search for: sipuser

...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 [36225 at default:1] Set("SIP/sipuser-00000001", "ABA=999999999") in new stack -- Executing [36225 at default:2] BackGround("SIP/sipuser-00000001", "telbank/999999999/en/thetimeis") in new stack -- <SIP/sipuser-00000001> Playing 'telbank/999999999/en/thetimeis.slin44' (language...

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,DBNAME...

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

...her 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 = $agi->get_variable("SIPUSER"); if ($ID['result'] == 0) { $agi->verbose("SIPUSER not set -- nothing to do"); exit(1); } $agi->set_variable("MSN", exec("/var/lib/asterisk/agi-bin/msn4sip 111 222 333 " .$ID['data'])); ?> Output from asterisk: -- E...

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. Ran...

----- 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 >&...

...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_devi...

.../IPorHost * and there is an optional [!dnid] argument you can append to alter the * To: header. (Note: I don't think I have ever seen that optional "!dnid" argument documented anywhere...?) So, the version with the username and password looks to me like what I want... Dial(SIP/${SIPuser}:${SIPpass}@${SIPhost}) or else Dial(SIP/${SIPuser}:${SIPpass}@${SIPhost}!${SIPdial}) would seem to be what I need (I need to authenticate to SIPhost with the credentials SIPuser and SIPpass and I want to dial on to SIPdial). However, doing this results in the NOTICE message: chan_sip.c:23862...

...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 aro...

...--------+------------------------+-------------------+-------+------------+--------+--------------+-------------+-----------+----------------+-------------------+-----------------+----------------+-------------+--------+----------------+-----------+-----------------+ 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,proxy0...

...: 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 upd...

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', ev...

...] argument you can append to alter the > * To: header. > > (Note: I don't think I have ever seen that optional "!dnid" argument > documented anywhere...?) > > So, the version with the username and password looks to me like what I > want... > > Dial(SIP/${SIPuser}:${SIPpass}@${SIPhost}) or else > Dial(SIP/${SIPuser}:${SIPpass}@${SIPhost}!${SIPdial}) > > would seem to be what I need (I need to authenticate to SIPhost with the > credentials SIPuser and SIPpass and I want to dial on to SIPdial). It turns out that the username needs to be include...

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/${EXTEN}@sipuser|30|) Where the 'sipuser' is a context on sip.conf [sipuser] fromuser=sipuser What I would like to do is embed the username:password in the Dial string, something like this: exten => 1234,1,Dial(SIP/${EXTEN}@user:pas...

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 everythi...