cc_sip_buddies not working (sort of)

zabata200 · **Joined:** Wed Dec 30, 2009 11:49 am **Posts:** 4

Hello

cc_sip_buddies table gets filled from a2billing admin as usual.
But when i try to register softphone based on settings in cc_sip_buddies it doesn't register and gives error that peer not found.
So, when i put the same values in sip.conf file, my softphone registers fine.

This is a weird problem because when i connect to asterisk using asterisk -vvvvvvvvvr command it tells me that:-
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
== Binding sipusers to mysql/a2billing/cc_sip_buddies
== Binding sippeers to mysql/a2billing/cc_sip_buddies

Clearly, it is fetching sipusers from the database rather than the sip.conf file. Then why does not my softphone register?

Kindly help me. I am a beginner in a2billing.

Regards

jroper · **Joined:** Fri Jun 23, 2006 3:56 pm **Posts:** 4065

Recheck your Asterisk realtime configuration.

Joe

zabata200 · **Joined:** Wed Dec 30, 2009 11:49 am **Posts:** 4

Yeah, I have double checked realtime configuration. It points to a2billing database. When i change database to asterisk's sip_buddies table it agains work fine.

So whats the problem ?

ferdorna · **Joined:** Wed Mar 10, 2010 5:42 pm **Posts:** 47

I am having the same problem.

I am using Ask 1.6.2.6 and A2B v1.6.0 stable.

Any help please?

Thanks!
Fernando

jroper · **Joined:** Fri Jun 23, 2006 3:56 pm **Posts:** 4065

Does this post help

viewtopic.php?f=13&t=7476

Joe

ferdorna · **Joined:** Wed Mar 10, 2010 5:42 pm **Posts:** 47

Thanks jroper,
I checked everything. I paste the results:

Code:

VGW*CLI> core  show config mappings
Config Engine: odbc
Config Engine: mysql
===> extensions (db=general, table=cc_sip_buddies)
===> iaxpeers (db=general, table=cc_iax_buddies)
===> iaxusers (db=general, table=cc_iax_buddies)
===> sippeers (db=general, table=cc_sip_buddies)
===> sipusers (db=general, table=cc_sip_buddies)
Config Engine: curl[/size]

[size=85]VGW*CLI> realtime load sipusers name 9781115678
                   Column Name  Column Value                  
          --------------------  --------------------          
                            id  10                            
                    id_cc_card  6                             
                          name  9781115678                    
                   accountcode  9781115678                    
                      regexten  9781115678                    
                      amaflags  billing                       
                     callgroup                                
                      callerid  9781115678                    
                   canreinvite                                
                       context  ipphones                      
                     DEFAULTip                                
                      dtmfmode  RFC2833                       
                      fromuser                                
                    fromdomain                                
                          host  dynamic                       
                      insecure                                
                      language                                
                       mailbox                                
                     md5secret                                
                           nat  yes                           
                          deny                                
                        permit                                
                          mask                                
                   pickupgroup                                
                          port                                
                       qualify  yes                           
                   restrictcid                                
                    rtptimeout                                
                rtpholdtimeout                                
                        secret  123456                        
                          type  friend                        
                      username  9781115678                    
                      disallow                                
                         allow  ulaw,alaw,gsm,g729            
                   musiconhold                                
                    regseconds  0                             
                        ipaddr                                
                cancallforward  yes                           
                   fullcontact                                
                        setvar                                
                     regserver                                
                        lastms                                
                   defaultuser                                
                          auth                                
                  subscribemwi                                
                       vmexten                                
                    cid_number                                
                   callingpres                                
                   usereqphone                                
                 incominglimit                                
              subscribecontext                                
                    musicclass                                
                    mohsuggest                                
                 allowtransfer                                
                   autoframing                                
                maxcallbitrate                                
                 outboundproxy                                
                  rtpkeepalive        

But when I try to register my x-ten I've got the following:

Code:

<--- SIP read from UDP:200.49.150.65:5060 --->
REGISTER sip:200.49.150.10 SIP/2.0
Via: SIP/2.0/UDP 200.49.150.65:5060;rport;branch=z9hG4bK336960FE040B427CA8E4B42BC23E5115
From: 9781115678 <sip:[email protected]>
To: 9781115678 <sip:[email protected]>
Contact: "9781115678" <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 8149 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-PRO build 1101
Content-Length: 0

<------------->
-- (11 headers 0 lines) ---
Sending to 200.49.150.65 : 5060 (no NAT)
[Apr 30 14:36:13] WARNING[3345]: chan_sip.c:23361 handle_common_options: Unknown directmedia mode '' on line 0
[Apr 30 14:36:13] WARNING[3345]: acl.c:392 ast_get_ip_or_srv: Unable to lookup ''

<--- Transmitting (no NAT) to 200.49.150.65:5060 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 200.49.150.65:5060;branch=z9hG4bK336960FE040B427CA8E4B42BC23E5115;received=200.49.150.65;rport=5060
From: 9781115678 <sip:[email protected]>
To: 9781115678 <sip:[email protected]>;tag=as46fa3d65
Call-ID: [email protected]
CSeq: 8149 REGISTER
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
[Apr 30 14:36:13] NOTICE[3345]: chan_sip.c:21538 handle_request_register: Registration from '9781115678 <sip:[email protected]>' failed for '200.49.150.65' - No matching peer found
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)

I do not know what is going on. Seems like Asterisk does not find a peer.

Thank you.
Fernando

ferdorna · **Joined:** Wed Mar 10, 2010 5:42 pm **Posts:** 47

I mention something else...very odd:
If I create a sip config from the customer´s list and try to register it goes OK. Then, if I edit the sip config and change the password (secret) (and also on my x-ten softphone) I can not register anymore nor with the new or the old password. Again I have to delete de sip config and create a new one.
If I don't use realtime everything goes OK....te problem is that I want to use realtime

)

Thank you.
Fernando

jroper · **Joined:** Fri Jun 23, 2006 3:56 pm **Posts:** 4065

Hi

If you have the rtcachefriends directive set to yes, you may have to do an asterisk reload.

Joe

ferdorna · **Joined:** Wed Mar 10, 2010 5:42 pm **Posts:** 47

Thanks jroper,
doing a sip reload it works...but is it correct to force a reload when I use realtime? I thought that using realtime the modifications on sip peers and sipusers didn't need a reload.

Thank you!!
Fernando

jroper · **Joined:** Fri Jun 23, 2006 3:56 pm **Posts:** 4065

Hi

Only if you have rtcachefriends on, which is only required if you are delivering DID back to the account number, e.g. sip/a2b-acc-no. if you were delivering DID to fixed IP for instance, you would not need it.

Realtime only keeps registrations for as long as required, which means you can have many many thousands of sip endpoints without undue load on the platform, unless of course you have rtcachefriends on, which holds the endpoints details in memory.

Joe

ferdorna · **Joined:** Wed Mar 10, 2010 5:42 pm **Posts:** 47

OK, I understand....so, then is OK that when I change a SIP-friend's password it will continue using the old one till I force a sip reload....
the last question: Doing a reload could have negative effects on the other users already registered...I mean they lost the registration or things like that?

Thank you for your help!

Fernando

jroper · **Joined:** Fri Jun 23, 2006 3:56 pm **Posts:** 4065

Hi

No there should be no ill effects, all you are doing is effectively refreshing Asterisk's knowledge and config of each of it's peers.

Joe

ferdorna · **Joined:** Wed Mar 10, 2010 5:42 pm **Posts:** 47

Thank you very much Joe!

You clarified me many things.

Regards!
Fernando

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