similar to: "dtmf transcoding" with asterisk

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm

Hello, ? We are planing to utilize Microsoft 2008 R2 Read Only Domain controller, and deploy RODC's in branches. ? If I would like to have SAMBA servers in those branches, will I be able to add them to the domain (using "net ads join") and work with them, when using the RODC's as domain controllers configured in my smb.conf & krb5.conf? ? I have looked around and did not

hello. i am running samba-2.2.7 on a RH7.2 Kernel 2.4.7-10. i am using the left side (the folders tree section) of the windows explorer on WinNT4 SP6 station, and I move folders arround. most of the times after i move a folder there is no change in the presentation, in order to see the new folders tree i have to refresh the screen (F5). i will be gratefull for any advice / replay. tx.

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Hello, I am working on RHEL4 update 5, using SAMBA 3.0.26a. I am trying to configure SAMBA as a domain member in a Win2K domain. I am using Kerberos authneticiaon (Security = ADS). I am getting an error when I am trying to join the domain (this is the output I get from running ?net ads join ?Uuser%password ?d10?: [2008/07/15 12:11:51.347873, 1, pid=5304]

I'm trying to make my computer work with a Fedora 32 live DVD. Video issues make it either crap out or only give me 640x480. Grrrr. Centos 7 is running just fine. I think I could make it go by booting to runlevel 3 and running startx manually. For that to work, I think I need to pass startx the right configuration file. To me, 'tain't obvious what to put in that file. Is there a way

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hello, I am working on RHEL 3 update 4. The Kerberos version that comes with the OS is 1.2.7. I have installed samba 3.0.14a and encountered multiple winbind crashes. I have done some debugging and found the cause, samba function "ads_cleanup_expired_creds" calls Kerberos function krb5_cc_remove_cred (if the ticket is expired), the Kerberos implementation holds a struct of function

We have multiple entries like the one below in our users.conf file... where the username. Contact and secret changes for different customers and we register on their behalf to the Service Provider. For the trunk below: when the call is placed out, Asterisk (1.4.18) sends the username of "abc.com" in the MD5 Auth .....which obviously does not match the trunk setup for this Customer with

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Hi * users, I am only seeing SendText in the available asterisk applications. But I have not seen Receive Text application. I tried on asterisk-1.0.7 and 1.0.9. Can anyone tell me how to use this receive text command. I want to use receivetext command and get text information from an softphone so that that can be routed to some other phone supporting text message. (my soft phones are SIP/IAX

Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0

I have tried all the suggestions up till now but the error message is still there. I have tried this configuaration for roundcube: $config['imap_conn_options'] = array( 'ssl' => array( 'peer_name' => '<FQDN_OF_DOVECOT_CERTIFICATE>', 'verify_peer' => true, 'verify_depth' => 3, // 'cafile' =>

Hello, I am using samba 3.14a on RedHat es3. About every day the winbind panics. this is the winbind log: ads_try_connect: trying ldap server '11.9.13.235' port 389 [2007/06/13 06:59:26.581501, 3, pid=18850] libads/ldap.c:ads_connect(285) Connected to LDAP server 11.9.13.235 [2007/06/13 06:59:26.582007, 3, pid=18850] libads/ldap.c:ads_server_info(2469) got ldap server name

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both set to register with the same account right now. I shut Asterisk down on

I am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone lines connected to the FXO ports one from telecom, another from vodaphone. He has set up the dialplan so that one of the trunks fails over to the other trunk. Everything seems to be working OK except for outgoing calls. He can call from

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

Im new to wine, I use linux as my desktop at work and my boss has demanded that we run liveperson/humanclick but it totally bombs out when I try to install it... Anyone have any luck getting this program to install/run. _________________________________________________________________ Get your FREE download of MSN Explorer at http://explorer.msn.com/intl.asp