search for: sipusers

Hi Gang, Hopefully somebody out there has a "doh" for this one. My dialplan announces the date and time using SayUnixTime. When I run this: exten => 36225,1,Set(ABA=999999999) exten => 36225,n,Background(telbank/${ABA}/${CHANNEL(language)}/thetimeis) exten => 36225,n,sayunixtime(,,Abe 'digits/at' IMP) I get this CLI output -- Executing

Hi, I have 2 databases DBNAME1 and DBNAME2. In each database I have a table called sipusers (so DBNAME1.sipusers and DBNAME2.sipusers). Can I use both sipusers tables in Asterisk RealTime? Something like this: /etc/asterisk/extconfig.conf: [settings] sipusers => odbc,DBNAME1,sipusers sippeers => odbc,DBNAME1,sipusers sipusers => odbc,DBNAME2,sipusers sippeers => odbc,DBNAME2...

I'm trying to get sipusers working with a realtime odbc database on Asterisk 1.6. We have sippeers working from the database, but need sipusers to be in a separate table for other implementation reasons. sip show user test load returns results from the database. CLI> sip show user test load * Name : test...

Still a couple of questions...... I did configure extconfig.conf ... ;iaxusers => odbc,asterisk ;iaxpeers => odbc,asterisk ;sipusers => odbc,asterisk sipusers => mysql,asterisk,sip_devices sippeers => mysql,asterisk,sip_devices ;sippeers => odbc,asterisk ;sipregs => odbc,asterisk ;voicemail => odbc,asterisk ;extensions => odbc,asterisk ;meetme => mysql,general ;queues => odbc,asterisk ;queue_members =&...

Hi, I would have expected that peers of type friend ( for example an SIP-phone) registring at Asterisk will be searched in sipusers. But the peers will be searched in sippeers. May be sombody can explain the difference? Asterisk 1.4 thanks Thomas

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username

Hello all, I try to use a agi script to get a variable from * und put them into a script which gives me another variablke and put this in *. My problem is now it seems the var ID is empty coz i always jump into the result 0 loop. The $MSN should be in the SetCIDNum. #!/usr/bin/php -q <?php include("/var/lib/asterisk/agi-bin/phpagi.php"); $agi = new AGI(); $ID =

hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old password. Rand...

----- Original Message ----- From: "kjcsb" <kjcsb@orcon.net.nz> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, January 24, 2007 8:24 AM Subject: Re: [asterisk-users] realtime sipusers and rtcachefriends... bigheadache!! > >> hi folks, >> >> I am using asterisk 1.2.13 (debian etch). >> >> My customer's sip accounts are stored in realtime sipusers. >> >> I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes >&g...

...valid realtime ; engine. There is: > ; odbc ... res_config_odbc > ; sqlite ... res_config_sqlite > ; pgsql ... res_config_pgsql > ; curl ... res_config_curl > ; ldap ... res_config_ldap > ; > ;iaxusers => odbc,asterisk > ;iaxpeers => odbc,asterisk > ;sipusers => odbc,asterisk > ;sipusers => mysql,asterisk,sip_devices > ;sippeers => mysql,asterisk,sip_devices > ;sipusers => mysql,general,sip_devices > ;sippeers => mysql,general,sip_devices > sipusers => mysql,default,sip_devices > sippeers => mysql,default,sip_devic...

Hi. I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and for some reason it's simply not doing it. I've even resorted to reading the source code to try and work out what I'm doing wrong... In channels/chan_sip.c I find: * SIP Dial string syntax: * SIP/devicename * or SIP/username at domain (SIP uri) * or

...t=UDP>' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/AsteriskHosted/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/AsteriskHosted/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/AsteriskHosted/sip_buddies I have the following in extconfig.conf : sipusers => mysql,Asterisk,sip_buddies sippeers => mysql,Asterisk,sip_buddies I have the following in res_mysql.conf : [general] dbhost...

Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already arou...

...--------+------------------------+-------------------+-------+------------+--------+--------------+-------------+-----------+----------------+-------------------+-----------------+----------------+-------------+--------+----------------+-----------+-----------------+ My other realtime stuff like sipusers and vmusers works just fine. extconfig.conf ; Primary Database Connection sipusers => odbc,proxy01,sipusers,1 sippeers => odbc,proxy01,sipusers,1 voicemail => odbc,proxy01,vmusers,1 meetme => odbc,proxy01,meetme,1 queues => odbc,proxy01,queue_table,1 queue_members => odbc,proxy01...

...: voipadmin] Other wild idea, is there a minimum number of fields that haved to be filled? And why is asterisk complaining about not being able to find the databse, when trying to fill it from the asterisk-CLI? My database _is_ named "asterisk"...... > kc3054*CLI> realtime update sipusers set SET port = 4343 WHERE name = > 00000277611 Failed to update. Check the debug log for possible SQL > related entries. > Command 'realtime update sipusers set SET port = 4343 WHERE name = > 00000277611' failed. > [May 18 18:47:16] WARNING[16718]: res_config_mysql.c:559 upda...

I've been going in circles for a few weeks now with Realtime SIP. My extconfig.conf has: sipusers => mysql,dbname,ast_sip_users sippeers => mysql,dbname,ast_sip_users When I do a 'sip show peers' I see all my phones. When I do a 'sip show users' I only see a few of them. I can't work out why this is the case. They are also coming up with NAT as 'RFC3581', eve...

On Sunday 25 October 2020 at 16:27:00, Antony Stone wrote: > Hi. > > I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and > for some reason it's simply not doing it. I've made a bit of progress - I can now get it to authenticate, although it's still not dialling on to the correct number. > I've even resorted to reading the source code

Hi All, I'm trying to get the user:pass embedded in a SIP Dial string instead of calling a SIPuser in sip.conf: Regular way, exten => 1234,1,Dial(SIP/${EXTEN}@sipuser|30|) Where the 'sipuser' is a context on sip.conf [sipuser] fromuser=sipuser What I would like to do is embed the username:password in the Dial string, something like this: exten =>

Hello, I have something like this: SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN After calling from SIP to PSTN (and from PSTN to SIP too) I can't hear anything only in my SIPUSER. At the PSTN side everything is OK. I have another network with another h323/sip (in the place of asterisk) and there everything is OK. In AUDIOCODES logs I see that everything goes