this behavior prevents the daemontools from controlling SER. What did anybody already experiment with this toolkit? What do you use to control you SER processes (auto-respawn, logging, etc.)? It would be also great to use Bernstein's multilog to collect SER logs in a comfortable manner (with per instance log, log rotation, timestamps, etc). What do you use here? Any recommendation? Best regards, Urs Uschmann

3 2

Re: [Serusers] chosing the right Via header in multihomed SER instance
by Samuel Osorio Calvo 27 Jan '05

27 Jan '05

Thanks! We just tried and it worked....just a "small" missing detail. Samuel Unclassified. >>> Jan Janak <jan(a)iptel.org> 01/27/05 06:55PM >>> Try to use mhomed=yes in the configuration file of SER_1. It should then put the IP of the outgoing interface (unicast) into Via. Jan. On 25-01 10:30, Samuel Osorio Calvo wrote: > Hi, > > I have received this and I don't know what's wrong, maybe any guru can > help: > > Samuel. > > Unclassified. > >In my test lab i found that SER adds a wrong VIA header. > >The scenario is like this: > >phone1 ---- SER_1 ----- SER_2 ---- phone2 > > > >phone 1 is registered at SER_1 (it's proxy) > >phone 2 is registered at SER_2 (it's proxy) > > > >When phone2 wants to call phone1, it sends the invite to SER2. SER2 > checks in their local database for phone1, and does not find it. SER2 > forwards the invite message on (using >multicast address 230.x.x.x). > SER1 is listening on two addresses: unicast one (192.y.y.y) and > multicast one (230.x.x.x). He receives the invite from the multicast > address, checks for >phone1 and finds it registered in his local > database. SER1 relays (t_relay) to phone one. Here is where the error is > introduced: SER1 adds a VIA header to the message relayed to >phone1, > but instead of this header containing the SER1_unicast address, it > contains the multicast one. This makes phone1 to then reply and send > messages to the multicast >address (taken from the via in the invite), > instead of following the recourd-route setting (for the OK message it > does follow it, not for ACK or 1xx). > > > >Any idea on what is wrong? > >I read in the code something about support for multi-homed proxies, as > well as a parameter called default_global_address. Might these help? > > > >Thanks in advance! > > > >Cesc > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] chosing the right Via header in multihomed SER instance
by Samuel Osorio Calvo 27 Jan '05

27 Jan '05

Hi, I have received this and I don't know what's wrong, maybe any guru can help: Samuel. Unclassified. >In my test lab i found that SER adds a wrong VIA header. >The scenario is like this: >phone1 ---- SER_1 ----- SER_2 ---- phone2 > >phone 1 is registered at SER_1 (it's proxy) >phone 2 is registered at SER_2 (it's proxy) > >When phone2 wants to call phone1, it sends the invite to SER2. SER2 checks in their local database for phone1, and does not find it. SER2 forwards the invite message on (using >multicast address 230.x.x.x). SER1 is listening on two addresses: unicast one (192.y.y.y) and multicast one (230.x.x.x). He receives the invite from the multicast address, checks for >phone1 and finds it registered in his local database. SER1 relays (t_relay) to phone one. Here is where the error is introduced: SER1 adds a VIA header to the message relayed to >phone1, but instead of this header containing the SER1_unicast address, it contains the multicast one. This makes phone1 to then reply and send messages to the multicast >address (taken from the via in the invite), instead of following the recourd-route setting (for the OK message it does follow it, not for ACK or 1xx). > >Any idea on what is wrong? >I read in the code something about support for multi-homed proxies, as well as a parameter called default_global_address. Might these help? > >Thanks in advance! > >Cesc

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[Serusers] SER crash !!! ( Problem with q_malloc ???? )
by PHAN, Quang-Minh (Docteur Ordinateur) 27 Jan '05

27 Jan '05

Hi all, I'm running SER 0.9 on an FEDORA CORE 3 machine and it crashes every time I try to use mysql module. I have compile SER in mode debug and spent some time trying to figure it out. I first found a bug in the q_find_free function in q_malloc.c:290: "if (f->size == size)" which cause memory access violation when f is NULL. I fixed this but it didn't solve the problem. I attached here the coredump and the ser.cfg file. The version of MySQL Server and libraries is 4.1 Any helps will be highly appreciated, Quang

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[Serusers] Having make all issues 8.14 src
by Mr B 27 Jan '05

27 Jan '05

I am trying to recompile from the src using the 8.14 src code located in the release version, but am having a lot of problems with files not found so it won't compile properly. Has anyone else had problems with this ? I would like to compile it with option for logging the acc transactions into mysql database . which is not turned compiled by default ( why this would be I don't know . should be one with mysql and one with the radius support) . but any way if someone could shed some light this would be nice or even nicer yet a compiled version for Fedora Core 3 Thanks b

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[Serusers] speedial module bug - not honoring use_domain modparam
by Java Rockx 27 Jan '05

27 Jan '05

Hi All. I think there may be a bug in the speeddial module with ser-0.9 I have this as my modparam modparam("speeddial", "use_domain", 0) Yet in my MySQL server logs I see ser issuing the following SQL: select new_uri from speed_dial where username='4071112222' AND domain='mycompany.com' AND sd_username='01' Given the fact that my modparam says not to use the domain shouldn't the sql be as follows? select new_uri from speed_dial where username='4071112222' AND sd_username='01' Is this a bug? Regards, Paul __________________________________ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250

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[Serusers] Accounting and missed calls..
by Amos Nungu 27 Jan '05

27 Jan '05

Hi all, Help needed. I'm running SER 0.8.14 (i386/linux) with serweb_2004-07-27 on a debian machine. I need to achieve the missed calls and accounting... Be able to see them from my serweb interface which I can't for now. I can register, authenticate and make calls normally. I tried to check from the database i.e without serweb, nothing is stored also. Here is my ser.cfg file: ................................ser.cfg..................... # # config script + accounting + misseed calls # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) modparam("usrloc", "db_mode", 2) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- acc params -- modparam("acc", "log_level", 1) #Accounting modparam("acc", "log_flag", 1 ) #Accounting modparam("acc", "log_missed_flag", 3 ) #missed calls # ------------------------- 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; }; # record-route INVITES to make sure BYEs will visit our server too if (method=="INVITE") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; # labeled all transaction for accounting setflag(1); if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); 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==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("xxx.org", "subscriber")) { www_challenge("xxx.org", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # call invitations to off-line users are reported using the # acc_request action; to avoid duplicate reports on request # retransmissions, request is processed statefuly (t_newtran, # t_reply) if ((method=="INVITE" || method=="ACK") && t_newtran() ) { t_reply("404", "Not Found"); #acc_request("404 Not Found"); #produce error (not found) break; }; # all other requests to off-line users are simply replied # statelessly and no reports are issued sl_send_reply("404", "Not Found"); break; } else { # user on-line; report on failed transactions; mark the # transaction for reporting using the same number as # configured above; if the call is really missed, a report # will be issued setflag(3); # forward to user's current destination t_relay(); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } #............................end....................

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