similar to: LDAP authentication

If you allow anonymous search on AD maybe you can try to set auth_bind = no . a. On 22/06/15 17:19, Luca Bertoncello wrote: > Hi again > > I'm trying to authenticate a user against an LDAP Server (well, our > AD, but it can LDAP). > > This is my configuration: > > hosts = my.server.local > auth_bind = yes > ldap_version = 3 > base =

Hi list! I'm always trying to configure Dovecot to ask our LDAP-Server (AD) in order to authenticate the users. I really don'know what can I do wrong... I configured my Dovecot so: hosts = chimaera.company.local dn = CN=mailproxy,CN=Users,DC=company,DC=local dnpass = SECRET sasl_bind = no tls = no debug_level = -1 auth_bind = yes ldap_version = 3 base = dc=company,dc=local deref =

I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing. At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls

I have configured my mysql database by following this link http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue The only difference is that I am using ODBC instead of MySQL with Realtime. Within extensions.conf I have the following for my queue exten => 9**2**1611,1,Answer exten => 9**2**1611,2,Queue(irock.com,tT,,,300) exten => *50,1,Answer exten =>

Hi list! I'm always trying to configure Dovecot to ask our LDAP-Server (AD) in order to authenticate the users. I really don'know what can I do wrong... I configured my Dovecot so: hosts = chimaera.company.local dn = CN=mailproxy,CN=Users,DC=company,DC=local dnpass = SECRET sasl_bind = no tls = no debug_level = -1 auth_bind = yes ldap_version = 3 base = dc=company,dc=local deref =

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

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, 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

I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware

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}@??????) ^^^^^^

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

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

Hi list! Finally I got the LDAP-Authentication work (it was a problem of the OU-Path... :( ). Now I can authenticate the user against the AD and forwarding the IMAP-Connection to the Exchange Server. Wow! My next problem: we have TWO ADs and TWO Exchange-Servers. The first AD has the users for the first Exchange, and the second AD for the second Exchange. I defined two files so:

I?m trying to use Playtones to have a tone played periodically throughout phone calls. Unfortunately, I can?t seem to get PlayTones to work. I never hear the audio tones. Here is the output on the Asterisk console. -- Executing [19525553312 at proxy-dial:2] PlayTones("SIP/testphone-00000032", "1400/500,2000/5000") in new stack [2014-10-30 14:28:31] WARNING[23154]:

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

When attempting to call one of the example numbers, 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

I have written this configuration script which uses OpenSky to make Skype calls directly from Asterisk devices using my companies SIP to Skype gateway. Users can dial skype_anyskypeusername or manually add names or extensions which can get mapped to the correct dialing sequence. The right sequence is username at opensky.gizmo5.com but that gets mapped to sipphone address so I set that up to map

Hello when i dial a toll free no using sipphone i get this error message. How do i solve this? Any help will be appreciated. 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

Hello All. I started setting up my Asterisk system yesterday and everything was going well, i have registered with sipphone.com and set-up my Asterisk system to register with sipphone per the sip.conf file below. It was registered perfectly but I could not receive calls so I added in the line "insecure-very" and I then used the Washington DC access number to test and the phone

Hi all We experience some unexpected behavior with dovecot. It happens that emails marked as read get marked as unread (MUA is Thunderbird 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//

I am getting the following on dial-out via Sipphone to a 1-800 number (numbers obscured): ------------------------------------------------- == Spawn extension (macro-sipphone, s, 3) exited non-zero on 'SIP/eric-9546' in macro 'sipphone' == Spawn extension (default, 1747xxxxxxx, 1) exited non-zero on 'SIP/eric-9546' -- Executing Macro("SIP/eric-8e80",