sr-users January 2005

sr-users@lists.kamailio.org

199 participants
399 discussions

[Serusers] Looking for SER + Asterisk-as-voicemail HOWTO

by Ashling O'Driscoll

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.

19 years, 2 months

[Serusers] No response from client

by support

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

19 years, 2 months

[Serusers] SER and NAT and RTProxy

by 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.

19 years, 2 months

[Serusers] AddPac-no media

by Jain, Hitesh

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.

19 years, 2 months

[Serusers] SER Gateway Authentication

by Tom Gaudasinski

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.

19 years, 2 months

[Serusers] Compiling Source From 8.14

by Mr B

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

19 years, 2 months

[Serusers] Straight MySQL or text CDR?

by Doug Eubanks

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

19 years, 2 months

[Serusers] Recommended rtpproxy version

by Luca Corti

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

19 years, 3 months

[Serusers] Looking for SER + Asterisk-as-voicemail HOWTO

by Henning Verbeek

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

19 years, 3 months

[Serusers] help!!!!

by zhang_zhaoxin＠pact518.hit.edu.cn

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

19 years, 3 months

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sr-users January 2005