similar to: Asterisk and SIPphone won't cooperate

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

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

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

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

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

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

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

Hello, I am new user of Asterisk. Yesterday I was trying to call from softphone to Asterisk, and that Asterisk routes this call to sipphone.com provider. I have found information on internet about how to register to sipphone and it seems that I have done. "sip show status" (or similar command) in CLI was showing me that I was registered. To call was not working, and on Asterisk's

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

Hello all, I'm having a problem with getting incoming callerid to a lan-connected phone. The Asterisk server is connected to the Internet, and a Grandstream BT101 phone on a lan interface: INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51) The phone registers with the Asterisk server as ext 20. I can initiate and receive calls from the Grandstream phone fine. The

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 =

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 =

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

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 = CN=Person,CN=Schema,CN=Configuration,DC=company,DC=local scope = subtree user_attrs = \ =home=/home/imapproxy/%u, \ =mail=maildir:/home/imapproxy/%u pass_attrs = uid=%u, userPassword=%w

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 =>

Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are

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",

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