search for: proxy01

.....user0. # T 192.168.168.23:389 -> 10.0.46.4:49204 [AP] 0........e................000004DC: LdapErr: DSID-0C0906E8, comment: In order to perform this operation a successful bind must be completed on the connection., data 0, v1db1. This is what I see in Dovecot's log: Jun 23 11:26:43 proxy01 dovecot: auth: Debug: client in: AUTH#0111#011PLAIN#011service=imap#011secured#011session=7z8T/SsZ2AAKADPG#011lip=10.0.46.4#011rip=10.0.51.198#011lport=143#011rport=38104#011resp=<hidden> Jun 23 11:26:43 proxy01 dovecot: auth: Debug: ldap(bertoncello,10.0.51.198,<7z8T/SsZ2AAKADPG>):...

.../%u, \ =mail=maildir:/home/imapproxy/%u pass_attrs = uid=%u, userPassword=%w pass_filter = (&(objectClass=user)(sAMAccountName=%u)) auth_bind_userdn = company\%u If I try to login on the LDAP-Server using ldapsearch it works, but with Dovecot not... I see this in my log: Jun 22 16:14:08 proxy01 dovecot: auth: Debug: client in: AUTH#0111#011PLAIN#011service=imap#011secured#011session=+agW4xsZ4gAKADPG#011lip=10.0.46.4#011rip=10.0.51.198#011lport=143#011rport=34018#011resp=<hidden> Jun 22 16:14:08 proxy01 dovecot: auth: Debug: client passdb out: OK#0111#011user=bertoncello Jun 22 1...

...'s Asterisk CVS HEAD that's possibly the problem... Finally, the machine that is connected to both FWD and SIPPHONE is on a public static IP address, so there are no NAT issues involved here, and no STUN services needed either. OK, here is the sip.conf entry: register=1747XXXXXXX:YYYYYYY@proxy01.sipphone.com/4321 [proxy01.sipphone.com] type=peer ;auth=md5 secret=YYYYYYY username=1747XXXXXXX fromuser=1747XXXXXXX fromdomain=proxy01.sipphone.com host=proxy01.sipphone.com nat=no qualify=no canreinvite=no disallow=all allow=ulaw ;context=default ;callerid="Hadar Pedhazur" <1747XXX...

...mbers, like 17474745000, I only get "488 Not Acceptable Here". It works fine when I configure the softphone (Xten X-Lite) to use sipphone's server directly. Am I missing something? Here's my relevant config sections: sip.conf: in [general]: register => 17472442457:mypassword@proxy01.sipphone.com [sipphone] type=friend host=proxy01.sipphone.com username=17472442457 secret=mypassword fromuser=17472442457 fromdomain=proxy01.sipphone.com ;also tried it with these ;disallow=all ;allow=ulaw ;allow=alaw ;allow=g729 extensions.conf: [default] exten => _1747NXXXXXX,1,SetCallerID(...

...--------+-----------+----------------+-------------------+-----------------+----------------+-------------+--------+----------------+-----------+-----------------+ 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,queue_member_table,1 extensions => odbc,proxy01,extensions,1 -------------- next part -------------...

With 1.6.1 svn: [2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout: -- Registration for '17470121145 at proxy01.sipphone.com' timed out, trying again (Attempt #30) [2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'proxy01.sipphone.com' [2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register: Probably a DNS error for registration to 1747yyyxxxx...

...; > pass_attrs = uid=%u, userPassword=%w > pass_filter = (&(objectClass=user)(sAMAccountName=%u)) > auth_bind_userdn = company\%u > > If I try to login on the LDAP-Server using ldapsearch it works, but > with Dovecot not... > I see this in my log: > > Jun 22 16:14:08 proxy01 dovecot: auth: Debug: client in: > AUTH#0111#011PLAIN#011service=imap#011secured#011session=+agW4xsZ4gAKADPG#011lip=10.0.46.4#011rip=10.0.51.198#011lport=143#011rport=34018#011resp=<hidden> > Jun 22 16:14:08 proxy01 dovecot: auth: Debug: client passdb out: > OK#0111#011user=bertonc...

...o map directly to the final address. I need a couple test sites so if anyone wants to test Skype calling on their Asterisk network please send me email and I'l enable longer calling. Also in the 2 hard coded examples below (563 and echo) i want to also reference gizmo5 and not repeatedly have proxy01.sipphone. Can someone tell me how to construct the tightest syntax for that? Thanks. -- MR ------------------------------------------------------------------------------------------------------------------------ [gizmo5] type=peer ;COPY THIS CONFIG hos...

for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^ exte...

...ing, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx username=1747xxxxxxx password=xxxxx context=fromsipphone dtmfmode=rfc2833 canreinvite=no Any ideas? Am I doing something wrong? Thanks! -JD-

...console message: Starting simple switch on 'Zap/2-1' -- Executing SetCallerID("Zap/2-1", "17473863282") in new stack -- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack -- Executing Dial("Zap/2-1", "SIP/18006526672@proxy01.sipphone.com") in new stack -- Called 18006526672@proxy01.sipphone.com -- SIP/proxy01.sipphone.com-8efc is making progress passing it to Zap/2-1 -- SIP/proxy01.sipphone.com-8efc answered Zap/2-1 Feb 3 22:15:57 NOTICE[1218901440]: channel.c:1481 ast_set_read_format: Unable to find...

...5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls externip=xxx.xxx.xxx.xxx localhost=10.0.0.15/255.255.255.0 #include "sip-phones.d/*.conf" register=17476003384:xxx@proxy01.sipphone.com/701 [proxy01.sipphone.com] type=friend secret=xxx username=17476003384 host=proxy01.sipphone.com dtmfmode=inband context=default nat=yes qualify=no reinvite=no canreinvite=no insecure=very Extensions.conf: include => sip-forced include => from-sipphone...

...seems weird is that is only happens when Gizmo originates the call. I can see the prompts and stuff playing on the CLI, but nothing gets sent to the other end. Also, if I answer a call, sound goes neither way. I've tried a bunch of things My SIP.conf has register => 1747xxxxxxx:password@proxy01.sipphone.com [gizmo-inbound] type=peer context=from-gizmo dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm nat=yes host=proxy01.sipphone.com insecure=very canreinvite=no externip=69.10.14.12 localnet=192.168.0.0/255.255.255.0 I have no idea what to check / try next... My...

...ird on port 143). Unfortunately this happens randomly, reproducing this issue is difficult. We could not find any pattern, it happens rarely. We are running dovecot version 2.2.24 on Debian Jessie (backports repository). /root at dovecot01:~# dovecot --version// //2.2.24 (a82c823)/ Our Setup: /ha-proxy01 -> dovecot-proxy01 (proxy=yes) -> dovecot01 (proxy=no) * IP is used for multiple services (ports). We use the haproxy to send traffic on different ports to different servers. / Below the configuration (IMAPC is active for migration purposes). Do you have any recommendations where we could fo...

I have a sipphone.com account, with asterisk set to answer incoming calls, using the following settings (phone number and password omitted) in the Peer Details for the SIP Trunk: allow=ulaw context=from-pstn dtmfmode=rfc2833 fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx host=proxy01.sipphone.com insecure=very secret=xxxxx type=peer username=1747xxxxxxx The Asterisk machine is behind a Linksys router (full cone NAT). About 25% of the time, when I call that number (from another sipphone account), asterisk answers the line,...

...xxxxxxx") in new stack -- Executing SetCallerID("SIP/eric-8e80", "1747xxxxxxx") in new stack -- Executing SetCIDName("SIP/eric-8e80", ""Eric Windisch"") in new stack -- Executing Dial("SIP/eric-8e80", "SIP/1800xxxxxxx@proxy01.sipphone.com||r") in new stack -- Called 18002877863@proxy01.sipphone.com -- SIP/proxy01.sipphone.com-8361 is making progress passing it to SIP/eric-8e80 == Spawn extension (macro-sipphone, s, 3) exited non-zero on 'SIP/eric-8e80' in macro 'sipphone' == Spawn extens...

...2.0/eXosip" for peer 200 -- Got SIP response 481 "Subcription Does Not Exist" back from 192.168.1.121 -- Executing SetCallerID("SIP/200-0e5a", ""Name" 17476304480") in new stack -- Executing Dial("SIP/200-0e5a", "SIP/1747blabla@proxy01.sipphone.com|20|r") in new stack -- Called 1747blabla@proxy01.sipphone.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)" back from 198.65.166.131 -- SIP/proxy01.sipphone.com-8a47 is circuit-busy == Everyone is busy/congested at this time...

Hello, I'm using a proxy squid with authentification NT (Challenge/response) but i have problem with domains trusted. I have 2 domains DOMAIN1 and DOMAIN2. When i use wbinfo i have these results : wbinfo -t secret is good wbinfo -m DOMAIN2 ./wbinfo -a DOMAIN2\\proxy%proxy01 plaintext password authentication succeeded challenge/response password authentication succeeded It's very good but when i tried to get the groups of my users in my domain trusted DOMAIN2 the result is : ./wbinfo -r DOMAIN2\\proxy%proxy01 Could not get groups for user DOMAIN2\proxy%proxy01 P...

...To get FWD, IPTel, or SIPPhone to work, I've deduced I need to do something like the following after reading various online email archives (please correct me if I'm wrong): sip.conf: [general] register => XXXXX:password@fwd.pulver.com/1000 register => 1747XXXXXXX:password@proxy01.sipphone.com/1000 register => username:password@iptel.org/1000 extensions.conf: [default] include=sip [sip] include=sip [fwd] exten => _91339.,1,SetCallerID(XXXXX) exten => _91339.,2,Dial,SIP/${EXTEN:1}@fwd.pulver.com,tr exten => _91747.,1,SetCallerID(1747XXXXXXX)...

Hey guys + gals, Would you know where to point me to for an open source or commercial implementation of Theora & Vorbis for your mobile phone? Maybe as a Symbian app? Maybe a Java midlet? I'd like to be able to download & play video files to the mobile as well as maybe stream them too? Am I asking too much? :-) Thanks in advance for your help! Callum.