similar to: SIP Calls Not Working

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

Hi guys, I realy need your help. It looks like SMBD doesn't want to talk with NMBD, but why? When I run the command: nmblookup torpc8 The result is: querying torpc8 on 10.5.255.255 Got a positive name query response from 10.5.2.27 ( 10.5.2.27 ) 10.5.2.27 torpc8<00> That is correct. When I run the command: smbclient -NL torpc8 The result is: resolve_wins: Attempting wins lookup for

Hi. I am trying to build a license free VPN for a product of mine. The VPN should have the following features: 1. VPN client should be able to pierce through corporate firewalls to replicate database enteries from client ( running windows xp ) to server (running enterprise linux) 2. VPN client if not able to connect directly to server due to network congestion should be able to route traffic

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

Hi, I am trying to write a dialplan that will use Dial() to call two local extensions. One extension will run an AGI script (a continuous background process, running until hangup), the other will connect the active channel to Jack() (also running as continuous process, until hangup). This is my current dialplan attempt: --------------------- [from-internal] exten = 514316XXXX,1,Answer() same

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Hi, Why are we getting message in the asterisk [Sep 10 12:55:23] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: Failed to authenticate device 601<sip:601 at 111.118.185.107>; tag=2f498fbd [Sep 10 12:55:24] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: Failed to authenticate device 601<sip:601 at 111.118.185.107>;tag=209a8aa9 Regards Deepak Bhatia --------------

I have just upgraded from courier to dovecot and things seem to be working okay (after a few problems getting shared folders working correctly). I notice the following error message in my logs:- Jun 23 11:17:32 calzone dovecot: child 1258 (auth) killed with signal 11 I am using the beta9 ebuild that is available from gentoo portage. I have followed the steps to try to get a core dump but it

I've found some manuals and tried this to do : Sip.conf [test] type=friend username=test1 secret=test1 host=192.168.1.238 context=tutorial fromuser=SIP Phone callerid=101 nat=no canreinvite=yes dtfmode=info disallow=all allow=ulaw [test] type=friend username=test secret=test host=192.168.1.240 context=tutorial callerid=100 nat=no canreinvite=yes dtfmode=info

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the

Hiya, here is a patch to fix that: [root@mailmx2 channels]# cvs diff chan_vpb.c Index: chan_vpb.c =================================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_vpb.c,v retrieving revision 1.9 diff -r1.9 chan_vpb.c 100,102c100,102 < static VPB_TONE Dialtone = {440, 440, 440, 0, 0, 0, 5000, 0 }; < static VPB_TONE Busytone = {440,

I'm stuck on queues! The way I read what documentation I have found, if I set up a queue like this: [general] persistentmembers = yes [testq] musiconhold=default strategy = ringall timeout = 10 retry = 5 context = testing member => SIP/100 and then add into extensions something like this: [incomingiax] exten => 1234,1,Dial(SIP/100,10) exten => 1234,2,Queue(testq|tTH|||300)

Hi all, With SELinux in permissive mode and iptables running, I'm unable to retrieve directory listings with ftp. stop iptables, and all appears again. This seems to be unrelated to passive/port modes for ftp client. If this is off topic, please let me know offlist and I'll take my question elsewhere. Otherwise I'll repost with output of # iptables status TIA, ~Ray

I searched the 22490 messages I have in my own personal asterisk-users archive and have not found the answer, and it also does not appear on the wiki. I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions are 100 and 101, respectively. On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do

hi guy, i have a problem, i have an sellvoip account and i want configure asterisk for outbound calls. this is my sip.conf register => XXXXX0000000000:PassWord@70.42.34.200 ; this is one of the sellvoip server [sellvoip_out] type=friend secret=PassWord username=XXXXXX0000000000 host=70.42.34.200 dtmfmode=rfc2833 context=testing disallow=all allow=ulaw extensions.conf this is a semplified

Hello all, I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to send the voicemails as attachment to e-mails and delete the voicemails from my PBX once it has been sent. But, I don't have a running MTA here even on the PBX itself. I just want to send the e-mails to my GMail account from my PBX. Can I just use the mail or mailx command to send the e-mail and attach the

Good day all I'm getting this error while trying to dial out on my asterisk server using a openline4 card "exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872" Please Help me

Asterisk will try calling both at once. As soon as one is answered it cancels the call to the other. What you can do is for extension 101 to put it in it's own context and then call the agi from the h extension. So something like this: [from-internal] exten = 514316XXXX,1,Answer() same => n,Playback(hello) same => n,Dial(LOCAL/100 at extensions&LOCAL/101 at extensions) [extensions]

Hi I'm new with asterisk, and I?m trying to install it with a X100P card. I stuck on some problems/issues. Could someone clarify me some points ? 1. I did a cvs checkout, did a 'make' and 'make install' . From the make output seems that some headers (.h) files for some codecs are missing. Is that ok? 2. Zaptel and Libpri did make and make install without noticeable