Asteisk 1.6.1.0 :SIP reg :"No matching peer found"

Hi,

I am downloaded latest 1.6.1.0 files and compiled and installed. I am testing SIP registrations with Asterisk. I also downloaded latest SVN Asterisk-GUI and compiled. I am using Ubuntu9.04 Distro.

Details:
$: wget downloads.digium.com/pub/asteris … 1.0.tar.gz
$: wget downloads.asterisk.org/pub/telep … .10.tar.gz
$: wget downloads.asterisk.org/pub/telep … 2.1.tar.gz
$: sudo svn co svn.digium.com/svn/asterisk-gui/branches/2.0/ asterisk-gui

Through WEB GUI i added a DIALPLAN, and then added a USER 1000 and updated and APPLIED the setting from GUI. In the CLI if i check “sip show users” or “sip show peers”, i don’t get anything.

I use X-lite(sip phone) and try to register the Entry “1000” with asterisk. I get following error:

CLI> [Jun 4 10:18:24] NOTICE[12669]: chan_sip.c:19536 handle_request_register: Registration from ‘"1000"sip:1000@192.168.53.114’ failed for ‘192.168.53.156’ - No matching peer found

GUI is writing correctly to the conf files. These i checked by editing the user.conf files. Even files permissions are fine.

So basically the user entries are not being read by the asterisk, although i cannot see any Parsing errors when Asterisk comes up.

Frustrated with this, i downloaded 1.4.25 version and compiled and entered the same Entries in Dialplan,Users through WEB GUI. It Works like a charm…!!!

So is there any problem with 1.6.1.0 version. Do i need a file a bug?

Please advice…

Regards,
Umesh

umesh.asterisk,

I have also ran into this problem over the course of 3 months. I have compiled CentOS 5.3 with the latest and greatest of everything. Lastnight I compiled Asterisk version 1.6.0 and 1.6.1 and seem to have the same issue. While in debug mode at the CLI, i see the users being created in the web gui and being parsed to the users.conf file. While in the CLI type sip show users. This will show you if those users you just created are in fact stored for asterisk to use them. I also read more into this issue, and it seems as if the file that should be getting parsed after the creation of the user, should be sip.conf not users.conf. I believe the real reason why nothing shows for me when i launch the command “sip show users” is because when the user is created in the GUI, it is not getting saved into the right location (sip.conf instead of users.conf). I spent hours latnight trying to find the file that contained the location that tells the GUI to parse it to users.conf. I would be interested to see what would happen when i save it to sip.conf.

Today I am going to try and “make menuselect” and create my users in just asterisk, bypassing the gui, and watch what the CLI says when the user is created. I will try to keep you posted with the results. If you find anything out. Please do not hesitate to contact us at sales@frigidinc.com. I am excited to hear what you find out. Thanks and enjoy the rest of your week!

Best Regards,

Rick

As I understand it, users.conf is intended as an alternative to sip.conf, etc., primarily for GUI configuration management programs.

While reading the entire users.conf file, (or the begining none the less) it states that it is used as a template or not using it as all the other configuration files. Not quite written clear enough. I will by this afternoon, try adding a user via the menu from the command line not the gui. I will let you know. Thanks David.

Thanks Rinc and david55 for the replies…

I still wonder how it works fine in 1.4.25. The whole point of the GUI is that we don’t have to edit the *.conf files and make it a mess. Something has changed from 1.4 1.4 to 1.6 which is creating this… Maybe some more testing is required. But something as basic as this seems broken. Or is it a settings issue?

I suppose more testing is required…

Regards,
Umesh

There are ongoing changes, resulting in the structure that corresponded to type=user being removed. It’s conceivable that the changes weren’t made properly for users.conf. You need to look at issues.asterisk.org, to see if there are any known or fixed problems.

I suspect users.conf is considered a secondary function by the core developers.

I just reverted back to 1.4 yesterday after testing and it does work correctly. Maybe there will be a fix for this soon. The only problem that i’m running into now is that my redirect statement in my http.conf does not work properly as it does in 1.6. Not sure what i’m doing wrong. Thanks guys!

-Rick

Any update for this problem???i am also having this type of problem.Any solution??Right now…I installed another ubuntu on my virtual box to test asterisk v1.4 if it really works well…

You need to search issues.asterisk.org/. If it has already been reported, you can find the status, and if not fixed in the development version, can add a “me too”, which might influence the priority.

If it is not already known, you will need to raise it there. Please double check the upgrade document in 1.6.x, first.

Well for those of you who are interested, I have been battling with this issue for a quite some time.
it appears, that everything works fine up to version 1.6.0.6
going beyond that, brings up the exact error.
I have a few ATAs connected and I make use of * through dahdi and conect to * via my cell-phone (nokia E90),when I am abroad or in a hotel with WiFi, which works like a charm. the moment I go beyond version 1.6.0.6 no SIP clients connect anymore.

I would be very grateful to hear how this pans out.

Hi there,
just to let everyone know:
I have opened a bugreport on this issue, specially since it might indirectly have something to do with the Asterisk-GUI (writing to user.conf rather than sip.conf)
It is bug number: issues.asterisk.org/view.php?id=15353

ZimboKraut

Thanks ZimboKraut for issuing the Bug report…

Regards,
Umesh

No problem, I just wish, that one of the developers would at least have a look at it…
So far, there is not even a note attached :frowning:
I know they have a lot of work to go through, but I am just soooooooo impatient
Well, I will keep on looking and trying.

ZimboKraut