similar to: SIP to H323 with no joy

Hi Folks, I am newbie to asterisk. Recentely I have installed asterisk on Linux Fedora 2 box. After reading some document, I tried to configure the server. When I connect to our server, SIP user-agent shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below. *CLI> sip show peers Name/username Host

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the

I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could

In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID "1000" and line 2 as SIP ID "2000". Basically I have this set up so that 1000 and 2000 are "lines in hunting" on incoming extension "555". I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here

I'm finding I can't run two festival commands in the same connection. Given the following: exten => 555,1,Answer exten => 555,2,Wait(1) exten => 555,3,Festival(mary had a little lamb) exten => 555,4,Wait(1) exten => 555,5,Festival(she also had a duck) exten => 555,6,Hangup Calling 555 gets the first line, then I get the error: May 20 17:59:16 WARNING[1301883824]:

i have problem with dial-plan in php. I have 3 extention in dial plan, 555,551 and 551. the problem is that READ PIN||3 works in 555 and hangs in other extentions. (after timeout asterisk writes thar "user entered nothing"). i can't get what's wrong... here is my dial-plan [incoming] exten => 555,1,Answer(); exten => 555,2,DigitTimeout(2); exten => 555,3,Wait(1) exten

If I try to execute this dialplan, and nobody picks up at any of the three extensions (7780 7781 and 7782), it's supposed to go to voice mail; instead, it hangs up and gives me a busy signal: exten => 2001,1,Dial(sip/7780,20) exten => 2001,2,Goto(2001,102) exten => 2001,102,Dial(sip/7781,20) exten => 2001,103,Goto(2001,203) exten => 2001,203,Dial(sip/7782,20)

Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) I end up getting this when I call from 2000 to 2001. 2000, 2002, and 2001 all exist in sip.conf and I connect using them. I have all three setup to use the from-sip context. Any suggestions on what is happening? [from-sip] exten =>

Hi All We are unable to hear any voice where as in tcpdum it shows that RTP is flowing both ways ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and

Hello all! I googled lists.digium.com and ser mailing list, but did not find any working configuration of asterisk used as voicemail for SER. This is my config if (uri==myself) { if (method=="REGISTER") { save("location"); log (1, "Registered\n"); break; };

Hrm, All of a sudden for some reason Wait() and Playback() are returning non-zero and its causing calls on my inbound SIP leg not to complete. I'm not sure why -- Executing Answer("SIP/2181-4518", "") in new stack -- Executing Playback("SIP/2181-4518", "silence/1") in new stack -- Playing 'silence/1' (language 'en') == Spawn

Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, Dave sip.conf -------- [general] port =

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't seem to get it to work correct. Could someone point me to what I need to do? I have attached what I believe are the relevant parts. [globals] ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 ; uncomment this line if you are using Ogg Vorbis

I cannot get Festival to work with asterisk. I have the following: exten => 555,1,Answer exten => 555,2,Festival(mary has a little lamb) exten => 555,3,Hangup I get the following from asterisk: "Festival returned ER" and the festival logs shows the following: client(1) Fri Jul 16 15:35:54 2004 : disconnected client(2) Fri Jul 16 15:40:26 2004 : accepted from localhost

hello, I got those extensions: exten => 555,1,MeetMeCount(500|count) exten => 555,2,Gotoif,$[${count} = 1]?6 exten => 555,3,Meetme,500|pMs|1234 exten => 555,4,Playback,goodbye exten => 555,5,Hangup exten => 555,6,Goto(from-internal-custom,556,1) exten => 555,7,hangup exten => 556,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/) exten =>

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

Asterisk cvs-head compiled 2005-10-07 11: Oct 12 18:35:12 NOTICE[21740]: chan_sip.c:10685 handle_request_register: Registr ation from 'sip:301495906@204.212.194.101' failed for '208.5.218.28' - Not a lo cal SIP domain The sip phone is a Cisco 7960 with one line defined, and registration with * is occuring just fine. Calls to/from the phone are fine. The phone is on a distant