RE: [Asterisk-Users] Dial T option not obeyed with Grandstream BT101
The T option is for the # transfer which is handled by Asterisk, in your case the phone has a transfer button and is able to send SIP messages telling Asterisk that the call should be transferred. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Barton Hodges Sent: Sunday, November 30, 2003 10:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dial T option not obeyed with Grandstream BT101 In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register = number:[EMAIL PROTECTED] extensions.conf: [from-sip] exten = s,1,Dial(SIP/111SIP/117) exten = 111,1,Dial(SIP/111,20) exten = 117,1,Dial(SIP/117,20) 1. The calling user dials number, which drops them into [from-sip] 2. Extensions 111 and 117 are Dialed. 3. The called user picks up extension 111. 4. The calling user presses Transfer on the Grandstream phone, then dials 117 and presses Send. 5. The called user on extension 111 is then transferred to extension 117. I don't believe this is supposed to happen because I have not specified the T option to the Dial command. Even without any options specified at all, both the calling and called users are able to transfer the call. I'm using a CVS snapshot from Sun, Nov 30th 04:04:45, 2003. What am I missing here? Barton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LCR with ENUM and DDNS: half the story
Brian West wrote:
Also I must point out that your NAPTR record is a bit wrong:
wrong:(bind9)
!+(.*)!iax2:foofone/1!
Read again, Brian.
The text clearly states that the shell eats up one of the slashes, so we have to
double-quote.
The only part I would add is the advice to make sure that you have bind 9 on both
master and slave servers, otherwise you propably will file bug reports on the
wrong piece of software...
/Olle ;-)
On Sun, 30 Nov 2003, William Waites wrote:
Ok, so you've read the Wiki and gotten call routing using ENUM to work
(http://www.voip-info.org/tiki-index.php?page=Asterisk%20E164%20Call%20Routing)
with your own ENUM-alike domain, e164.example.com.
But how do you populate it with data? You can do it manually, but that gets
very tedious very quickly. Or you can use the nifty DDNS updating program
that comes with bind9.
The first thing is to set configure your e164.example.com to allow ddns updates.
A very good document describing how to do this (just ignore the DHCP stuff) is
http://ops.ietf.org/dns/dynupd/secure-ddns-howto.html
In a nutshell (I used TSIG keys for simplicity, the procedure is analogous with
SIG(0) asymettric keys) this is how you do it.
On the client computer that will be allowed to update the database do:
% dnssec-keygen -a HMAC-MD5 -b 512 -n HOST client.example.com
Kclient.example.com.+157+13404
This creates the shared key, which will live in a file called
Kclient.example.com.+157+13404.key and .private
% cat Kclient.example.com.+157+13404.private
Private-key-format: v1.2
Algorithm: 157 (HMAC_MD5)
Key:
I9FvX+F3fcSVLkzlPSVR9THww+oN6o0mj/JgKTu9auzMx0IM7lmBd9RIfk2cbHvoV9drGQVsk+svkrf+AeN0JQ==
Now on the server, let that key update e164.example.com. To do this, change named.conf
to have
key client.example.com. {
algorithm HMAC-MD5;
secret
I9FvX+F3fcSVLkzlPSVR9THww+oN6o0mj/JgKTu9auzMx0IM7lmBd9RIfk2cbHvoV9drGQVsk+svkrf+AeN0JQ==;
};
zone e164.example.com {
type master;
file dynamic/e164.example.com;
update-policy {
grant client.example.com. subdomain e164.example.com. ANY;
};
};
and restart the nameserver.
That's it for the configuration.
Now, say you have just found a very good IAX2 peer, FooFone that offers /wonderful/
rates
to the ficticious country code 666. You can use a script like this, to tell the
asterisk application EnumLookup (see the howto above) to use this peer for that
country:
#!/bin/sh
TTL=3600
SERVER=nameserver.example.com
SERVER=sparx
ZONE=e164.example.com
KEYFILE=Kclient.example.com.+157+13404.key
nsupdate -v -k ${KEYFILE} EOF
server ${SERVER}
zone ${ZONE}
update delete *.6.6.6.e164.example.com.
update add *.6.6.6.e164.example.com. ${TTL} NAPTR 100 100 u E2U+IAX2
!+(.*)!iax2:foofone/1! .
update add *.6.6.6.e164.example.com. ${TTL} TXT greate $0.00/minute rate from
FooFone!
show
send
EOF
the first update line deletes any existing records for +666, the second adds the NAPTR
record for ENUM call routing, and the third adds a nice informational message in the
DNS
which is useful if you want a quick way to find out how much a call will be billed at.
Note the escaped-escaped-escape characters. The first is because the shell will try to
interpret \, so what actually gets sent to nsupdate is \\ which is correct for what
BIND
wants.
And the second half of the puzzle? Figuring out how to know what to put in the DNS,
calculating the best rates...
Hope someone finds this useful,
-w
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Runbovägen 10, 192 48 Sollentuna, Sweden
Phone: +46 8 594 78 810, Fax: +46 8 594 78 820
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[Asterisk-Users] Another * crash
I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes. When I debug sip I get a noisy feedback from SER, and then asterisk crashes. the only debug information asterisk is leaving is segmentation fault, dumping core. anyone got a clue? i'm not running the latest CVS so maybe i should upgrade first... rgds, /staffan kerker -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Skickat: den 30 november 2003 20:04 Till: [EMAIL PROTECTED] Ämne: Re: [Asterisk-Users] asterisk server crashing From the console, I see where the call comes in and I can see where the party from the outside hangs up. The next thing that is said is as follows: libgcc_s.so.1 must be installed for pthread_cancel to work. Now I've taken a look on my system and I do in fact have the libgcc_s.so.1 on my system. The location is as follows: /lib/libgcc_s.so.1 It is part of the libgcc-3.2.2-5 package that I have installed on my system. I'm not a programmer, just a novice so I'm not quite sure how to run a backtrace or where the core file would be located. Thanks for your help so far. AJ On Sun, 30 Nov 2003, Roy Sigurd Karlsbakk wrote: - What's the console output after the crash when starting asterisk with -gvvvc? - After the crash, run a backtrace of the core file and send the output here ...perhaps this should be on the FAQ? ...and perhaps the FAQ should be linked to from asterisk.org? roy On Sunday, Nov 30, 2003, at 14:14 Europe/Oslo, [EMAIL PROTECTED] wrote: I deleted all the asterisk related directories and their subdirectories from /usr/src/ and did a brand new check out of zaptel, zapata, libpri, asterisk-addons and asterisk. AJ On Sat, 29 Nov 2003, Tilghman Lesher wrote: On Saturday 29 November 2003 20:36, [EMAIL PROTECTED] wrote: Quoting [EMAIL PROTECTED]: In the zaptel zapata and libpri directories I executed a make clean and did a cvs update and then ran make install. In the asterisk directory I did a make clean, a cvs update and a make upgrade. So I guess the answer to your question is yes I did take care of the other things as well. At least as far as I can see and as far as I know. AJ I don't know if your situation is the same as mine but I have been burned in the past by assuming that cvs update will provide all the lastest files. It only updates files that have previously been downloaded, soo, if you do not have a file that is now part of zaptel for instance, you will still not have that file. Do a fresh checkout to make sure you have all of the needed files. By the way, zapata is no longer needed. It has been incorporated into one of the others. Perhaps you mean subdirectories? True, 'cvs update' will not typically create new subdirectories, so you can do a 'cvs update -d' to have the update create new subdirectories, as 'cvs checkout' does, but 'cvs update' should create new files (in existing directories) just fine. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as SIP Proxy
Olle,
This is the complete extensions.conf. I wasnt getting the SIPDOMAIN right.
Rest of your script/configuration works only if ${SIPDOMAIN} works
Am I missing anything in this? I had the latest CVS checkout this morning,
i.e., 1st Dec. 12.00 Noon GMT +5.30.
;;;-
[globals]
[macro-stdexten]
exten = s,1,Dial(${ARG1},20,tr)
exten = s,2,Ringing
exten = s,3,Answer
exten = s,4,VoiceMail2,u${MACRO_EXTEN}
exten = s,5,Hangup
;All sip users will refer to this context
[pandora]
;;These two lines I added to test if domain dialing works
exten = evaro,1,Macro(stdexten,SIP/[EMAIL PROTECTED])
exten = john,1,Macro(stdexten,SIP/[EMAIL PROTECTED])
exten = 9001,1,Macro(stdexten,SIP/walter)
exten = 9002,1,Macro(stdexten,SIP/sridhar)
exten = 9003,1,Macro(stdexten,SIP/gopi)
exten = 9004,1,Macro(stdexten,SIP/jay)
exten = 9005,1,Macro(stdexten,SIP/ranga)
exten = 9006,1,Macro(stdexten,SIP/bharath)
;;;-;
And the outcome is as usual. SIPDOMAIN is blank. I checked 'grep SIPDOMAIN
chan_sip.c'. I found the line
pbx_builtin_setvar_helper(tmp, SIPDOMAIN, i-domain);
After this, I altered my extensions.conf like this.
;;--
[globals]
MYCURRENTDOMAIN=192.168.68.15
[macro-stdexten]
exten = s,1,Dial(${ARG1},20,tr)
exten = s,2,Ringing
exten = s,3,Answer
exten = s,4,VoiceMail2,u${MACRO_EXTEN}
exten = s,5,Hangup
;All sip users will refer to this context
[pandora]
exten = 9001,1,Macro(stdexten,SIP/walter)
exten = 9002,1,Macro(stdexten,SIP/sridhar)
exten = 9003,1,Macro(stdexten,SIP/gopi)
exten = 9004,1,Macro(stdexten,SIP/jay)
exten = 9005,1,Macro(stdexten,SIP/ranga)
exten = 9006,1,Macro(stdexten,SIP/bharath)
exten =_.,1,SetGlobalVar(sipto=${EXTEN})
exten =_.,2,SetGlobalVar(sipdom=${SIPDOMAIN})
;Every extension will go here, including h, t, s
;Filter out hangups
exten =_.,3,gotoif,$[${sipto} = h]?30|1:5|1
;---Test if external dial - on domain name
exten =5,1,gotoif($[${SIPDOMAIN} = ${MYCURRENTDOMAIN}]?20,1:10,1)
exten =10,1,Dial(SIP/[EMAIL PROTECTED])
exten =10,2,Hangup
exten =20,1,Goto(${sipto},1)
exten = 30,1,Hangup
;--;
And here is the console capture
-- Got SIP response 481 Subscription does not exist back from
192.168.68.12
-- Executing SetGlobalVar(SIP/sridhar-2364, sipto=evaro) in new
stack
-- Setting global variable 'sipto' to 'evaro'
-- Executing SetGlobalVar(SIP/sridhar-2364, sipdom=) in new stack
-- Setting global variable 'sipdom' to ''
-- Executing GotoIf(SIP/sridhar-2364, 0?30|1:5|1) in new stack
-- Goto (pandora,5,1)
WARNING[1217603008]: File ast_expr.y, Line 346 (ast_yyerror): ast_yyerror():
syntax error: parse error
-- Executing GotoIf(SIP/sridhar-2364, 0?20|1:10|1) in new stack
-- Goto (pandora,10,1)
-- Executing Dial(SIP/sridhar-2364, SIP/evaro@) in new stack
WARNING[1217603008]: File chan_sip.c, Line 749 (create_addr): No such host:
NOTICE[1217603008]: File app_dial.c, Line 516 (dial_exec): Unable to create
channel of type 'SIP'
== Everyone is busy at this time
-- Executing Hangup(SIP/sridhar-2364, ) in new stack
== Spawn extension (pandora, 10, 2) exited non-zero on 'SIP/sridhar-2364'
-- Executing SetGlobalVar(SIP/sridhar-2364, sipto=h) in new stack
-- Setting global variable 'sipto' to 'h'
-- Executing SetGlobalVar(SIP/sridhar-2364, sipdom=) in new stack
-- Setting global variable 'sipdom' to ''
-- Executing GotoIf(SIP/sridhar-2364, 1?30|1:5|1) in new stack
-- Goto (pandora,30,1)
-- Executing Hangup(SIP/sridhar-2364, ) in new stack
== Spawn extension (pandora, 30, 1) exited non-zero on 'SIP/sridhar-2364'
thanks and regards
-Ranga
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 29, 2003 11:53 PM
Subject: Re: [Asterisk-Users] Asterisk as SIP Proxy
You need to check the SIPDOMAIN early in the outgoing sip context for the
callee. I don't know your context here.
Also, please make sure you have an new CVS checkout, I don't know which
version
you're running. Run 'grep SIPDOMAIN' in chan_sip.c to make sure it's
there.
Here's what I do early in thte outgoing SIP context for clients:
;OUTGOING CALLS FROM SIP-
[sip-callers]
;Match everything
exten =_.,1,SetGlobalVar(sipto=${EXTEN})
exten =_.,2,SetGlobalVar(sipdom=${SIPDOMAIN})
;Every extension will go here, including h, t, s
;Filter out hangups
exten =_.,3,gotoif,$[${sipto} = h]?30|1:5|1
;---Test if external dial - on domain name
exten =5,1,gotoif($[${SIPDOMAIN} = ${MYCURRENTDOMAIN}]?20,1:10,1)
;--
MYCURRENTDOMAIN is set early in extensions.conf to the servers SIP realm.
/Olle
RE: [Asterisk-Users] cisco 7960 power suplies?
Also, I see them on eBay all the time for around $35 US. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lists Sent: Sunday, November 30, 2003 5:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cisco 7960 power suplies? Does anyone know where to get cisco 7960 power suplies? What should they cost? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cisco 7960 power suplies?
That is their new price Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: Monday, 1 December 2003 7:00 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] cisco 7960 power suplies? Also, I see them on eBay all the time for around $35 US. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lists Sent: Sunday, November 30, 2003 5:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cisco 7960 power suplies? Does anyone know where to get cisco 7960 power suplies? What should they cost? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as SIP Proxy
ranga wrote:
This is the complete extensions.conf. I wasnt getting the SIPDOMAIN right.
Rest of your script/configuration works only if ${SIPDOMAIN} works
Am I missing anything in this? I had the latest CVS checkout this morning,
i.e., 1st Dec. 12.00 Noon GMT +5.30.
Ranga,
I agree, seems like the client is not sending an INVITE that Asterisk
is able to parse the SIPDOMAIN from.
Turn on SIP DEBUG in your Asterisk CLI and catch the INVITE from the client.
Check if the invite goes to [EMAIL PROTECTED] or only to user without a domain?
I haven't got sjphone, so I can't try myself.
Please add a SIP DEBUG output with the INVITE.
/Olle
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Re: [Asterisk-Users] Asterisk as SIP Proxy
Here it goes
Sip read: CLI
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Content-Length: 116
Contact: sip:192.168.68.12
Call-ID: [EMAIL PROTECTED]
Content-Type: application/sdp
Max-Forwards: 70
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
CSeq: 1 INVITE
To: sip:[EMAIL PROTECTED]
Via: SIP/2.0/UDP 192.168.68.12:5060
v=0
o=- 3279257833 3279257833 IN IP4 192.168.68.12
s=-
c=IN IP4 192.168.68.12
t=0 0
m=audio 16390 RTP/AVP 8 0
10 headers, 6 lines
Using latest request as basis request
Sending to 192.168.68.12 : 5060 (non-NAT)
Found audio format ALAW
Found audio format UNKN
Capabilities: us - 524302, them - 12/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.68.12:5060
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as78933dd8
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm=asterisk, nonce=25230b01
Content-Length: 0
to 192.168.68.12:5060
Sip read: CLI
ACK sip:[EMAIL PROTECTED] SIP/2.0
Content-Length: 0
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as78933dd8
Via: SIP/2.0/UDP 192.168.68.12:5060
7 headers, 0 lines
Sip read: CLI
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Content-Length: 116
Contact: sip:192.168.68.12
Call-ID: [EMAIL PROTECTED]
Content-Type: application/sdp
Max-Forwards: 70
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
CSeq: 2 INVITE
To: sip:[EMAIL PROTECTED]
Via: SIP/2.0/UDP 192.168.68.12:5060
Proxy-Authorization: Digest
username=sridhar,realm=asterisk,nonce=25230b01,uri=sip:[EMAIL PROTECTED]
68.6,response=bb1576d7abea9f08c07d598c7d6686a0
v=0
o=- 3279257833 3279257833 IN IP4 192.168.68.12
s=-
c=IN IP4 192.168.68.12
t=0 0
m=audio 16390 RTP/AVP 8 0
11 headers, 6 lines
Using latest request as basis request
Sending to 192.168.68.12 : 5060 (non-NAT)
Found audio format ALAW
Found audio format UNKN
Capabilities: us - 524302, them - 12/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for ranga in pandora
list_route: hop: sip:192.168.68.12
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.68.12:5060
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as62db81f5
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 192.168.68.12:5060
-- Executing SetGlobalVar(SIP/sridhar-51cd, sipto=ranga) in new
stack
-- Setting global variable 'sipto' to 'ranga'
-- Executing SetGlobalVar(SIP/sridhar-51cd, sipdom=) in new stack
-- Setting global variable 'sipdom' to ''
-- Executing GotoIf(SIP/sridhar-51cd, 0?30|1:5|1) in new stack
-- Goto (pandora,5,1)
-- Executing GotoIf(SIP/sridhar-51cd, 0?20|1:10|1) in new stack
-- Goto (pandora,10,1)
-- Executing Dial(SIP/sridhar-51cd, SIP/ranga@) in new stack
== Everyone is busy at this time
-- Executing Hangup(SIP/sridhar-51cd, ) in new stack
== Spawn extension (pandora, 10, 2) exited non-zero on 'SIP/sridhar-51cd'
-- Executing SetGlobalVar(SIP/sridhar-51cd, sipto=h) in new stack
-- Setting global variable 'sipto' to 'h'
-- Executing SetGlobalVar(SIP/sridhar-51cd, sipdom=) in new stack
-- Setting global variable 'sipdom' to ''
-- Executing GotoIf(SIP/sridhar-51cd, 1?30|1:5|1) in new stack
-- Goto (pandora,30,1)
-- Executing Hangup(SIP/sridhar-51cd, ) in new stack
== Spawn extension (pandora, 30, 1) exited non-zero on 'SIP/sridhar-51cd'
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.68.12:5060
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as62db81f5
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 192.168.68.12:5060
Sip read: CLI
ACK sip:[EMAIL PROTECTED] SIP/2.0
Content-Length: 0
Call-ID: [EMAIL PROTECTED]
CSeq: 2 ACK
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as62db81f5
Via: SIP/2.0/UDP 192.168.68.12:5060
7 headers, 0 lines
localhost*CLI
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 01, 2003 2:16 PM
Subject: Re: [Asterisk-Users] Asterisk as SIP Proxy
ranga wrote:
This is the complete extensions.conf. I wasnt getting the SIPDOMAIN
right.
Rest of your script/configuration works only if ${SIPDOMAIN} works
Am I missing anything in this? I had the latest CVS checkout this
morning,
i.e., 1st Dec. 12.00 Noon GMT +5.30.
Ranga,
I agree, seems like the client is not sending an INVITE that Asterisk
is able to parse the SIPDOMAIN
Re: [Asterisk-Users] app_queue behavior followup
Joe Dennick wrote: I think you need to better define your Queue Environment in extensions.conf. Below is what I've got in mine, and it seems to work quite well: exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(welcome) exten = s,6,Queue(tech-queue) The queue definitions in queue.conf should take care of keeping the caller on hold until there is an agent available to take the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Yurchenko Sent: Sunday, November 30, 2003 8:33 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] app_queue behavior followup Anton Yurchenko wrote: also if I build my dialplan like : exten = 101,1,Answer exten = 101,2,Queue(phila) The musionhold plays only until the track is finished, and then it hangsup. How to make it loop? it seems that after the period defined in the timeout in queue.conf the call is dropped and the control goes to the t extension. I thought that the call would not be dropped and the all operators would be ringed again. is it so or is this an aobsolute timeout? -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Another * crash
put the core file into gdb, backtrace it and then we'll have some useful information: # gdb asterisk corefile and issue bt on gdb console or run asterisk directly into gdb : # gdb --args asterisk -vvvgc play with it and when it seg faults, issue a 'bt' command matteo. Il lun, 2003-12-01 alle 08:20, Kerker Staffan ha scritto: I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes. When I debug sip I get a noisy feedback from SER, and then asterisk crashes. the only debug information asterisk is leaving is segmentation fault, dumping core. anyone got a clue? i'm not running the latest CVS so maybe i should upgrade first... rgds, /staffan kerker -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Skickat: den 30 november 2003 20:04 Till: [EMAIL PROTECTED] Ämne: Re: [Asterisk-Users] asterisk server crashing From the console, I see where the call comes in and I can see where the party from the outside hangs up. The next thing that is said is as follows: libgcc_s.so.1 must be installed for pthread_cancel to work. Now I've taken a look on my system and I do in fact have the libgcc_s.so.1 on my system. The location is as follows: /lib/libgcc_s.so.1 It is part of the libgcc-3.2.2-5 package that I have installed on my system. I'm not a programmer, just a novice so I'm not quite sure how to run a backtrace or where the core file would be located. Thanks for your help so far. AJ On Sun, 30 Nov 2003, Roy Sigurd Karlsbakk wrote: - What's the console output after the crash when starting asterisk with -gvvvc? - After the crash, run a backtrace of the core file and send the output here ...perhaps this should be on the FAQ? ...and perhaps the FAQ should be linked to from asterisk.org? roy On Sunday, Nov 30, 2003, at 14:14 Europe/Oslo, [EMAIL PROTECTED] wrote: I deleted all the asterisk related directories and their subdirectories from /usr/src/ and did a brand new check out of zaptel, zapata, libpri, asterisk-addons and asterisk. AJ On Sat, 29 Nov 2003, Tilghman Lesher wrote: On Saturday 29 November 2003 20:36, [EMAIL PROTECTED] wrote: Quoting [EMAIL PROTECTED]: In the zaptel zapata and libpri directories I executed a make clean and did a cvs update and then ran make install. In the asterisk directory I did a make clean, a cvs update and a make upgrade. So I guess the answer to your question is yes I did take care of the other things as well. At least as far as I can see and as far as I know. AJ I don't know if your situation is the same as mine but I have been burned in the past by assuming that cvs update will provide all the lastest files. It only updates files that have previously been downloaded, soo, if you do not have a file that is now part of zaptel for instance, you will still not have that file. Do a fresh checkout to make sure you have all of the needed files. By the way, zapata is no longer needed. It has been incorporated into one of the others. Perhaps you mean subdirectories? True, 'cvs update' will not typically create new subdirectories, so you can do a 'cvs update -d' to have the update create new subdirectories, as 'cvs checkout' does, but 'cvs update' should create new files (in existing directories) just fine. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
David M. Wilson schrieb: Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. - 3x 6 b-channels. - ~20 Budgetone (and some others) handsets. Can anyone answer these questions: - Will the 3 ISDN cards function correctly in one host? - Will running all 3 cards flat out require particularly beefy hardware? - Will the Grandstream phones provide a good equivilant to professional dedicated PBX phones? (assuming a good network) I have read lots about echo problems and so on, is this an issue? Any help in the matter would be very much appreciated. Thanks in advance! You can try http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue behavior followup
Anton Yurchenko wrote: Joe Dennick wrote: I think you need to better define your Queue Environment in extensions.conf. Below is what I've got in mine, and it seems to work quite well: exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(welcome) exten = s,6,Queue(tech-queue) The queue definitions in queue.conf should take care of keeping the caller on hold until there is an agent available to take the call. how can I build a queue envirnoment so that, caller calls, the queue tries to reach the operators, and if they are all busy then it actually does Answer plays a specified message and then stars musiconhold and the caller is kept there until the queue times out. the object is not to pick up the phone immediatly, or should I requeue people again into the queue? like this: exten = 101,1,Queue(q1) ; queue has a timeout of 15 secs exten = t/101,1, Playback(all-busy) exten = t/101,2, Queue(q2); queue has a long timeout and musiconhold -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk behind NAT How to do it. (Leif Madsen)
I'm pretty sure that is incorrect. The inside_net is the ip address of the asterisk server, and the inside_mask is the subnet mask. At least that is how I have mine setup in my sip.conf, and it works. inside_mask for the internal mask would make more sense to me as well :) -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com In my configuration I have internal SIP clients registering from 192.168.0.0/28 and my * address is at 192.168.0.100. Using the host address of the * box as the inside_net variable the audio from 192.168.0.0/28 was sent to the outside_addr variable giving one-way speech. Setting internal_net to the subnet address of 192.168.0.0 and inside_mask to 255.255.255.0 the call behaved correctly. darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why * dont disconnect call.
hallo all. how i must setting extension.conf or other conf, if i want do this. If asterisk receive this error: Failed to authenticate on INVITE asterisk still giving normaly call, normaly signal dont fast busy. I want - fast busy, and disconnect all connection. (in sip.conf - have: reinvite=no). Radek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Party in Paris
Michael Devenijn wrote: count me in I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark I'll be there :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
Thomas Dingermann wrote: David M. Wilson schrieb: Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. - 3x 6 b-channels. - ~20 Budgetone (and some others) handsets. Can anyone answer these questions: - Will the 3 ISDN cards function correctly in one host? - Will running all 3 cards flat out require particularly beefy hardware? - Will the Grandstream phones provide a good equivilant to professional dedicated PBX phones? (assuming a good network) I have read lots about echo problems and so on, is this an issue? Any help in the matter would be very much appreciated. Thanks in advance! You can try http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO Thomas Better get tw quadbri's from kapejod at http://www.junghanns.net/ and combine them with ISDN phones. that would give you perfect quality. cheers Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * Party in Paris
Mark, We're happy to host something in London if you were dropping round these sides. Tan Telappliant.com Voiptalk.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: 30 November 2003 20:45 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] * Party in Paris Make sure and let us know anytime you are stopping by London.. :) (Just not between the 2nd-22nd December cos I will be away) Later.. Mark Spencer wrote: I'll be there until jan 5. The 19th would definitely be too early, maybe the 20-22? Possibly even after the new year, jan 2 or 3. Mark On Sun, 30 Nov 2003, zoa wrote: Count me and one of my collegue's in. How long are you staying in Paris ? The 19th might be a bit early for us, but then again maybe not :) Zoa. At 23:28 29/11/2003 -0600, you wrote: I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Asterisk European Tour: was RE: [Asterisk-Users] * Party in Paris
And while you are in Europe, why not also do Brussels ? ;) zoa. At 11:16 1/12/2003 +, you wrote: Mark, We're happy to host something in London if you were dropping round these sides. Tan Telappliant.com Voiptalk.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: 30 November 2003 20:45 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] * Party in Paris Make sure and let us know anytime you are stopping by London.. :) (Just not between the 2nd-22nd December cos I will be away) Later.. Mark Spencer wrote: I'll be there until jan 5. The 19th would definitely be too early, maybe the 20-22? Possibly even after the new year, jan 2 or 3. Mark On Sun, 30 Nov 2003, zoa wrote: Count me and one of my collegue's in. How long are you staying in Paris ? The 19th might be a bit early for us, but then again maybe not :) Zoa. At 23:28 29/11/2003 -0600, you wrote: I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
Michael Bielicki wrote: Thomas Dingermann wrote: David M. Wilson schrieb: Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. - 3x 6 b-channels. - ~20 Budgetone (and some others) handsets. Can anyone answer these questions: - Will the 3 ISDN cards function correctly in one host? - Will running all 3 cards flat out require particularly beefy hardware? - Will the Grandstream phones provide a good equivilant to professional dedicated PBX phones? (assuming a good network) I have read lots about echo problems and so on, is this an issue? Any help in the matter would be very much appreciated. Thanks in advance! You can try http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO Thomas Better get tw quadbri's from kapejod at http://www.junghanns.net/ and combine them with ISDN phones. that would give you perfect quality. Kapejod, have you released your cards ? cheers Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- - Best Regards, Pavel Litvinenko. ICQ: 16224754 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting disable in sip
Paul Liew wrote: - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 29, 2003 3:34 AM Subject: Re: [Asterisk-Users] call waiting disable in sip what would happend if all operators are busy? would app_queue exit? would it schedule the call to wait and until the number of them reaches the maxlen ( it is defined in queues.conf) ? Hi Anton, Before I submitted the patch to bugtracker to fix this problem, I tested this for both the Dial and Queue apps, and it works as per other channels, ie when all the queue operators are busy, the calling party will stay in the queue until an agent becomes free. All parameters within the queue.conf apply. The only parameter you need to specify in sip.conf is the incominglimit for this to work. For GS phones, set this to 1. By the way, this is no longer a patch as it has been incorporated into the CVS as of 26/11/03. Let me know if you encounter any problems. Paul I have a problem, when caller is in Queue and the operator is busy answering other call he/she still hears the call waiting signal. I have the latest cvs and incominglimit is set to 1. But here is what * shows when the operator is answering ( that is his phone is busy): UsernameincomingLimit outgoingLimit 107 0 1 0 1 and operator is getting a call waiting tone. Coould I be missing something? here is my sip.conf: [107] type=friend host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info defaultip=172.22.0.137 mailbox=201 ; Mailbox for message waiting indicator callerid=ipphone1 201 callgroup=1 pickupgroup=1 incominglimit=1 outgoinglimit=1 extensions.conf is very simple. it just calls Queue: exten = 101, 1, Queue(phila) may I be missing something in granstream phones? Thanks a lot -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk European Tour: was RE: [Asterisk-Users] * Party in Paris
And while you are in Europe, why not also do Brussels ? ;) zoa. Hey, surprise! Just discovered it on the web: http://graphics.cs.uni-sb.de/~rainer/tour.jpg Mark is going on tour! SCNR, Rainer -- http://graphics.cs.uni-sb.de/VoIP/ pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] call waiting disable in sip
On Mon, Dec 01, 2003 at 03:33:50PM +0200, Anton Yurchenko wrote: I have a problem, when caller is in Queue and the operator is busy answering other call he/she still hears the call waiting signal. I have the latest cvs and incominglimit is set to 1. But here is what * shows when the operator is answering ( that is his phone is busy): UsernameincomingLimit outgoingLimit 107 0 1 0 1 and operator is getting a call waiting tone. Coould I be missing something? here is my sip.conf: [107] username=107 // this is required for chan_sip.c to find the username. type=friend host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info defaultip=172.22.0.137 mailbox=201 ; Mailbox for message waiting indicator callerid=ipphone1 201 callgroup=1 pickupgroup=1 incominglimit=1 outgoinglimit=1 extensions.conf is very simple. it just calls Queue: exten = 101, 1, Queue(phila) Put the `username` parameter in your stanza of the sip.conf for the device. This is necessary for the incominglimit code to find the device that is making the call. If you look in your `debug` logs you'll probably see that the `user` variable is null. may I be missing something in granstream phones? Thanks a lot -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Second that ! -Original Message- From: Cees de Groot [mailto:[EMAIL PROTECTED] Sent: Monday, December 01, 2003 2:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Sofia (Bulgaria) !!! :))) Cees de Groot wrote: zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! Oslo! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Low, Adam wrote: Second that ! -Original Message- From: Cees de Groot [mailto:[EMAIL PROTECTED] Sent: Monday, December 01, 2003 2:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! Warsaw !! :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Cees de Groot wrote: zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! Feel forced to add STOCKHOLM! /O ;-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
On Fri, 28 Nov 2003 18:15:38 +0100, Peer Oliver schmidt wrote David M. Wilson wrote: Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. As far as I know, AVM only allows a single Fritz!Card PCI in a PC. I /think/ there is a patch out there to allow more than one. Search the archives to find out more. I am sure, you will get better results by putting in an active card. Either AVM or EICON. I have /heard/ the EICON cards are preferable because of the on board echo cancellation -- Best regards Peer Oliver Schmidt the internet company I'm using two Fritz!Cards in one box without any problems (yes, a patch was required iirc but I found it quite easily via google... somewhere on isdn4linux.de methinks). It works as a router/nat, iax-pstn gateway, mail server, nfs (/home) and smb file server and php+mysql app server for about a dozen clients. It worked for quite a time on a p2/400 but we're currently moving it to a celeron/1700 as the db is getting bigger. No asterisk related problems whatever, except for dead channels left sometimes (to clear them reliably, we need to shut down * and restart it). My 0.02PLN Grzegorz Nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Amsterdam!! I had my laptop and suitcase stolen in Amsterdam the one time I went there, after hearing someone talk about how safe a city it was over dinner. Most importantly, also stolen was my (apparently irreplacable) copyleft shirt (yellow/gold with large blue backwards (C) symbol on front and GPL preamble on back) which no amount of effort has managed to find a replacement for and it's *that* part i've never really gotten over. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Why not Sardinia, in Italy? good food, nice people :) and real italian pizza coffee.. matteo Il lun, 2003-12-01 alle 14:35, Cees de Groot ha scritto: zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
On Mon, 2003-12-01 at 15:17, Olle E. Johansson wrote: Cees de Groot wrote: zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! Feel forced to add STOCKHOLM! Well, in this case, I have to add BARCELONA !!! ;) /O ;-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Announcment while ringing
Hello, Can somebody help mw with set up Announcment while phone is ringing ? Is it suppouse to be like this: Dial(SIP/[EMAIL PROTECTED],A(test)) ? Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
I had my laptop and suitcase stolen in Amsterdam the one time I went there, after hearing someone talk about how safe a city it was over dinner. Most importantly, also stolen was my (apparently irreplacable) copyleft shirt (yellow/gold with large blue backwards (C) symbol on front and GPL preamble on back) which no amount of effort has managed to find a replacement for and it's *that* part i've never really gotten over. Those things generally happen in Amsterdam. And in Kristiania in Copenhagen. The usual problem: Smoking too much pot /me hides :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] door phone
For a door release, I have a cheap radio-shack device which is supposed to light up a lamp when a phone rings. Basically, it has a contact which is activated by the ring signal on a telephone line. I wired this up to the door release in the office, and have it hooked up to our (non asterisk) PBX. So, anyone can open the door by dialing the extension. The ringing itself opens it. The radio shack doojigger was probably about 10 bucks. -SteveK On Nov 26, 2003, at 6:12 PM, Jon Pounder wrote: Hi, Anyone know anything about Asterisk's support for door phones? Receiving the call from the door intercom system, opening the door, etc? Any hardware recommendations? I understand that the equipment we have now is Panasonic proprietary and came with the currently deployed Panasonic TD12-32 pbx. I just use an ordinary disposable phone, and put the zap channel in immediate mode. lift the phone and it starts to ring the extensions in the context it jumps to. I can also call the doorphone just like any other extension and it rings when I do so. Basically the keypad is ignored on the phone. as for door release, I have an electric strike on my dsc alarm system, but I have just not gotten around to making an agi that I can use to flip a bit on the parallel port and have that release the strike as well. (2n, a 10k resistor, and a pcb mount 12v relay, and a flyback diode, hooked up to the data bit, and in parallel with the alarm release relay.) We intend to deploy Asterisk in a 72 extensions + 16 trunks in a while, so any info will be great. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Mark Spencer wrote: Amsterdam!! I had my laptop and suitcase stolen in Amsterdam the one time I went there, after hearing someone talk about how safe a city it was over dinner. Most importantly, also stolen was my (apparently irreplacable) copyleft shirt (yellow/gold with large blue backwards (C) symbol on front and GPL preamble on back) which no amount of effort has managed to find a replacement for and it's *that* part i've never really gotten over. Mark I think many of us can understand that feeling.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as SIP Proxy
Ranga,
I'm sorry, I can't find the error in this configuration. I called on IP address myself,
and my Asterisk picked out the IP address into the domain part and dialed out.
I'm stuck. Anyone else that see the problem?
/O
ranga wrote:
Here it goes
Sip read: CLI
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Content-Length: 116
Contact: sip:192.168.68.12
Call-ID: [EMAIL PROTECTED]
Content-Type: application/sdp
Max-Forwards: 70
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
CSeq: 1 INVITE
To: sip:[EMAIL PROTECTED]
Via: SIP/2.0/UDP 192.168.68.12:5060
v=0
o=- 3279257833 3279257833 IN IP4 192.168.68.12
s=-
c=IN IP4 192.168.68.12
t=0 0
m=audio 16390 RTP/AVP 8 0
10 headers, 6 lines
Using latest request as basis request
Sending to 192.168.68.12 : 5060 (non-NAT)
Found audio format ALAW
Found audio format UNKN
Capabilities: us - 524302, them - 12/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.68.12:5060
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as78933dd8
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm=asterisk, nonce=25230b01
Content-Length: 0
to 192.168.68.12:5060
Sip read: CLI
ACK sip:[EMAIL PROTECTED] SIP/2.0
Content-Length: 0
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as78933dd8
Via: SIP/2.0/UDP 192.168.68.12:5060
7 headers, 0 lines
Sip read: CLI
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Content-Length: 116
Contact: sip:192.168.68.12
Call-ID: [EMAIL PROTECTED]
Content-Type: application/sdp
Max-Forwards: 70
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
CSeq: 2 INVITE
To: sip:[EMAIL PROTECTED]
Via: SIP/2.0/UDP 192.168.68.12:5060
Proxy-Authorization: Digest
username=sridhar,realm=asterisk,nonce=25230b01,uri=sip:[EMAIL PROTECTED]
68.6,response=bb1576d7abea9f08c07d598c7d6686a0
v=0
o=- 3279257833 3279257833 IN IP4 192.168.68.12
s=-
c=IN IP4 192.168.68.12
t=0 0
m=audio 16390 RTP/AVP 8 0
11 headers, 6 lines
Using latest request as basis request
Sending to 192.168.68.12 : 5060 (non-NAT)
Found audio format ALAW
Found audio format UNKN
Capabilities: us - 524302, them - 12/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for ranga in pandora
list_route: hop: sip:192.168.68.12
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.68.12:5060
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as62db81f5
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 192.168.68.12:5060
-- Executing SetGlobalVar(SIP/sridhar-51cd, sipto=ranga) in new
stack
-- Setting global variable 'sipto' to 'ranga'
-- Executing SetGlobalVar(SIP/sridhar-51cd, sipdom=) in new stack
-- Setting global variable 'sipdom' to ''
-- Executing GotoIf(SIP/sridhar-51cd, 0?30|1:5|1) in new stack
-- Goto (pandora,5,1)
-- Executing GotoIf(SIP/sridhar-51cd, 0?20|1:10|1) in new stack
-- Goto (pandora,10,1)
-- Executing Dial(SIP/sridhar-51cd, SIP/ranga@) in new stack
== Everyone is busy at this time
-- Executing Hangup(SIP/sridhar-51cd, ) in new stack
== Spawn extension (pandora, 10, 2) exited non-zero on 'SIP/sridhar-51cd'
-- Executing SetGlobalVar(SIP/sridhar-51cd, sipto=h) in new stack
-- Setting global variable 'sipto' to 'h'
-- Executing SetGlobalVar(SIP/sridhar-51cd, sipdom=) in new stack
-- Setting global variable 'sipdom' to ''
-- Executing GotoIf(SIP/sridhar-51cd, 1?30|1:5|1) in new stack
-- Goto (pandora,30,1)
-- Executing Hangup(SIP/sridhar-51cd, ) in new stack
== Spawn extension (pandora, 30, 1) exited non-zero on 'SIP/sridhar-51cd'
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.68.12:5060
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as62db81f5
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 192.168.68.12:5060
Sip read: CLI
ACK sip:[EMAIL PROTECTED] SIP/2.0
Content-Length: 0
Call-ID: [EMAIL PROTECTED]
CSeq: 2 ACK
From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as62db81f5
Via: SIP/2.0/UDP 192.168.68.12:5060
7 headers, 0 lines
localhost*CLI
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 01, 2003 2:16 PM
Subject: Re: [Asterisk-Users] Asterisk as SIP Proxy
ranga wrote:
This is the complete extensions.conf. I wasnt getting the SIPDOMAIN
right.
Rest of your script/configuration works only if ${SIPDOMAIN} works
Am I
Re: [Asterisk-Users] Announcment while ringing
no, the A option is used to play an announce to the called party as soon as he answers. matteo. Scrive Bartosz Jozwiak [EMAIL PROTECTED]: Hello, Can somebody help mw with set up Announcment while phone is ringing ? Is it suppouse to be like this: Dial(SIP/[EMAIL PROTECTED],A(test)) ? Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[offtopic] Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Those things generally happen in Amsterdam. And in Kristiania in Copenhagen. The usual problem: Smoking too much pot Actually we just had dinner and had left our things in his car which (according to the police inspector) was entered through the trunk using a half a tennis ball. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Show Dialplan
In my extensions.conf I have two 2 contexts (sip and pstn) with two extensions each one, but the command show dialplan, on the CLI, show me only the 2 contexts without any extensions and more the config in parking.conf. I can't do any call ! What I can do correct it ? Best regards, Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] door phone
That's an interesting solution. Caveats : - To have a door phone and door release would require 2 ports on the pbx unless you don't want to be able to call the doorphone, or release the door while the caller is still on the line. - if the doorphone is a regular phone and uses this method, anyone could simply apply 120VAC to the phone jack for it and presto, the door opens. For a door release, I have a cheap radio-shack device which is supposed to light up a lamp when a phone rings. Basically, it has a contact which is activated by the ring signal on a telephone line. I wired this up to the door release in the office, and have it hooked up to our (non asterisk) PBX. So, anyone can open the door by dialing the extension. The ringing itself opens it. The radio shack doojigger was probably about 10 bucks. -SteveK On Nov 26, 2003, at 6:12 PM, Jon Pounder wrote: Hi, Anyone know anything about Asterisk's support for door phones? Receiving the call from the door intercom system, opening the door, etc? Any hardware recommendations? I understand that the equipment we have now is Panasonic proprietary and came with the currently deployed Panasonic TD12-32 pbx. I just use an ordinary disposable phone, and put the zap channel in immediate mode. lift the phone and it starts to ring the extensions in the context it jumps to. I can also call the doorphone just like any other extension and it rings when I do so. Basically the keypad is ignored on the phone. as for door release, I have an electric strike on my dsc alarm system, but I have just not gotten around to making an agi that I can use to flip a bit on the parallel port and have that release the strike as well. (2n, a 10k resistor, and a pcb mount 12v relay, and a flyback diode, hooked up to the data bit, and in parallel with the alarm release relay.) We intend to deploy Asterisk in a 72 extensions + 16 trunks in a while, so any info will be great. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: * Party in Paris
On Sat, Nov 29, 2003 at 11:28:56PM -0600, Mark Spencer wrote: I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Hi Mark, Nice to hear you are coming to Paris! I am based in Paris and will certainly be around during these times. You can count me in for any meeting, presentation, event, drink, orgywhatever takes place. My company already sells some * integration services in France and would like to go one step further, especially with regard to Digium hardware. Please let me know if I can help with your logistics and planning. Cheers, -- Linux: The Ultimate NT Service Pack ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk European Tour: was RE: [Asterisk-Users] * Party in Paris
Hey, surprise! Just discovered it on the web: http://graphics.cs.uni-sb.de/~rainer/tour.jpg Mark is going on tour! Not sure if this is real info or just a JPG that someone created. Is Stuttgart a definate date on the 30th? If so, where in Stuttgart?? Robert Friedrichshafen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
At 08:33 1-12-2003 -0600, you wrote: Amsterdam!! I had my laptop and suitcase stolen in Amsterdam the one time I went there, after hearing someone talk about how safe a city it was over dinner. Most importantly, also stolen was my (apparently irreplacable) copyleft shirt (yellow/gold with large blue backwards (C) symbol on front and GPL preamble on back) which no amount of effort has managed to find a replacement for and it's *that* part i've never really gotten over. Hey Mark! Must say that was a stroke of bad luck :-P But don't let it stop you - there are thieving morons in every town, I'm sure of it :) One tip though, where-ever you may go: Make sure you've got plenty of Asterisk t-shirts in the suitcase, so if the culprit takes them, at least he may advertise for you :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Why stick with moving from city to city in Europe? Why not just rent the whole nation of Liechtenstein and have an * party? http://www.rentastate.com/en/flash5.html This (almost) unreal extension of capitalistic excess brings up an interesting point: if this whole nation is small enough to rent for a party, I think it would be a great idea to see if the whole nation would want to convert to Asterisk as their phone platform. It would be a great demo. Yes, we converted an entire nation over to Asterisk. I hereby volunteer myself for such a task, provided suitable compensation is involved. :-) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as SIP Proxy
Perhaps it's because the Contact: field does not have an extension in it, just an IP address? This is a guess without really thinking about it too much. JT Ranga, I'm sorry, I can't find the error in this configuration. I called on IP address myself, and my Asterisk picked out the IP address into the domain part and dialed out. I'm stuck. Anyone else that see the problem? /O ranga wrote: Here it goes Sip read: CLI INVITE sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 116 Contact: sip:192.168.68.12 Call-ID: [EMAIL PROTECTED] Content-Type: application/sdp Max-Forwards: 70 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 CSeq: 1 INVITE To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.68.12:5060 v=0 o=- 3279257833 3279257833 IN IP4 192.168.68.12 s=- c=IN IP4 192.168.68.12 t=0 0 m=audio 16390 RTP/AVP 8 0 10 headers, 6 lines Using latest request as basis request Sending to 192.168.68.12 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Capabilities: us - 524302, them - 12/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.68.12:5060 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 To: sip:[EMAIL PROTECTED];tag=as78933dd8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=25230b01 Content-Length: 0 to 192.168.68.12:5060 Sip read: CLI ACK sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 To: sip:[EMAIL PROTECTED];tag=as78933dd8 Via: SIP/2.0/UDP 192.168.68.12:5060 7 headers, 0 lines Sip read: CLI INVITE sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 116 Contact: sip:192.168.68.12 Call-ID: [EMAIL PROTECTED] Content-Type: application/sdp Max-Forwards: 70 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 CSeq: 2 INVITE To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.68.12:5060 Proxy-Authorization: Digest username=sridhar,realm=asterisk,nonce=25230b01,uri=sip:[EMAIL PROTECTED] 68.6,response=bb1576d7abea9f08c07d598c7d6686a0 v=0 o=- 3279257833 3279257833 IN IP4 192.168.68.12 s=- c=IN IP4 192.168.68.12 t=0 0 m=audio 16390 RTP/AVP 8 0 11 headers, 6 lines Using latest request as basis request Sending to 192.168.68.12 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Capabilities: us - 524302, them - 12/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for ranga in pandora list_route: hop: sip:192.168.68.12 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.68.12:5060 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 To: sip:[EMAIL PROTECTED];tag=as62db81f5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.68.12:5060 -- Executing SetGlobalVar(SIP/sridhar-51cd, sipto=ranga) in new stack -- Setting global variable 'sipto' to 'ranga' -- Executing SetGlobalVar(SIP/sridhar-51cd, sipdom=) in new stack -- Setting global variable 'sipdom' to '' -- Executing GotoIf(SIP/sridhar-51cd, 0?30|1:5|1) in new stack -- Goto (pandora,5,1) -- Executing GotoIf(SIP/sridhar-51cd, 0?20|1:10|1) in new stack -- Goto (pandora,10,1) -- Executing Dial(SIP/sridhar-51cd, SIP/ranga@) in new stack == Everyone is busy at this time -- Executing Hangup(SIP/sridhar-51cd, ) in new stack == Spawn extension (pandora, 10, 2) exited non-zero on 'SIP/sridhar-51cd' -- Executing SetGlobalVar(SIP/sridhar-51cd, sipto=h) in new stack -- Setting global variable 'sipto' to 'h' -- Executing SetGlobalVar(SIP/sridhar-51cd, sipdom=) in new stack -- Setting global variable 'sipdom' to '' -- Executing GotoIf(SIP/sridhar-51cd, 1?30|1:5|1) in new stack -- Goto (pandora,30,1) -- Executing Hangup(SIP/sridhar-51cd, ) in new stack == Spawn extension (pandora, 30, 1) exited non-zero on 'SIP/sridhar-51cd' Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.68.12:5060 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 To: sip:[EMAIL PROTECTED];tag=as62db81f5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.68.12:5060 Sip read: CLI ACK sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 To: sip:[EMAIL PROTECTED];tag=as62db81f5 Via: SIP/2.0/UDP 192.168.68.12:5060 7 headers, 0 lines localhost*CLI - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 01, 2003 2:16 PM Subject: Re: [Asterisk-Users] Asterisk as SIP Proxy ranga
RE: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Amsterdam!! I had my laptop and suitcase stolen in Amsterdam the one time I went there, after hearing someone talk about how safe a city it was over dinner. Most importantly, also stolen was my (apparently irreplacable) copyleft shirt (yellow/gold with large blue backwards (C) symbol on front and GPL preamble on back) which no amount of effort has managed to find a replacement for and it's *that* part i've never really gotten over. Mark Well the Aussie's recently announced an additional travel warning for The Netherlands due to the increased level of petty crime although I feel it was a little extreme. The petty crime problem is very much specific to Amsterdam and foreign crims come into the city specifically to target tourists and their valuables. I've lived out here for 3 years now and enjoy exceptional safety where I live in Haarlem so perhaps an alternative major city such as Haarlem or Den Haag might be an option ? Hmmm... what size was that T shirt ? (c; * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [offtopic] Re: [Asterisk-Users] Re: Asterisk European Tour: w as RE: * Party in Paris
Those things generally happen in Amsterdam. And in Kristiania in Copenhagen. The usual problem: Smoking too much pot Actually we just had dinner and had left our things in his car which (according to the police inspector) was entered through the trunk using a half a tennis ball. Mark Yep I have seen it done, its amazing, place half a tennis ball over the lock (with specific central locking systems from almost all manufacturers) and give it a punch and the air pressure does its magic ... * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Issues with Privacy Manager and Zapateller
Anyone have any thoughts on this since last week?
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller doesn't send the
tones
If I call from a phone after dialing *67 zapateller doesn't send the
tones
In the last 2 cases, the display on the phone shows -Blocked Call-
PrivacyManager always gives the following messages:
-- Executing PrivacyManager(SIP/8475551212-9ec4, ) in new
stack
-- CallerID Present: Skipping
Even when the phone shows -Blocked Call- and even when zapateller
sends
tones.
Here is the Dial-Plan for the extension
exten = _NXXNXX/,1,Zapateller
exten = _NXXNXX,1,NoOp
exten = 847666,2,PrivacyManager
exten = 847666,3,Dial(SIP/${EXTEN},,r)
exten = 847666,4,Hangup
Stephen
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RE: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Those things generally happen in Amsterdam. And in Kristiania in Copenhagen. The usual problem: Smoking too much pot I have to object to that, as a rule of thumb the Dutch only rob tourists who are dressed like tourists and act like tourists, that's what we all agreed to here and live by -- please just dress local and act local, so we can finally stop smoking pot just to keep up foreign misconception ... :-) Regards, Hans Vledder The Netherlands Hans although your somewhat right I don't think its fare to ask all tourists to leave their clothes at customs and to don clogs and ride a battered old bike around the city. I also must say that from my experience its very rarely (I've never heard of it) the native Dutch that perform these crimes. Sorry for the off topic ... * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting disable in sip
Walker Haddock wrote: On Mon, Dec 01, 2003 at 03:33:50PM +0200, Anton Yurchenko wrote: I have a problem, when caller is in Queue and the operator is busy answering other call he/she still hears the call waiting signal. I have the latest cvs and incominglimit is set to 1. But here is what * shows when the operator is answering ( that is his phone is busy): UsernameincomingLimit outgoingLimit 107 0 1 0 1 and operator is getting a call waiting tone. Coould I be missing something? here is my sip.conf: [107] username=107 // this is required for chan_sip.c to find the username. thanks, I think that is working i`ll try that in production environment, tommorow, and`ll report that. BTW right now without specifiing username, and incominglimit set to 1, I after a while see that it shows 1 but the phone is not in use at all. And this phone is stuck in this position until reload. Anyone have seen happen? type=friend host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info defaultip=172.22.0.137 mailbox=201 ; Mailbox for message waiting indicator callerid=ipphone1 201 callgroup=1 pickupgroup=1 incominglimit=1 outgoinglimit=1 extensions.conf is very simple. it just calls Queue: exten = 101, 1, Queue(phila) Put the `username` parameter in your stanza of the sip.conf for the device. This is necessary for the incominglimit code to find the device that is making the call. If you look in your `debug` logs you'll probably see that the `user` variable is null. may I be missing something in granstream phones? Thanks a lot -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk European Tour: was RE: [Asterisk-Users] * Party in Paris
rnc Info Lists wrote: Hey, surprise! Just discovered it on the web: http://graphics.cs.uni-sb.de/~rainer/tour.jpg Mark is going on tour! Not sure if this is real info or just a JPG that someone created. Is Stuttgart a definate date on the 30th? If so, where in Stuttgart?? Robert Friedrichshafen there's a sticker with sold out on top of that date : ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk European Tour: was RE: [Asterisk-Users] * Party in Paris
Not sure if this is real info or just a JPG that someone created. Is Stuttgart a definate date on the 30th? If so, where in Stuttgart?? These dates were just made up bye Rainer and me. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature
RE: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
Why not create a listing of the Asterisk resellers. Have a link off the main Digium page and post what asterisk services that particular reseller offers. This way people who are just getting into asterisk know where they can go for commercial support. Maybe the reseller could offer some sort of discounted rate for people who are referred to them through digium? This would boost sales for Digium big time because people like things that work right away and if they have the resources to quickly launch their Asterisk solution they will be motivated to invest in more hardware IMHO. Edwin Silva WW Works Inc. 3060 Mainway Dr. Unit 104 Burlington, ON L7M 1A3 -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED] Sent: Friday, November 21, 2003 12:38 AM To: [EMAIL PROTECTED] Subject: Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again) I am not sure a newbies list would help all that much, all that would happen is that they would cross post to both lists and we would get everything twice.. What may be better would be either a better way to search the list archive or a new users FAQ, of course the FAQ option requires that someone maintain it which is also a problem.. You know, it strikes me that the best group to service newbies is probably the resellers. Maybe there's a logical way to connect them together through a mailing list? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Destination number
Hello: I need to prepare some detailed stats from asterisk, and I'm asked to show data I don't know how to obtain it: It's the 'final' number (don't know what's its name) In the stats I have to show the caller_id (I have it), the called_id (I have it) and the final number that actually accepted the call. In extensions.conf file, I try to pass the call to several numbers in sequence so if one line is busy or doesn't answer I pass it to the next one. I have to know who answered the call, how can I do this? I'm currently looking for a Dial as the last command and getting the data for that command, but doesn't seem a solid solution. Best regards, Robert T. _ Una mejor experiencia en Internet. Prueba gratis dos meses MSN 8. http://join.msn.com/?pgmarket=es-esXAPID=1577DI=1055 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
What does it mean ?? WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 framesWARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 framesWARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
[Asterisk-Users] Call Announcement - How To ...
All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Il lun, 2003-12-01 alle 15:36, Brancaleoni Matteo ha scritto: Why not Sardinia, in Italy? good food, nice people :) Since this thread has already grown way larger than it should, may I add Venice? :) -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as SIP Proxy
John Todd wrote: Perhaps it's because the Contact: field does not have an extension in it, just an IP address? This is a guess without really thinking about it too much. The Contact: field sure looks weird, but the SIPDOMAIN comes from the INVITE - or? Ranga, please check your debug log in /var/log/asterisk too see if the SIP channel chokes on the Contact: field and gives up parsing. Maybe there's an error message in there. Just guessing here. /O Ranga, I'm sorry, I can't find the error in this configuration. I called on IP address myself, and my Asterisk picked out the IP address into the domain part and dialed out. I'm stuck. Anyone else that see the problem? /O ranga wrote: Here it goes Sip read: CLI INVITE sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 116 Contact: sip:192.168.68.12 Call-ID: [EMAIL PROTECTED] Content-Type: application/sdp Max-Forwards: 70 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 CSeq: 1 INVITE To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.68.12:5060 v=0 o=- 3279257833 3279257833 IN IP4 192.168.68.12 s=- c=IN IP4 192.168.68.12 t=0 0 m=audio 16390 RTP/AVP 8 0 10 headers, 6 lines Using latest request as basis request Sending to 192.168.68.12 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Capabilities: us - 524302, them - 12/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.68.12:5060 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 To: sip:[EMAIL PROTECTED];tag=as78933dd8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=25230b01 Content-Length: 0 to 192.168.68.12:5060 Sip read: CLI ACK sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 To: sip:[EMAIL PROTECTED];tag=as78933dd8 Via: SIP/2.0/UDP 192.168.68.12:5060 7 headers, 0 lines Sip read: CLI INVITE sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 116 Contact: sip:192.168.68.12 Call-ID: [EMAIL PROTECTED] Content-Type: application/sdp Max-Forwards: 70 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 CSeq: 2 INVITE To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.68.12:5060 Proxy-Authorization: Digest username=sridhar,realm=asterisk,nonce=25230b01,uri=sip:[EMAIL PROTECTED] 68.6,response=bb1576d7abea9f08c07d598c7d6686a0 v=0 o=- 3279257833 3279257833 IN IP4 192.168.68.12 s=- c=IN IP4 192.168.68.12 t=0 0 m=audio 16390 RTP/AVP 8 0 11 headers, 6 lines Using latest request as basis request Sending to 192.168.68.12 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Capabilities: us - 524302, them - 12/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for ranga in pandora list_route: hop: sip:192.168.68.12 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.68.12:5060 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 To: sip:[EMAIL PROTECTED];tag=as62db81f5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.68.12:5060 -- Executing SetGlobalVar(SIP/sridhar-51cd, sipto=ranga) in new stack -- Setting global variable 'sipto' to 'ranga' -- Executing SetGlobalVar(SIP/sridhar-51cd, sipdom=) in new stack -- Setting global variable 'sipdom' to '' -- Executing GotoIf(SIP/sridhar-51cd, 0?30|1:5|1) in new stack -- Goto (pandora,5,1) -- Executing GotoIf(SIP/sridhar-51cd, 0?20|1:10|1) in new stack -- Goto (pandora,10,1) -- Executing Dial(SIP/sridhar-51cd, SIP/ranga@) in new stack == Everyone is busy at this time -- Executing Hangup(SIP/sridhar-51cd, ) in new stack == Spawn extension (pandora, 10, 2) exited non-zero on 'SIP/sridhar-51cd' -- Executing SetGlobalVar(SIP/sridhar-51cd, sipto=h) in new stack -- Setting global variable 'sipto' to 'h' -- Executing SetGlobalVar(SIP/sridhar-51cd, sipdom=) in new stack -- Setting global variable 'sipdom' to '' -- Executing GotoIf(SIP/sridhar-51cd, 1?30|1:5|1) in new stack -- Goto (pandora,30,1) -- Executing Hangup(SIP/sridhar-51cd, ) in new stack == Spawn extension (pandora, 30, 1) exited non-zero on 'SIP/sridhar-51cd' Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.68.12:5060 From: Ranga Rao Vutukurusip:[EMAIL PROTECTED];tag=21632105 To: sip:[EMAIL PROTECTED];tag=as62db81f5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.68.12:5060 Sip read: CLI ACK sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK From: Ranga Rao Vutukurusip:[EMAIL
RE: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Hans although your somewhat right I don't think its fare to ask all tourists to leave their clothes at customs and to don clogs and ride a battered old bike around the city. I also must say that from my experience its very rarely (I've never heard of it) the native Dutch that perform these crimes. You forgot constant cheese eating, complaining to the Germans about the return of bicycles, insiting that the trains are better than the UK, all while while naked or shoving a banana in some orifice Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames
Don't use dtmfmode=inband on GSM codec it'll only work on G711. Martin On Mon, 1 Dec 2003, Bartosz Jozwiak wrote: What does it mean ?? WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[offtopic] RE: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Hmmm... what size was that T shirt ? (c; Large. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm Update available [Ringtones, Intercom, UI improvements]
On Mon, 01 Dec 2003 00:46:22 +0100, Brancaleoni Matteo [EMAIL PROTECTED] wrote: Hi. Isn't possible to have a statically linked version for linux? [EMAIL PROTECTED] iaxcomm]$ ./iaxcomm ./iaxcomm: error while loading shared libraries: libwx_gtk_xrc-2.4.so: cannot open shared object file: No such file or directory [EMAIL PROTECTED] iaxcomm]$ I replaced iaxcomm-lin-20031129.tar.gz with a new file (same name) that now has libwx-gtk-xrc-2.4.a linked in. Please let me know how it works for you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Mark Spencer [EMAIL PROTECTED] said: Most importantly, also stolen was my (apparently irreplacable) copyleft shirt (yellow/gold with large blue backwards (C) symbol on front and GPL preamble on back) which no amount of effort has managed to find a replacement for and it's *that* part i've never really gotten over. Yeah, the cities in the Netherlands suck. That's of course nothing too surprising, but someone telling you that Amsterdam is safe, especially regarding theft, it just plain stupid. Anyway, as soon as I manage to make some money with *, I promise to create a replacement t-shirt on cafepress.com and send it to you :-) -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Announcement - How To ...
Yes I would like it too ! - Original Message - From: Vledder, Hans [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 01, 2003 1:50 PM Subject: [Asterisk-Users] Call Announcement - How To ... All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I get caller's number in oh323 ?
We have an h.323 based IVR platform. When we make a call to it using an h.323 phone, it can see the callers number (ANI), but when we make a call to it via asterisk, the call goes through OK, but we don't get the number. How can I make this work? h323.conf === [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=alaw dtmfmode=inband [ivr] type=h323 context=default extensions.conf === exten = 602,1,Dial,h323/[EMAIL PROTECTED] exten = 602,2,HangUp Phil Skuse [EMAIL PROTECTED] *** UNIX System Administrator. NIC Handle: MBJEJPIEUI Vicorp UK Limited: The Telephony Engine Company. Tel +44 (0)1753 660523 http://www.vicorp.com *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] channel offset between Asterisk and PBX
You might need to edit the code of chan_zap.c You need two things to fix: outgoing calls and incoming calls. Outgoing you should be able to find pri_call call and do chan-1 for chans16. For incoming calls you need to find the handling of PRI_EVENT_RING or something like that and do chan+1 for chans16. regards Martin On Fri, 28 Nov 2003, Roman Sidler wrote: Hi We interfaced our ASCOTEL PBX to Asterisk. by EuroISDN PRI , DSS1 It works fine on channels 1- 15, but on 17-31 the miststood each other. Asterisk speaks in Timeslots, the PBX in B-channels The signalling is ok, but the bridging is shifted. The first incoming connection is bridged to nirwana also no indication is hearable, calling a second internal subcribes bridges them to the first. The PBX sends a SETUP message with channel identification 30 and Asterisk bridges them to Zap-30, instead of Zap-31. The configuration - Digium TE410p card, set for E1 in zaptel.conf span=1,1,1,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 in zapata.conf signalling = pri_cpe switchtype = euroisdn context = pri1-in pridialplan = unknown channel = 1-15 channel = 17-31 What's wrong? Thanks in advance Roman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
That's a good idea. There is already a resellers list on the Digium site, but perhaps a line or two about specialities could be added. Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edwin Silva Sent: Monday, December 01, 2003 4:26 PM To: [EMAIL PROTECTED] Subject: RE: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again) Why not create a listing of the Asterisk resellers. Have a link off the main Digium page and post what asterisk services that particular reseller offers. This way people who are just getting into asterisk know where they can go for commercial support. Maybe the reseller could offer some sort of discounted rate for people who are referred to them through digium? This would boost sales for Digium big time because people like things that work right away and if they have the resources to quickly launch their Asterisk solution they will be motivated to invest in more hardware IMHO. Edwin Silva WW Works Inc. 3060 Mainway Dr. Unit 104 Burlington, ON L7M 1A3 -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED] Sent: Friday, November 21, 2003 12:38 AM To: [EMAIL PROTECTED] Subject: Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again) I am not sure a newbies list would help all that much, all that would happen is that they would cross post to both lists and we would get everything twice.. What may be better would be either a better way to search the list archive or a new users FAQ, of course the FAQ option requires that someone maintain it which is also a problem.. You know, it strikes me that the best group to service newbies is probably the resellers. Maybe there's a logical way to connect them together through a mailing list? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
At 16:46 1-12-2003 +0100, you wrote: Well the Aussie's recently announced an additional travel warning for The Netherlands due to the increased level of petty crime although I feel it was a little extreme. The petty crime problem is very much specific to Amsterdam and foreign crims come into the city specifically to target tourists and their valuables. I've lived out here for 3 years now and enjoy exceptional safety where I live in Haarlem so perhaps an alternative major city such as Haarlem or Den Haag might be an option ? How about Enschede ? ;-) BTW Adam, kick your people please, I still haven't heard anything from them :-P Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Announcement - How To ...
--- Vledder, Hans [EMAIL PROTECTED] wrote: I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Use the A option to the Dial application: 'A(x)' -- play an announcement to the called party, using x as file Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PREPAID APPLECATION
I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart
Re: [Asterisk-Users] Outgoing-call and enter user in Conference - repost
Outlook Express mangled my message before, so I've reattached it... Hopefully, it'll go this time... - Original Message - From: Areski [EMAIL PROTECTED] To: Asterisk-Users Mailing-list [EMAIL PROTECTED] Sent: Tuesday, November 25, 2003 12:13 PM Subject: [Asterisk-Users] Outgoing-call and enter user in Conference - repost Hi all, Just wondering if someone have already done something like that : SIP Client_A --- 1)call --- ASTERISK --- 2)outgoingcall-PSTN--Client_B | | 3) Enter conference | MeetMe ' with user A Make 2 user in conference (point 1 and 2), it's definitely easy, but call an other user and put the both in conference,I still don't have any idea how to do it! I'm not speaking from experience, but couldn't you set up an extension for meetme, and just transfer your callers into it? 1) Make/take call. 2) transfer caller to meetme 3) dial another user 4) transfer to meetme 5) lather, rinse, repeat... - Andrew Thompson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Message Waiting Indicator Bugs?
I have had several cases where the message waiting indicator was stuck in the on state with Cisco 7960 SIP phones. Here are the two cases: 1. Single extension that mapped to a single voice mailbox. Restarting Asterisk or getting a new voicemail then clearing it fixed the problem. 2. Three SIP extensions that mapped to a single voice mailbox. Getting a new voicemail and then clearing it did not fix the problem. Have not restarted Asterisk to clear this but I assume it will work. Has anyone else seen problems like this? Is there any limitations on how many SIP extensions can share the same voicemail box as far as MWI goes? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Destination number
Hi! I have to know who answered the call, how can I do this? I'm currently looking for a Dial as the last command and getting the data for that command, but doesn't seem a solid solution. Very good question - I've also run into this problem. I do think that the CDR could use some improvement here. Next question: Under which circumstances is the value of billed seconds higher than that of the entire call's seconds? Yesterday I found such a record... Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk behind NAT How to do it. (Leif Madsen)
On Mon, 2003-12-01 at 05:52, Darren McIntosh wrote: In my configuration I have internal SIP clients registering from 192.168.0.0/28 and my * address is at 192.168.0.100. Using the host address of the * box as the inside_net variable the audio from 192.168.0.0/28 was sent to the outside_addr variable giving one-way speech. Setting internal_net to the subnet address of 192.168.0.0 and inside_mask to 255.255.255.0 the call behaved correctly. Aha! I had not tried this configuration. Now I see how that makes more sense! I will make note of that :) Thanks Darren! -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T400P and 2.4.23 kernels
Anyone able to confirm whether the T400P (or any other Zap device) works with the 2.4.23 kernels? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Mini meeting next week lubomir ? i'll be there starting on monday :) zoa. At 16:00 1/12/2003 +0200, you wrote: Sofia (Bulgaria) !!! :))) Cees de Groot wrote: zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PREPAID APPLECATION
--- Bartosz Jozwiak [EMAIL PROTECTED] wrote: I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Why don't you release it without the prompts then? It would probably be nice if you posted it to bugs.digium.com in experimental features or something like that. Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PREPAID APPLECATION
Bartosz Jozwiak wrote: I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. I don't know about the rights to the Cisco prompts, so be sure to remove them and then release it. User other sound files as props and make sure the scripts are there, so we know what to replace it with. Maybe someone can fund a recording with the Asterisk voice if the application is interesting enough. If you want it to be part of Asterisk after testing by the community, make sure you sign the disclaimer and fax to Digium. You'll find it on bugs.digium.com. You still have the copyright to your application, but give Digium the right to include it in Asterisk if they want to. Thank you for contributing to Asterisk! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PREPAID APPLECATION
Probably a good idea to re-record the prompts, to avoid intellectual property issues later on. Plus, you'll likely need to add more prompts in the future, and so you can have the voice match what you've already recorded. Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Monday, December 01, 2003 6:19 PM To: ASTERISK USERS Subject: [Asterisk-Users] PREPAID APPLECATION I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PREPAID APPLECATION
I would love to send it to couple of peoples so thay can write some docs and clean th code. OK ? Who would like to do it ? Bart - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 01, 2003 3:57 PM Subject: RE: [Asterisk-Users] PREPAID APPLECATION Probably a good idea to re-record the prompts, to avoid intellectual property issues later on. Plus, you'll likely need to add more prompts in the future, and so you can have the voice match what you've already recorded. Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Monday, December 01, 2003 6:19 PM To: ASTERISK USERS Subject: [Asterisk-Users] PREPAID APPLECATION I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PREPAID APPLECATION
I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart I will agree with the comments of others on this topic. You should _not_ include the prompts from Cisco. That is almost certainly a copyright violation. For a very low price, you can have Allison Smith (http://www.theivrvoice.com/) re-record the prompts, as she is the person that did almost all the current Asterisk vocalizations (except for the tt-monkeys.gsm file: that was me.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PREPAID APPLECATION
Speaking of voice prompts, could anyone tell me why the pre-recorded prompts sometimes sound garbled, but the voicemail messages themselves sound fine? Is it the format of the prompts? Stephen I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart I will agree with the comments of others on this topic. You should _not_ include the prompts from Cisco. That is almost certainly a copyright violation. For a very low price, you can have Allison Smith (http://www.theivrvoice.com/) re-record the prompts, as she is the person that did almost all the current Asterisk vocalizations (except for the tt-monkeys.gsm file: that was me.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Consultant / integrator needed
Hi All, I hope this is the right list for this sort of request. I'm wondering if you all could recommend (or are) an asterisk integrator. I've been following the lists, etc, and have played with the software, but just don't have the time to really figure it out, nor to deliver a solution in a fixed time. I need someone who can help me spec the hardware and configure * for a small office pbx. At the minimum I need really solid ACD functionality set up. We'll need to use a channel bank to terminate our current incoming centrex lines in the short term, although I'm open to recommendations as to how to better integrate with SBC. In the future I'd like to integrate outbound and possibly inbound VOIP as well. If you've got the time to and expertise to do some consulting, please contact me. We're in the southwest michigan area, so somebody local would be even better. I do have budget allocated for the project, and would like to get started ASAP. Thanks --erik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Yeah, the cities in the Netherlands suck. That's of course nothing too surprising, but someone telling you that Amsterdam is safe, especially regarding theft, it just plain stupid. I've heard that a Canadian not even living in Amsterdam told Mark that it was safe to put that stuff in the trunk. I am sure the junkie wasn't scoping you guys doing that and only was counting the cobblestones in the street :) Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
you are welcome zoa :) I'll be happy if we make a little * party here ;))) Lubo zoa wrote: Mini meeting next week lubomir ? i'll be there starting on monday :) zoa. At 16:00 1/12/2003 +0200, you wrote: Sofia (Bulgaria) !!! :))) Cees de Groot wrote: zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Cees de Groot wrote: Mark Spencer [EMAIL PROTECTED] said: Yeah, the cities in the Netherlands suck. That's of course nothing too surprising, but someone telling you that Amsterdam is safe, especially regarding theft, it just plain stupid. well. they'd be a lot better if there weren't all those stupid speed and red-light radars all over :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing-call and enter user in Conference - repost
Andrew Thompson wrote: Outlook Express mangled my message before, so I've reattached it... Hopefully, it'll go this time... mebbe you should switch to a better mailer, like mozilla for instance ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: Kerio SIPPS problems
Anyone have tried * with kerio SIPPS softphone? It registers ok with *, but I get missing sdp body message when dialing any extension. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Hi! I've heard that a Canadian not even living in Amsterdam told Mark that it was safe to put that stuff in the trunk. I am sure the junkie wasn't scoping you guys doing that and only was counting the cobblestones in the street :) During first visit to Amsterdam by car (with a German number plate - aah aah) exactly that cobblestone landed in my side window. Nothing stolen, just a friendly welcome message... ;- Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Announcement - How To ...
what would be nice is to get this on MeetMe app. so that you can announce someone joining the conf call Dave [EMAIL PROTECTED] 12/1/2003 11:11:49 AM --- Vledder, Hans [EMAIL PROTECTED] wrote: I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Use the A option to the Dial application: 'A(x)' -- play an announcement to the called party, using x as file Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm Update available [Ringtones, Intercom, UI improvements]
uh-oh :) 22:30:13: can't read from file descriptor 4 (error 21: Is a directory) 22:30:13: Failed to read PID from lock file. never used iaxcomm on that box :) Matteo. Il lun, 2003-12-01 alle 18:14, Michael Van Donselaar ha scritto: On Mon, 01 Dec 2003 00:46:22 +0100, Brancaleoni Matteo [EMAIL PROTECTED] wrote: Hi. Isn't possible to have a statically linked version for linux? [EMAIL PROTECTED] iaxcomm]$ ./iaxcomm ./iaxcomm: error while loading shared libraries: libwx_gtk_xrc-2.4.so: cannot open shared object file: No such file or directory [EMAIL PROTECTED] iaxcomm]$ I replaced iaxcomm-lin-20031129.tar.gz with a new file (same name) that now has libwx-gtk-xrc-2.4.a linked in. Please let me know how it works for you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial T option not obeyed with Grandstream BT101
http://bugs.digium.com It is appreciated if you submit your own code; otherwise I doubt anything will be done. On the Grandstream phones I think the call would be dropped if the transfer fails by disabling it in asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Barton Hodges Sent: Monday, December 01, 2003 10:41 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dial T option not obeyed with Grandstream BT101 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Barton Hodges Sent: Sunday, November 30, 2003 10:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dial T option not obeyed with Grandstream BT101 In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register = number:[EMAIL PROTECTED] extensions.conf: [from-sip] exten = s,1,Dial(SIP/111SIP/117) exten = 111,1,Dial(SIP/111,20) exten = 117,1,Dial(SIP/117,20) 1. The calling user dials number, which drops them into [from-sip] 2. Extensions 111 and 117 are Dialed. 3. The called user picks up extension 111. 4. The calling user presses Transfer on the Grandstream phone, then dials 117 and presses Send. 5. The called user on extension 111 is then transferred to extension 117. I don't believe this is supposed to happen because I have not specified the T option to the Dial command. Even without any options specified at all, both the calling and called users are able to transfer the call. I'm using a CVS snapshot from Sun, Nov 30th 04:04:45, 2003. What am I missing here? Barton [EMAIL PROTECTED] wrote: The T option is for the # transfer which is handled by Asterisk, in your case the phone has a transfer button and is able to send SIP messages telling Asterisk that the call should be transferred. That confirms my suspicions. What is the correct avenue for reporting this, and a few other problems as bugs? I am also interested in submitting some patches. Barton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nicolas Gudino wrote: | If the terminating tail circuit has cancelable echos and if the echo | canceler is enabled, you will hear echo for the first few utterances and | then it will die away. After a few seconds of speech, the echo should be | gone or at least very quiet compared to the echo level at the beginning of | the call. This is the signature of a working echo canceler. In my situation where we use VoIP softphones connecting to an X101P card to the PSTN, the other end hears us just fine and there is no echo. However, there is substantial echo on the VoIP clients. I did one extended test with X-Lite (or maybe it was DIAX, but I have the feeling the performance would have been the same) where I was in an extended conversation. Echo was quite bad for the first portion of the call, but after approximately 1 - 2 minutes, it progressively got better until after that time, it was still there, but at a very controlled, curtailed level. The beginning and end of the echoed portions were chopped off and the volume level of the echo was quite a bit lower than the audio level of the two parties, to the point that it was no longer distracting for the VoIP client to talk. This was using the MARK2 with AGGRESSIVE enabled suppressor. |Headsets are particularly notorious for poor echo performance.. This |is due to lack of acoustic isolation. Perhaps you could test using |headphones and a mic. What does this mean? That the earphones are feeding back into the wrap-around mic? I have a USB Plantronics DSP 400 headset running under ALSA sound system. Is it feeding through the plastic parts and entering the microphone sitting out near my mouth? - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/y7ULuYsUrHkpYtARAkxjAJ4jPSeS6vbcnLpEkUrYS/VrVs0klACfVuF7 npoIHQr3q6n/5H4zyrGoaWw= =XjPE -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PREPAID APPLECATION
I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart I will agree with the comments of others on this topic. You should _not_ include the prompts from Cisco. That is almost certainly a copyright violation. For a very low price, you can have Allison Smith (http://www.theivrvoice.com/) re-record the prompts, as she is the person that did almost all the current Asterisk vocalizations (except for the tt-monkeys.gsm file: that was me.) JT Warning: Making this type of announcement in a public forum apparently attracts Cisco's lawyers, who will scan your site. I received (less than two hours after posting this) a fax from Cisco's legal department demanding that I remove the (unbeknownst to me) confidential Cisco .pdf's from my web site, describing *XML configuration of Cisco devices. I have since complied, because I respect Cisco's legal ownership of such documentation, but I find the timing of their notice suspiciously close to my post on this list. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound Breaks
Hello. Do you have your linux starting in Graphical mode ( init mode 5 ) ? I also had a problem with audio on my sip phones and it was generated because of the frame buffering that my video drivers use ( I have * installed in my personal computer ), so I changed the startup mode to 3 and only startx when needed ( like when I am going to a DVD movie ; ). Luck. Carling R. Messina wrote: Hi I'm currently running asterisk with an fxo X100P and aTDM one port card in a small not world connected subnet, I've sucessfully setup two sip phone and one analog extension everything works fine with the analog phone but when you talk to someone on the sip phone the person at the sip phone can be heard with inteference. I've looked aroud the archives and found nothing specific. Can anybody give some pointers? Carling ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PREPAID APPLECATION
Sure I agree so I've already removed the cisco prompts. And will record something eals. Bart Quoting John Todd [EMAIL PROTECTED]: I would like to release prepaid application. But I have a small problem, we are using their Cisco prompts (nice lady voice) And I do not know if it is ok to release it. Bart I will agree with the comments of others on this topic. You should _not_ include the prompts from Cisco. That is almost certainly a copyright violation. For a very low price, you can have Allison Smith (http://www.theivrvoice.com/) re-record the prompts, as she is the person that did almost all the current Asterisk vocalizations (except for the tt-monkeys.gsm file: that was me.) JT Warning: Making this type of announcement in a public forum apparently attracts Cisco's lawyers, who will scan your site. I received (less than two hours after posting this) a fax from Cisco's legal department demanding that I remove the (unbeknownst to me) confidential Cisco .pdf's from my web site, describing *XML configuration of Cisco devices. I have since complied, because I respect Cisco's legal ownership of such documentation, but I find the timing of their notice suspiciously close to my post on this list. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceGlo
Hi, VoiceGlo is comercial version of Asterisk? :))) loo Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/
Re: [Asterisk-Users] Consultant / integrator needed
Erik, I have just finished a job in Toronto and will be in the area for a few days, I could arrange to stop by on my way home. Contact me off list - [EMAIL PROTECTED], 512-789-5214 Robert J Rae Softprofit Solutions - Original Message - From: Erik LaBianca [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 01, 2003 1:57 PM Subject: [Asterisk-Users] Consultant / integrator needed Hi All, I hope this is the right list for this sort of request. I'm wondering if you all could recommend (or are) an asterisk integrator. I've been following the lists, etc, and have played with the software, but just don't have the time to really figure it out, nor to deliver a solution in a fixed time. I need someone who can help me spec the hardware and configure * for a small office pbx. At the minimum I need really solid ACD functionality set up. We'll need to use a channel bank to terminate our current incoming centrex lines in the short term, although I'm open to recommendations as to how to better integrate with SBC. In the future I'd like to integrate outbound and possibly inbound VOIP as well. If you've got the time to and expertise to do some consulting, please contact me. We're in the southwest michigan area, so somebody local would be even better. I do have budget allocated for the project, and would like to get started ASAP. Thanks --erik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tone Detection Problem
This is a resend - customer complains thatsome phone systems they call don't respond to key tones, please advise. Thanks Rob. - Original Message - From: Softprofit Solutions To: [EMAIL PROTECTED] Sent: Monday, November 24, 2003 7:34 PM Subject: tone detection problem Some commercial systems don't hear the tones dialed from an analoghandset, same call is OK from cisco 7940 or sjphone. Calls are placed through asterisk T1 card w/12x12 carrier access channel bank, various analog sets. Thanks for your help. Robert J Rae Softprofit Solutions
