Re: [Asterisk-Users] Asterisk as a gatekeeper
Asterisk can only be configured as a GW AFAIK, what ever flavor of H323 you use with Asterisk it will not work as a GK. Atif rommel malana wrote: Hello, Right now i'm trying to set-up a gatekeeper and i'm having a hardtime doing it, what i'm thinking is instead of having a gatekeeper i'll use the asterisk to be a gatekeeper. Can the asterisk be a gatekeeper? Thanks a lot, Rommel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with EWSD v16
Dear Gulzar, Thank you for your reply, I am using same configs. I have tried both 0 & 1 in timing but no luck. I will try again with 'timing' parameter = 1 in zapata.conf best Regards, -- Atif Rasheed Gulzar Hussain wrote: I am using EWSD's PRIs and I am not having this problem my configs are Zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us Zapata.conf [channels] language=en context=ext-acd switchtype=euroisdn signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes group=1 channel => 1-15 channel => 17-31 pridialplan=private prilocaldialplan=private overlapdial=yes usecallerid=yes hidecallerid=no immediate=no usecallingpres=no --- Atif Rasheed <[EMAIL PROTECTED]> wrote: if any EWSD guru out there..please help ??? Hello all, I am running Asterisk with Digium E1 card with zaptel, libpri, asterisk cvs v1-2. My server is interfaced with EWSD v16 using a PRI on E1. I am running into a problem that at my telco's end alot of trunks are getting BPRM (Block permanant) status. I am not sure why EWSD is blocking trunks. config at my end::: coding = hdb3 format = ccs,crc4 signalling = euroisdn, pri_cpe config at my telco's end coding = hdb3 format = crc4mf signalling = euroisdn, pri_net Is there any EWSD guru around who can explain why trunks are getting BPRM status in EWSD switch. I will really appriciate your help Thank you -- Atif Rasheed ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk with EWSD v16
if any EWSD guru out there..please help ??? Hello all, I am running Asterisk with Digium E1 card with zaptel, libpri, asterisk cvs v1-2. My server is interfaced with EWSD v16 using a PRI on E1. I am running into a problem that at my telco's end alot of trunks are getting BPRM (Block permanant) status. I am not sure why EWSD is blocking trunks. config at my end::: coding = hdb3 format = ccs,crc4 signalling = euroisdn, pri_cpe config at my telco's end coding = hdb3 format = crc4mf signalling = euroisdn, pri_net Is there any EWSD guru around who can explain why trunks are getting BPRM status in EWSD switch. I will really appriciate your help Thank you -- Atif Rasheed ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk with EWSD v16
Hello all, I am running Asterisk with Digium E1 card with zaptel, libpri, asterisk cvs v1-2. My server is interfaced with EWSD v16 using a PRI on E1. I am running into a problem that at my telco's end alot of trunks are getting BPRM (Block permanant) status. I am not sure why EWSD is blocking trunks. config at my end::: coding = hdb3 format = ccs,crc4 signalling = euroisdn, pri_cpe config at my telco's end coding = hdb3 format = crc4mf signalling = euroisdn, pri_net Is there any EWSD guru around who can explain why trunks are getting BPRM status in EWSD switch. I will really appriciate your help Thank you -- Atif Rasheed ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: binding asterisk-h323 on two interfaces
I have cvs-head of Aug-2. README has no information on how to bind asterisk-h323 on multiple interfaces. actually this was my question that can we bind asterisk-h323 on multiple interfaces ? as h323.conf says that "bindaddr" should contain a single valid IP. if we bind h323 to 0.0.0.0 as we do in SIP, it sends 0.0.0.0 as it is to the caller and callee. Use cvs -head code from the last day or two and read the README. Jeremy McNamara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can asterisk send Remote-Party-ID header ???
Hello all, Kevin P Fleming once said that a patch will be released very soon to send Remote-Party-ID header from Asterisk. and this was said probably in Feburary. is that patch released yet or not ? if some please comment, I will really appriciate Regards, -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 vs chan_oh323 & chan_ooh323
hello there, can somebody please comment which one of these channel drivers will give best output doing g729|g723 pass-thru. only pass-thru is needed no transcoding. please share your experience. if somebody has some figures (simultanous calls using a certain channel driver) it will be apericiated. I have installed chan_h323 (by McNamara) and its working fine with me. I just want to know if I run this driver on a Dual-Xeon machine. can it handle 500 or > 500 simultanous calls in pass-thru mode. Regards, -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] looking for some draft (sip - iax2 mapping)
hello all, is there any draft available for sip-iax2 mapping. I mean sip 4XX server failure messages, 5XX Server Failure messages. how these SIP messages are mapped to IAX messages thank you -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] live monitoring of SIP calls chan_spy
hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application thank you regards, -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not relaying back the SIP response messages
HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling Asterisk. Now the problem is that error responses from the Softswitch to the Relaying Asterisk are not relayed back to the Calling Asterisk. Instead a 403 forbidden error message is sent back to the Calling Asterisk whatever the error response (503, 484, etc). Is there a way to relay back error responses through configuration scripts or do I have to dig in the source code -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC dimensioning
hello there, any one who used ASTCC in a real enviroment, or has successfully handled above 1k simultanous calls. need some evalution of ASTCC. if any one has such an experience please share it with the rest thank you Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] "*" behaviour in agentcallbacklogin
when an agent logs in using AgentCallbackLogin(), during a call when agent presses * call is hanged up. how can I get rid of this behaviour. that nothing should happen by pressing *. thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cvs stable
on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: pattern matching problems
thank you people for your help, I have done it, and in a different way,
like
exten => _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten => _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten => _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
exten => _011X.,1,AGI(iax.agi)
exten => _011X.,2,Dial(${MAG}/${EXTEN:3},45,tT)
exten => _011X.,103,playback(no-service)
I made the _011. more precise, I should say
--
Atif
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[Asterisk-Users] pattern matching problems
this is from my extensions.conf, the first three patterns are for
toll-free numbers, and fourth pattern is for other numbers, where an AGI
is called for authentication.
now when I dial 011448000664327 if falls into the fourth pattern, where
as it should be matched by the first pattern. Any suggestions
1 - exten => _01144800XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
2 - exten => _01144808XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
3 - exten => _01144500XXX,1,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten => _011.,1,AGI(iax.agi)
4 - exten => _011.,2,Dial(${MAG}/${EXTEN:3},45,tT)
4 - exten => _011.,103,playback(no-service)
thank you
--
Atif
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[Asterisk-Users] Re: SIP Provider in India/Pakistan/Bengladesh
PTA(Pakistan Telecommunication Authority) has recently issued LDI licences to number of contenders and use VoIP. noone yet has announced but very soon someone from them will announce SIP termination in Pakistan. can't say anything about India/Bengladesh -- Atif ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip phone, receiving calls but not placing any call
Hello there, I am configuring a sip-phone, it is receiving calls but its not placing calls. sip debug shows that asterisk received digits from phone. but why its not placing calls please help I have dialed 13 from sip-phone, here is some sip-debug INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKfLZ1GRUt2 Max-Forwards: 70 From: chinee ;tag=82veOQ0zKConAx6y To: 13 Call-ID: y2gsu70CXGySlU0s CSeq: 1 INVITE Contact: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 191 thank you -- Atif ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meetme conf-background.agi
Hello there!
Somebody tried the meetme|b which initiates the conf-background
AGI…
Actually I want to originate another call from a conference…my
AGI originates the call and connects it to the conference, but the call is nowhere
My extension
exten => 21,1,meetme(21|pb)
and my AGI
#!/usr/bin/perl -w
$aginame="conf-background.agi";
use File::Copy cp;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
print STDERR "Dialing your number\n";
$srcfile="/tmp/mycall";
$dstfile="/var/spool/asterisk/outgoing/mycall";
open(MYCALL,">$srcfile") || die "Cant't
open file :$srcfile $!\n";
print MYCALL "Channel:Zap/1/13\n";
print MYCALL "MaxRetries:2\n";
print MYCALL "RetryTime:60\n";
print MYCALL "WaitTime:30\n";
print MYCALL "Context:default\n";
print MYCALL "Extension:22\n";
print MYCALL "Priority:1\n";
close MYCALL;
cp($srcfile,$dstfile);
#used to hold the AGI, otherwise it quits
$AGI->get_data('ccs-getnumber','1','2');
print STDERR "dialing complete...\n";
Some one can sort out, where things are going wrong
Thank you
Atif
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