similar to: can't dial some particular numbers (providers ?) with asterisk sip / zap channels

I have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really

Hey, Im running Asterisk 1.2.2 and im having problems with the audio when bridging calls between the zap interfaces and sip. zap to zap work fine, as do sip to sip (but asterisk isnt in the media stream, as it doesnt need to be) and terminating the call and playing a test message via either sip or zap work fine. Basically, the only time I see this problem is trying to bridge between sip and

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip-> asterisk server -sip-> quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip->

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

Hi, on a configuration with one external ISDN S bus (to telco) and one internal S bus (to ISDN telephone), where Asterisk is in the middle (using HFC hardware), I noted the following: - when a GSM phone or ISDN phone calls in, the Transfer capability is Requested transfer capability: 0x00 - SPEECH - when an analog phone calls in (either from an analog line or an analog ISDN

Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as "switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile

let's suppose I have this dialplan : exten => _X.,1,Playtones(ring) exten => _X.,2,Dial(CAPI/contr1/${EXTEN},,g) exten => _X.,3,AGI(update) where "update" updates some db tables we have based on the type of extension Now, from the wiki : If the /g/ option is specified, and the called party hangs up before the calling party, then Dial exits with a return code of 0 to

In case this matters... RH AS 3 / samba 3.0.0-14-3E Just switched dial in from Windows RAS server to Linux PPP server. I can't seem to figure out a way to allow Windows users to log in with Domain name i.e. /etc/ppp/pap.secrets client server password ip address DOMAIN\user * my_pass 111.222.333.444 "DOMAIN\user" * my_pass 111.222.333.444

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

I just need a "temporary" pppoe server, that only uses PAP ___Can someone post a howto, just in a few lines, what to do?___ If I just do "pppd require-pap" then it gives this error: pppd: The remote system is required to authenticate itself pppd: but I couldn't fin any suitable secret (password) for it to use to do so. pppd is by default installed. and there are just a

I'm trying to set up a Linux-based dialin server on our company network. I'd like to have PPP authenticate using winbindd, if possible. I feel like I've almost gotten it to work, but I can't quite get there. Files: /etc/pam.d/ppp: #%PAM-1.0 auth required pam_nologin.so auth sufficient /lib/security/pam_winbind.so account required

Hi folks, Don't know if it could be interesting or not, even useful, but past days i was spending my time trying to use an old gsm motorola v150 mobile phone to get access to my host from my palm device with pssh (http://www.sealiesoftware.com/pssh/), these are the steps i did to accomplish it, feel free to suggest or improve it, anyway i found it usefull. First, this motorolla has an usb

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Hello, I am trying to walk through the following document: http://homepages.lu/charlesschwartz/radius/freeRadius_AD_tutorial.pdf in order to authenticate Cisco router and switch logins against FreeRadius/Active Directory. Using the HowTo, I have successfully joined a FC2 box to our Windows 2003 AD for testing purposes. I have also successfully used the manual ntlm_auth command to authenticate

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

Hi, I have a freeradius server that is working well in university. We use EAP-TTLS and PAP protocols. Users from Windows can use Securew2. Users from Linux and Mac OS X luckily have native support for EAP-TTLS and PAP. (if you think is Off Topic, keep reading on). On Ubuntu I can use the nm-applet for setting the connection up. But I'd want to find a way to automatize it, that it finds the

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

When I try to run wine /mnt/cdrom/setup.exe on the Monopoly CD-ROM it returns "Terminated" on the console. This has happened since version 0.9.20. It worked almost perfect in wine version 0.9.19. Monopoly is a Windows 16 bit game. Any idea why this is happening now? Thanks in advance, Charlie

Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: <Proxy_1_> _sip._udp.example.com </Proxy_1_> However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine.

My goal is to make VPN access to our Samba PDC (FreeBSD 7.0) so that users can access there home shares from Windows clients. I have read the instructions at http://samba.org/ftp/unpacked/lorikeet/pppd/final-report.pdf, but I can't make it work. Don't know if is due to my lack of skills or has something to do with the Popop functionality in FreeBSD. Following the instructions, i made a