[Asterisk-Users] empty username in authorization section ?!
Hi all, Why does asterisk send empty username in authorization section ? My setup is as follows: SIP UA(x-lite) -- ASTERISK -- SER -- I am trying to have asterisk route calls to a sip proxy. The sip proxy requires authorization on calls to pstn. The problem is that asterisk sends an empty username in the Authorization section (INVITE message) in response to the Unauthorized challenge message sent by proxy: Authorization: Digest username="", realm="myrealm", algorithm=MD5, uri="sip:[EMAIL PROTECTED] Below is the configuration info and the full sip debug. Thanks, Dave Sip.conf [general] … register => 4804915:[EMAIL PROTECTED] [sip-proxy] type=peer secret=4804915 username=4804915 fromuser=4804915 host=myrealm [4804915] type=friend username=4804915 fromuser=4804915 secret=4804915 callerid="XX 1" <4804915> host=dynamic nat=no canreinvite=yes disallow=all allow=gsm SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.140:5060;branch=z9hG4bK4bd7f6a2;rport=5060;received=80.55.21.254 From: "XX 1" ;tag=as5fc122cc To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.e518 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE WWW-Authenticate: Digest realm="myrealm", nonce="41ac519aee7825573a272767883112d560eb089d" Server: Sip EXpress router (0.8.14-2 (i386/linux)) Content-Length: 0 Warning: 392 213.241.58.141:5060 "Noisy feedback tells: pid=1908 req_src_ip=80.55.21.254 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==1" Nov 30 10:50:10 VERBOSE[1089325632]: Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.140:5060;branch=z9hG4bK4bd7f6a2 From: "XX 1" ;tag=as5fc122cc To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.e518 Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 213.241.58.141:5060 Nov 30 10:50:10 VERBOSE[1089325632]: Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.140:5060;branch=z9hG4bK5c85c28f From: "XX 1" ;tag=as5fc122cc To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Authorization: Digest username="", realm="myrealm", algorithm=MD5, uri="sip:[EMAIL PROTECTED]", nonce="41ac519aee7825573a272767883112d560eb089d", response="1069242044fd7063e9579a5df68cbb83", opaque="" Date: Tue, 30 Nov 2004 09:50:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 263 v=0 o=root 3738 3739 IN IP4 192.168.2.140 s=session c=IN IP4 192.168.2.140 t=0 0 m=audio 15096 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 213.241.58.141:5060 Nov 30 10:50:10 VERBOSE[1089325632]: Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.140:5060;branch=z9hG4bK5c85c28f;rport=5060;received=80.55.21.254 From: "XX 1" ;tag=as5fc122cc To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.155e Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE WWW-Authenticate: Digest realm="myrealm", nonce="41ac519aee7825573a272767883112d560eb089d" Server: Sip EXpress router (0.8.14-2 (i386/linux)) Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SS7 to Pri
A switch ? ;-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joseph Sent: Friday, June 25, 2004 4:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SS7 to Pri Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] International Talking Clocks
What you can also do is call hotels (asking for prices ;-)), this would allow you test the quality in both directions - this is what I do when I need to test voice quality. Talking clocks may not always be accessible over voip - talking clock services may not be accessible to the operator that provides your voip provider with call termination in a given country. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Tuesday, June 15, 2004 1:36 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] International Talking Clocks At 1:34 PM -0700 on 6/14/04, Aaron Clauson wrote: >Hi, > >Does anyone know of a list of internationally >accessible PSTN talking clocks? > >I find talkjing clocks a good way to test the call >quality to a particular country. > >There are a quite a few available in the US but the >only other two countries I have found numbers for are >the UK and Sweden. Other countries obviously have them >but they generally don't seem accessible from >international numbers. > >Talking Clock Numbers: >Sweden: +46-3390510 >UK: +44-8451249068 >US: +1-2027621401 > >Anyone know (or provide access to) any others? > >Regards, >Aaron I find that a better solution is to use movie theaters. Google can usually find something in each nation/region/city, and most Western nations have answering machines on those lines which talk for quite a while. Talking clocks are getting harder to find, while movie theater auto-recordings are growing in number. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:Asterisk PRI calls to SER problem
brake up your dial plan on asterisk, only forward to ser numbers that
actually exist
on ser return 404 response error, for example, if the user is unavailable
BR
Dawid
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Aimable
Sent: Friday, June 11, 2004 3:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE:Asterisk PRI calls to SER problem
I have checked my SER configs and for cpb numbers validation I don't know
what it means .Can anyone who does help me?
Thanks
the reason is that you have a bug in your config files, most probably on SER
which sends provisional response instead of an error response to * which in
turn translates that to alerting on isdn. Verify your configs on SER and
make sure you send an error back to * when the sip phone is unavailbale. You
might also want to validate your cpb numbers on * so that if the number is
invalid you send back a release with invalid number format back to the
switch instead of forwarding the call to SER.
BR
Dawid
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Aimable
Sent: Friday, June 11, 2004 12:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk PRI calls to SER problem
Hi all,
I need help. I have a Linux box with SER as a proxy server with ip phones
attached on it , and another linux box with Asterisk and T410 card connect
to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk
and then to SER box and then to the phone .every time an invalid number
dialed from PSTN to SIP phones connected to SER asterisk says
that the call is progressing while it is not the case and send an alerting
message to the Nortel DMS switch attached to it. Is there any way I can
remove that alerting message and send the collect message to the switch? I
think that the reason is that * is not directly connected to the phones it
is calling
my setup is like this.
SIP
phones>SER--->Asterisk>PSTN(PRI
connected to NORTEL DES 100 switch)
I would like to find a way of
informing Asterisk that the call is progressing or something like that,
not ringing until it gets the correct message from SER .
I am using Asterisk CVS-04/06/04-10:46:21 on Red Hat 9 and Sip Express
Router version 12 on Red Hat 9.
I tried to use PRI_causes and "r" extension in extension.conf but still
the problem is there.
Any idea on how I can solve this problem?
--=_NextPart_000_0005_01C44FB3.2240AA20
Content-Type: text/html;
charset="us-ascii"
Content-Transfer-Encoding: quoted-printable
http://www.w3.org/TR/REC-html40"; xmlns:o =3D=20
"urn:schemas-microsoft-com:office:office" xmlns:w =3D=20
"urn:schemas-microsoft-com:office:word">
@page Section1 {size: 8.5in 11.0in; margin: 1.0in 1.25in 1.0in =
1.25in; }
P.MsoNormal {
FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: "Times New Roman"
}
LI.MsoNormal {
FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: "Times New Roman"
}
DIV.MsoNormal {
FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: "Times New Roman"
}
A:link {
COLOR: blue; TEXT-DECORATION: underline
}
SPAN.MsoHyperlink {
COLOR: blue; TEXT-DECORATION: underline
}
A:visited {
COLOR: purple; TEXT-DECORATION: underline
}
SPAN.MsoHyperlinkFollowed {
COLOR: purple; TEXT-DECORATION: underline
}
SPAN.EmailStyle17 {
COLOR: windowtext; FONT-FAMILY: Arial; mso-style-type:
personal-compose
}
DIV.Section1 {
page: Section1
}
the=20
reason is that you have a bug in your config files, most probably on SER =
which=20
sends provisional response instead of an error response to * which in =
turn=20
translates that to alerting on isdn. Verify your configs on SER and make =
sure=20
you send an error back to * when the sip phone is unavailbale. You might =
also=20
want to validate your cpb numbers on * so that if the number is invalid =
you send=20
back a release with invalid number format back to the switch =
instead=20
of forwarding the call to SER.
BR=20
Dawid
-Original Message-From:=20
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of=20
AimableSent: Friday, June 11, 2004 12:05 =
PMTo:=20
[EMAIL PROTECTED]Subject: [Asterisk-Users] =
Asterisk=20
PRI calls to SER problem
Hi=20
all,
I need help. I have a =
Linux box=20
with SER as a proxy server with ip phones attached on it , and another =
linux=20
box with Asterisk and T410 card connect to an E1 line .Whenever there =
is=20
a call from PSTN it is passed to Asterisk and then to SER box =
and then=20
to the phone .every time an invalid number dialed from PSTN to SIP =
phones=20
connected to SER asterisk says
that the call is =
progressing while=20
it is not the case and send an alerting message to the Nortel DMS =
switch=20
attached to it. Is there any way I can remove that alerting message =
and send=20
the collect
RE: [Asterisk-Users] Asterisk PRI calls to SER problem
the reason is that you have a bug in your config files, most probably on SER which sends provisional response instead of an error response to * which in turn translates that to alerting on isdn. Verify your configs on SER and make sure you send an error back to * when the sip phone is unavailbale. You might also want to validate your cpb numbers on * so that if the number is invalid you send back a release with invalid number format back to the switch instead of forwarding the call to SER. BR Dawid -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of AimableSent: Friday, June 11, 2004 12:05 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk PRI calls to SER problem Hi all, I need help. I have a Linux box with SER as a proxy server with ip phones attached on it , and another linux box with Asterisk and T410 card connect to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk and then to SER box and then to the phone .every time an invalid number dialed from PSTN to SIP phones connected to SER asterisk says that the call is progressing while it is not the case and send an alerting message to the Nortel DMS switch attached to it. Is there any way I can remove that alerting message and send the collect message to the switch? I think that the reason is that * is not directly connected to the phones it is calling my setup is like this. SIP phones>SER--->Asterisk>PSTN(PRI connected to NORTEL DES 100 switch) I would like to find a way of informing Asterisk that the call is progressing or something like that, not ringing until it gets the correct message from SER . I am using Asterisk CVS-04/06/04-10:46:21 on Red Hat 9 and Sip Express Router version 12 on Red Hat 9. I tried to use PRI_causes and “r” extension in extension.conf but still the problem is there. Any idea on how I can solve this problem?
RE: [Asterisk-Users] Asterisk and SER Setup Questions.
Hi
Shad,
1. You
configure that in extensions.conf
exten
=> _[prefix to forward to SER].,1,Dial(SIP/[EMAIL PROTECTED] SER
IP],10)
and
register your Asterisk to SER in sip.conf
register => asterisk:[EMAIL PROTECTED] SER
IP]/asterisk
2. you
can do that in extensions.conf for example
exten
=> _[prefix to forward to SER].,1,SetCallerID([prefix to append to CPA
number]${CALLERIDNUM})
regards,
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Shad
MortazaviSent: Tuesday, June 01, 2004 4:07 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk
and SER Setup Questions.
Dear
All,
I have the following
setup.
Quad T1's<->Asterisk
(PBX)<->(LAN<->DMZ)<->SER<->(Firewall)<->(Internet)
|
Local US Help Desk (Snom
200')
This setup works well. I
can pass calls from over the internet to the Asterisk PBX via SER using X-Ten
Lit.
I have a couple of
questions;
How do I tell Asterisk
to forward all outbound URI calls to the SER proxy? This works for anyone on
the ser itself, but what about someone on another system on the internet?
Lets say I wanted to call someone at IPTEL.ORG? How do I tell Asterisk to
forward calls that are not on it to ser?
How do I append the
caller ID so that my calls do not appear to come from Asterisk?
Thanks and
Regards
Shad
Mortazavi
---
Nexus Technical
Manager
n|m Nexus Management Inc
Sydney
RE: [Asterisk-Users] * as pri_net?
I have digium E1s as pri_net connected to nms based softswitch - no problems Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bruce Komito Sent: Friday, May 28, 2004 3:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * as pri_net? If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Wokflow diagram
Search google for something called 'sip scenario' - its a very nice command line win program for creating html based call flows. It can take Ethereal trace dumps and really works nice !! Regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ignace CARIA Sent: Friday, May 07, 2004 3:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Wokflow diagram Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mediatrix 1204 (4x FXO)
And what problem do you have with registering ? Jeremy Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you might reference that, configuring 1204 should be very similar to that of 1104. Regards, Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Wojciech TrycSent: Thursday, May 06, 2004 5:27 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Mediatrix 1204 (4x FXO) I have successfully implemented 1204 in semi production environment. Just want to share that it works very well, through the firewall (NATed). Unfortunately, it can not register with the server (and authenticate) but otherwise everything is fine. The audio quality is very good. Regards, Wojtek
RE: [Asterisk-Users] Ser and Asterisk together
#setflag(1);
# native SIP destinations are handled using our USRLOC DB
# going to our sip users ?
if (uri=~"sip:32679*" || uri=~"sip:58279*") {
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
# going to pstn
} else {
# };
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
# coming from fax ?
if (search("(f|From): [EMAIL PROTECTED]")) { # fax numbers
# forward to fax gw
rewritehostport("192.168.0.250:5060");
} else {
# forward to voice gw
rewritehostport("yyy.yyy.yyy.yyy:5060");
};
};
setflag(1);
route(1);
# if (!t_relay()) {
# sl_reply_error();
#};
}
route[1]
{
#if (method == "BYE" || method == "CANCEL"){
# setflag(1);
#};
# if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
# sl_send_reply("479", "We don't forward to private IP addresses");
# break;
# };
if (isflagset(6)){
force_rtp_proxy();
};
t_on_reply("1");
if (!t_relay()){
sl_reply_error();
};
}
onreply_route[1]
{
# nated ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]"){
fix_nated_contact();
force_rtp_proxy();
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Barry
Flanagan
Sent: Thursday, April 22, 2004 3:19 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Ser and Asterisk together
On Thu, 2004-04-22 at 13:47, Dawid Mielnik wrote:
> In my setup * is talking to sip us through ser - this is done by setting
the
> record route parameter in ser routing logic. A laso pass the media stream
> thorugh ser - this is done through the rtpproxy module (ser).
>
Any chance of seeing your ser.cfg file?
Thanks.
--
-Barry Flanagan
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ser and Asterisk together
In my setup * is talking to sip us through ser - this is done by setting the
record route parameter in ser routing logic. A laso pass the media stream
thorugh ser - this is done through the rtpproxy module (ser).
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Barry
Flanagan
Sent: Thursday, April 22, 2004 1:46 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Ser and Asterisk together
I am finally making some progress on this.
I now have SER passing off PSTN calls to * OK. Calls are being
connected, however, I don't hear anything on the SIP end, and asterisk
gives the following error:
WARNING[9226]: chan_sip.c:457 __sip_xmit: sip_xmit of 0x80e2dec (len
642) to 212.17.32.215 returned -1: Operation not permitted
Below is the context of this. I am using nathelper on SER, but I am not
at all confident of my config file (it being a patchwork of bits from
different examples. I attach my SER conf at the end of this message.
Should * be talking directly with the SIP UA, or should it be talking to
SER?
Any help would be appreciated! Even better would be a sample ser.cfg
which supports nathelper and using * for VM and PSTN!!
to 212.17.32.215:3568
Apr 22 12:31:38 WARNING[9226]: chan_sip.c:457 __sip_xmit: sip_xmit of
0x80e2dec (len 642) to 212.17.32.215 returned -1: Operation not
permitted
Retransmitting #2 (no NAT):
INVITE sip:[EMAIL PROTECTED]:3568 SIP/2.0
Via: SIP/2.0/UDP 212.17.35.184:5060;branch=z9hG4bK4c8263f2
From: ;tag=as4e38a4ab
To: "Ray Naughton"
;tag=e64bcbbe63564744
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 164
v=0
o=root 21443 21445 IN IP4 213.137.65.251
s=session
c=IN IP4 213.137.65.251
t=0 0
m=audio 16670 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
to 212.17.32.215:3568
Apr 22 12:31:39 WARNING[9226]: chan_sip.c:457 __sip_xmit: sip_xmit of
0x80e2dec (len 642) to 212.17.32.215 returned -1: Operation not
permitted
zeppelin*CLI>
= ser.cfg
#
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
#
# simple quick-start config script
#
# --- global configuration parameters
debug=7 # debug level (cmd line: -dd)
fork=no
log_stderror=yes # (cmd line: -E)
listen=213.159.144.8
#listen=127.0.0.1
# hostname matching an alias will satisfy the condition uri==myself".
alias=voip.edo.ie
alias=avmx.edo.ie
# Uncomment these lines to enter debugging mode
/*
debug=7
fork=no
log_stderror=yes
*/
check_via=no# (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
alias=voip.edo.ie avmx.edo.ie localhost
# -- module loading --
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# load the voicemail module
#loadmodule "/usr/local/lib/ser/modules/vm.so"
# load the enum module
loadmodule "/usr/local/lib/ser/modules/enum.so"
# load the group module, to verify if a user forwards to voicemail
loadmodule "/usr/local/lib/ser/modules/group.so"
# load the nathelper module
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# - setting module-specific parameters ---
# -- registrar parameter
# special NAT flag indicates that a registered client is behind NAT
modparam("registrar", "nat_flag", 6)
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
#modparam("usrloc", "db_url", "mysql://ser:[EMAIL PROTECTED]/ser")
modparam("usrloc|auth_db|acc|group|msilo|uri","db_url","mysql://ser:[EMAIL PROTECTED]
calhost/ser")
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
#modparam("auth_db", "db_url", "mysql://ser:[EMAIL PROTECTED]/ser")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- voicemail params --
#modparam("voicemail", "db_url","mysql://ser:[EMAIL PROTECTED]/ser")
# -- voicemail params --
#modparam("group", "db_url","mysql:/
RE: [Asterisk-Users] Ser and Asterisk together
We are using ser together with *. Ser is used as a SIP proxy/registrar, * is used as a sip - pstn gateway and voicemail/forward/conference server. Advanteges - scalable, very large number of sip clients with easier radius/database user management, advanced sip logic/routing options, better sip interoperability disadvantages - you've got two boxes, no iax on ser so you still have to manage iax users on asterisk In my opinion, if you plan to deploy large number of sip clients - it's a good idea Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of AJ Grinnell Sent: Wednesday, April 21, 2004 3:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Ser and Asterisk together Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Faxing
There is no support in Asterisk for FoIP. You might use alaw or ulaw g.711 if youre on the same LAN as your *. Otherwise its best effort transmission and not really reliable Regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Barry Fawthrop Sent: Sunday, February 22, 2004 4:10 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Faxing What is the best or simplest method to connect 4 fax machines into a * system? Fax -> ATA-186 -> Switch -> * Server -> VoIP or PSTN Fax -> * Fax Server with TDM 400P -> Switch -> * Server -> VoIP or PSTN Would like a dedicated # on the T1 to do direct to the fax machine. Would love your comments? Thanks in advance Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip - t.38 through asterisk
Hi, If I have two sip t.38 enabled gateways connected to asterisk, will I be able to send a fax from one to the other with the mediastream passing through asterisk ? Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] audiocodes and asterisk interoperability
Hi, Has anyone managed to register an audiocodes SIP device with asterisk ? No matter what I try it asterisk fails to authenticate the device. When I comment out secret= in sip.conf it registers without problems. Any ideas ? Best regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] softphone configs?
try removing the entry from sip.conf and putting autocreatepeer=yes, see if it registers then just a thought Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Wednesday, February 18, 2004 7:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] softphone configs? I checked them out and they looked fine...still got the error. I removed them out of the sip.conf entirely...still got the errorsigh... Thanks for the response though. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Girish Gopinath Sent: Wednesday, February 18, 2004 1:32 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] softphone configs? > >Feb 18 11:30:16 NOTICE[1125329600]: chan_sip.c:5577 handle_request: >Registration from 'Mark ' failed for >'192.168.5.64' > Pls check your 'username' and 'secret ' entries in your sip.conf (or remove those entries). I'm using SJPhone here. Girish _ Contact brides & grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] softphone configs?
check your username and authentication username / passwords on x-lite and verify that with sip.conf start off with the same entry for all then change passwords, then move on to authentication username (if you want to have it different) regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Wednesday, February 18, 2004 7:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] softphone configs? I've tried using the x-lite softphone as well as sjphone. I've gone over my configurations a dozen times...and I always seem to get the following error: Feb 18 11:30:16 NOTICE[1125329600]: chan_sip.c:5577 handle_request: Registration from 'Mark ' failed for '192.168.5.64' FYI...I'm trying to do all my voip internally, nothing to the outside world yet. If anyone could give me an idea I'd appreciate it. Thanks. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXO gateways on Asterisk
Hi, I have tested, and still am testing different sip equipment with asterisk. In my case I'm testing fxs gateways but this is what I have found so far: - audiocodes mp-104 fxs sip: I'm still testing this one but so far I can not get it to register properly with asterisk, it fails to authenticate. Messages are exchanged properly but asterisk just fails to accept its proxy-authorization data, as if md5 hashed password was not accepted. If you comment out secret= in sip.conf it registers properly, but there is no authenticartion. Sometimes I get retransmission RTP errors (asterisk/syslog). RFC2833 dtmf - ok. I heard that it hungs... ?? Pitty cause it looks solid, works quietly. - mediatrix 1104 fxs sip: Registers properly. RFC2833 dtmf signalling. G729 - ok. Very nice VAD mechanizm. Every now and then I got unhandled sip request errors (asterisk/syslog). Basically the only bad thing I can say about this box is that it's pretty loud. regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Clif Jones Sent: Wednesday, February 18, 2004 6:00 AM To: asterisk users Subject: [Asterisk-Users] FXO gateways on Asterisk I have been struggling with several mediocre SIP FXO gateways on Asterisk for the past 6 months and have found that each one has at least one major problem with it. I am looking for any success stories using 1 to 4 port SIP FXO gateways on Asterisk. I need for them to support RFC2833 DTMF bridging each way and G729 codecs. Multi-port FXO gateways need to have some sort of grouping and/or routing of SIP calls to specific channels. So far, I have evaluated an Audiocodes MP-104 (4-port) and a Multitech MVP-130 (1 port). If anyone has found something that works in these scenario's, I would love to hear from you. I want to deploy many small FXO gateways over a large geographical area and would like to find something that actually works. Thanks for the help! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AudioCodes MP-104 - help !
I have really got things upto a point where I have no clue why Asterisk doesnt authorise the audiocodes fxs box. I can not find anything in the archives - some posts that could actually be useful are already deleted. Can anyone please help me out ?! My setup: PSTN - Asterisk ---router/nat-- Audiocodes - POTS xxx.xxx.xxx.xxx 80.54.223.79 192.168.0.1192.168.0.249 Debug Log: Sip read: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKacEKUOEAs From: ;tag=1c20095 To: Call-ID: [EMAIL PROTECTED] CSeq: 16212 REGISTER Expires: 3600 Contact: ;expires=3600 Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 80.54.223.79 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKacEKUOEAs;received=80.54.223.79 From: ;tag=1c20095 To: ;tag=as0b288949 Call-ID: [EMAIL PROTECTED] CSeq: 16212 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.54.223.79:1025 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKacEKUOEAs;received=80.54.223.79 From: ;tag=1c20095 To: ;tag=as0b288949 Call-ID: [EMAIL PROTECTED] CSeq: 16212 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="3465f8ca" Content-Length: 0 to 80.54.223.79:1025 asterisk*CLI> Sip read: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKactrVcPNj From: ;tag=1c20095 To: Call-ID: [EMAIL PROTECTED] CSeq: 16213 REGISTER Contact: ;expires=3600 Proxy-Authorization:Digest username="mp_104_test",realm="asterisk",nonce="3465f8ca",uri="sip:xxx.xxx.xx x.xxx",Algorithm="MD5",response="a41319cc5c8f9ddab6be04b2afe3d0ba" Supported: em,timer,100rel Expires: 3600 User-Agent: Audiocodes-Sip-Gateway/MP-104 FXS/v.4.20.299.410 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 80.54.223.79 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKactrVcPNj;received=80.54.223.79 From: ;tag=1c20095 To: ;tag=as0b288949 Call-ID: [EMAIL PROTECTED] CSeq: 16213 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.54.223.79:1025 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.54.223.79;branch=z9hG4bKactrVcPNj;received=80.54.223.79 From: ;tag=1c20095 To: ;tag=as0b288949 Call-ID: [EMAIL PROTECTED] CSeq: 16213 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.54.223.79:1025 Feb 14 12:39:49 NOTICE[1133718080]: chan_sip.c:5405 handle_request: Registration from '' failed for '80.54.223.79' in sip.conf [mp_104_test] type=friend username=mp_104_test secret=mp_104_test auth=md5 disallow=all allow=g729 allow=alaw host=dynamic nat=yes qualify=200 dtmftone=rfc2833 context=default On the AudioCodes gateway I use authentication per endpoint. Thanks - I would really appreciate any help what so ever Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AudioCodes MP-104, register
Hi all, I am testing Audiocodes MP 104 fxs gateway with Asterisk but I already have problems with registering. I was wondering whether anyone has used AudioCodes fxs gateways with Asterisk and could help me out here. SIP debug log: to 80.54.223.79:1025 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.249;branch=z9hG4bKacMDmwMyQ;received=80.54.223.79 From: ;tag=1c20095 To: ;tag=as2491631b Call-ID: [EMAIL PROTECTED] CSeq: 27511 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="56a089dd" Content-Length: 0 asterisk*CLI> to 80.54.223.79:1025 asterisk*CLI> Sip read: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.249;branch=z9hG4bKacwSyaJmZ From: ;tag=1c20095 To: Call-ID: [EMAIL PROTECTED] CSeq: 27512 REGISTER Contact: ;expires=3600 Proxy-Authorization:Digest username="mp_104_test",realm="asterisk",nonce="56a089dd",uri="sip:xxx.xxx.xx x.xxx",Algorithm="MD5",response="a0a130b0820da3b7d67f88e858851814" Supported: em,timer,100rel Expires: 3600 User-Agent: Audiocodes-Sip-Gateway/MP-104 FXS/v.4.20.299.410 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.0.249 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.249;branch=z9hG4bKacwSyaJmZ;received=80.54.223.79 From: ;tag=1c20095 To: ;tag=as2491631b Call-ID: [EMAIL PROTECTED] CSeq: 27512 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.54.223.79:1025 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.249;branch=z9hG4bKacwSyaJmZ;received=80.54.223.79 From: ;tag=1c20095 To: ;tag=as2491631b Call-ID: [EMAIL PROTECTED] CSeq: 27512 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 80.54.223.79:1025 Feb 12 20:09:31 NOTICE[1133718080]: chan_sip.c:5405 handle_request: Registration from '' failed for '80.54.223.79' in sip.conf [mp_104_test] type=friend username=mp_104_test secret=mp_104_test auth=md5 disallow=all allow=g729 allow=alaw host=dynamic nat=yes qualify=200 dtmftone=rfc2833 context=default On the AudioCodes gateway I use authentication per endpoint. Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setting up callback
Is there any way to setup callback (for DISA) without going through writing an AGI script ? I have tried to use exten => h,1,System(callback) but this is what I get: Feb 12 10:09:29 WARNING[1082809536]: asterisk.c:255 listener: Select retured error: Interrupted system call Feb 12 10:09:29 WARNING[1236268096]: app_system.c:57 system_exec: Unable to execute 'callback' == Spawn extension (default, h, 1) exited non-zero on 'SIP/3267915-ab61' Yesterday this has also managed to crash asterisk. Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186
Cisco ATAs come in two types ATA186-I1 with 600 ohm impedance and ATA186-I2 with complex impedance (270 ohm in series with 750 ohm and 150 NF in parallel) What is the difference between the two ? Which one is suitable for Europe ? Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk and fax over ip - concept
That is a nice thought and takes care of problem with fax transmission over
IP. However not applicable in my case though. Most 'fax users' will not have
Asterisk on site but some other fxs gateways instead. Also from what I've
read RxFax tends to cause problems.
Best regards,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Peer Oliver
schmidt
Sent: Monday, February 09, 2004 9:52 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] asterisk and fax over ip - concept
Would the following be a doable solution:
1. An Asterisk-box on site with FXS
2. Plug Fax into FXS
3. User uses facsimile machine to call a number - Asterisk answers
4. Stores called number into variable ${FAXDESTINATION}
5. Use RcfFax of * to store fax within asterisk
6. mail stored fax together with ${FAXDESTINATION} off to central office
with outgoing PSTN?
No PSTN line needed, user does not need to change his/her way of doing
things.
If you want to provide feedback with regards of fax success, prefix the
phonenumber with a 3 letter account number specifying the sender to be
used for later eMail notification.
Just a thought.
rgds
pos
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and fax over ip - concept
I will be using Asterisk to connect remote offices to PSTN - over IP (SIP). These offices will use fxs gateways such as madiatrix and audiocodes to send VoIP traffic to Asterisk. Asterisk will in turn push their traffic to PSTN. The other way round will also work ie. Asterisk will forwards traffic from PSTN to those gateways (offices). The problem that I have though is that most companies send and recive faxes - and so will these. I dont want these offices to bypass faxes directly to PSTN but would somehow like to workaround fax over ip and Asterisk. What came to my mind was to add an fxo gateway (connected to PSTN) beside Asterisk to break out faxes to PSTN. Such a fxo gateway could communicate with the fxs gateways at customer premises and could reliably transmit faxes using t.38. The way I would imagine it is as such. When Asterisk would see a 'call' coming from a specified number (assigned to a sip port of an fxs gateway) instead of bridging the call directly to PSTN over a zap interface the 'call' coule be forwarded to the fxo gateway (over sip) that could reliably handle the 'call' (fax transmission). Would the t.38 transmission be properly handled by the t.38 supporting end points whith mediastrem passing through Asterisk ? (dont have much experience with t.38) Has anyone ever tried anything similar / different / wierder to try and deal with fax over ip and Asterisk ? Any suggestions and comments are welcome. Regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial via sip gateway?
The mediatrix does have unique username/passwd for each port. At least the 1104 FXS does. Each port can be registered separately with *. I assume other way round should work as well then. regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Greg Hill Sent: Sunday, February 01, 2004 10:01 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial via sip gateway? On Sun, 1 Feb 2004, Rich Adamson wrote: > I don't believe the above will work. There is only one IP address for > the box, and no way that I've found to send a sip packet to the box with > "additional" information that would suggest using port 1 vs port 2. From > what others have hinted at (and it seems the majority of us are limited > either to what's printed or experimentation), the 1204 has an internal > function that kind of resembles a trunk group. "It" decides which port > to use. > > As mentioned previously, the sip "register" function in the box is inop > in both directions, therefore there does not seem to be a way to address > the ports through contexts or anything else. Mediatrix has provided the > mib variables where one can enter a different password for each port, > but that has no value either since the register function doesn't work. What happens if you don't use a register => line in sip.conf, but do include a section like: [mediatrixport1] username= password= host= Just to check my theory, I did some testing via fwd. I discovered that if I include a register => line with my fwd info, then when I call my fwd number (outbound through iaxtel) it rings in. But I can't call out via fwd. So then I put in my [fwd] service definition, removed the register line, and waited for the old registration to expire. Then I tried calling my fwd number (again through iaxtel). This time I got the message about the user being offline. But now I can call out via fwd, even though calls wouldn't come in. This demonstrates that the [fwd] section is used by Dial() when I try to place a call out through that service, and that the register line isn't needed for the outbound call. Somebody mentioned that the mediatrix lets you set a unique username/password for each of its ports. It seems that you could set up four service definitions, each using a different user/pwd pair. Then * will use a different user/pwd pair to log in to the mediatrix, depending upon which service definition was called for by the Dial() statement. Or does the mediatrix not really have a distinct user/pwd pair for accessing each port? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mediatrix, dtmf
Hi, Thanks to the lag, I have sorted this out myself. Out of band singalling, codec change fixed it. Regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dawid Mielnik Sent: Friday, January 30, 2004 2:07 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] mediatrix, dtmf Hi, I have problems with Asterisk recognizing dtmf tones sent by Mediatrix 1104 FXS. I can not enter mailbox number (voicemail) or pin code (meet-me). Asterisk shows 'username not entered' when dialing in voicemail. Both asterisk and Mediatrix have set inband dtmf. Can anyone help me out ? Best regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] List traffic
Michael, I have the same thing -- 1 to 4 posts a day ! regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Graves Sent: Thursday, January 29, 2004 12:06 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] List traffic All of a sudden my list traffic appears to have dropped to a few messages/day the past few days. I anyone else seeing this as well? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] "...I believe in love, its all we've got." - Elton John ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mediatrix, dtmf
Hi, I have problems with Asterisk recognizing dtmf tones sent by Mediatrix 1104 FXS. I can not enter mailbox number (voicemail) or pin code (meet-me). Asterisk shows 'username not entered' when dialing in voicemail. Both asterisk and Mediatrix have set inband dtmf. Can anyone help me out ? Best regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP - fax / voicemail
Hi, Just to clear things out.. Can asterisk transmit faxes over IP ? If not, are there any works being done towards implementing t.38 on asteisk ? Also dialing in from a mediatrix fxs sip gateway to voicemail, asterisk does not see the digits entered after mailbox prompt. I have dtmftone settings correct - inband (also tried others to make sure), however asterisk shows 'username not entered'. Any clues how to tackle this ? Chenking voicemail from x-lite for example I dont have problems. regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk 0.7.1 - mysql
WipeOut, nope, did not build asterisk-addons thanks.. regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Thursday, January 22, 2004 2:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk 0.7.1 - mysql Dawid Mielnik wrote: >Hi, > >Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this >new version of * only work through ODBC ? Do I have connect to MySQL through >ODBC now ? > >Regards, > >Dave > >_ > Did you rememebr to build the Asterisk-Addons??.. The MySQL support has removed from the Asterisk core a while back and is now in "asterisk-addons" on the CVS server.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk 0.7.1 - mysql
Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? Regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1104 register problem ?
Hi, I am trying to test a Mediatrix 1104 FXS SIP gateway with Asterisk, but I have some problems. When registering the Mediatrix gw doesnt respond to Asterisk's 'proxy authrisation required' messages as if it didnt understand them. Strnage thing, when I have type=friend, asterisk says that the Mediatrix is unauthorised - get fast busy in handset. When I put type=peer in sip.conf, I can dial out through Asterisk to PSTN - the phone rings but thereis no voice channel - both ends cant hear anything. I have seen some posts on the list suggesting a bug on Asterisk (chan_sip.c) in REGISTER message response: http://www.mail-archive.com/[EMAIL PROTECTED]/msg01887.html But I have also seen posts where someone has acutally setup Asterisk with a Mediatrix SIP gateway: http://lists.digium.com/pipermail/asterisk-users/2003-November/027761.html The Mediatrix gw is behind nat/firewall, but I dont think this is the problem (working to rule this out). Can anyone help me out ? Does anyone have experience with Mediatrix and Asterisk ? Best regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] open h323
Has anyone installed openh323 on redhat 9 ? I have problems compiling pwlib: In file included from /usr/include/openssl/ssl.h:179, from ../../ptclib/pssl.cxx:195: /usr/include/openssl/kssl.h:72:18: krb5.h: No such file or directory >From what I see (google) there seems to be a general problem with pwlib, openssl and redhat 9. Can anyone help me out ? Regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pri gateways and asterisk
Hi all, I am planning to use VoIP gateways to connect remote offices to Asterisk. Not having much experience with these and Asterisk I would appreciate any info/experience that you might share with me as to their interoperability with Asterisk. I am interested with in rather bigger gateways (order of E1's) from: AudioCodes - Mediant Mediatrix - 1531 Quintum tenor Multupath D3000 Has anyone used these with Asterisk ? I would appreciate any comments you might have regarding these. Best regards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] one way choppy sound problem !
Hi ! To TeleSIP, Steve Dolloff, Nicolas Gudino, Steven Critchfield and all of you who have had problems with one way periodic choppy sound problem in the direction SIP ---> PSTN. The source of the problem, in my case was a worm on my windows os. The worm(s) used DoS against microsoft's ip addresses. It sent a large number of SYN requests every second which caused caused my out bandwidth to jump to about 40 Kbps and probably queued/delayed the RTP stream and in turn causing the chops in sound heard on the PSTN side. Hope this helps for the rest of you with similar problems. The worms I have found were: worm.blaster.A - msblast.exe worm.blaster.E - mslaugh.exe worm.blaster.F - TFTP1516 Regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dawid Mielnik Sent: Monday, January 05, 2004 1:29 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] one way choppy sound problem ! Hi Again, Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and some others. I have tried different codecs - GSM, aLAW uLAW. They all give the same result. In the direction PSTN user ---> Softphone user the sound is crystal clear (also tried on a dial-up connection), in the other direction however the sound is a bit choppy. The chops occur at regular intervals of time, at about 1-2 seconds !? When analyzing *'s ethernet interface with tcpdump (raw tcpdump -i eth0) I have noticed that the scrolling slows down during the times when chops occur in the sound. I have tested things using different softphones and different internet connections (user side) - always yelding the same result. In other words this is probably a problem on asterisk, either the hardware (ehternet interface/E100p) or a swoftware bug, incoming RTP buffering maybe ? Has anyone actually obtained a good quality sound in a similar setup ? Internet 2 x E1 x-lite <---> Asterisk ---> PSTN Any help appreciated ! Best regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nicolas Gudino Sent: Friday, January 02, 2004 6:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] one way choppy sound problem ! I have a similar problem, with GS phones, X-Lite or Kphone. I tried all the codecs with the same result. Choppy sound in the direction SIP-Phone -> pstn, but crystal clear sound the other way around. The only difference in my case is that I have two asterisks servers connected together via IAX2, the PSTN call is received in one asterisk, while the sip phones are in the other asterisk. Ex: pstn -> * --iax2--> * ->sip phone (GS, Xlite or Kphone) If I use an Xlite in the same asterisk as the pstn line, the sound is perfect in both ways. But when I answer the call in the second asterisk, the sound from the sip phone to pstn is choppy, with or without silence detection, and the sound from pstn to sip phone is perfect. The asterisk server with the pstn line is an old pentium 133, maybe thats the problem, I will try with a better machine and see how it goes. On Fri, 2004-01-02 at 06:23, Dawid Mielnik wrote: > Hi all, > > I have my asterisk setup as following: > > IP 2 x E1 > x-lite <---> Asterisk ---> PSTN > > > When I place a call from x-lite to PSTN, the quality of the sound in the > direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, > heard by the PSTN user is choppy and makes communication not very pleasant. > The sound is choppy as if bits of data were lost. The strange thing is that > the x-lite user hears the PSTN user fine ! > > In x-lite, I have swithed off sience detection (transmit silence - yes), > this has improved the sound quality but did not eliminated the problem. I > have fed a countinious sound into the microphone and still got chops in the > sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get the > same problem with all of them. Maybe the problem lies somewhere in audio > buffering settings on x-lite ? > > Has anyone ever had this sort of problem and managed to deal with it ? I > would greatly appreciate your help ! > > Best regards, > > Dave > > > ___ > 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] manipulating with numbers - StripMSD, Prefix
Tilghman,
That looks even beter than with StripMSD, Prefix !
Thanks,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher
Sent: Wednesday, January 07, 2004 5:27 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] manipulating with numbers - StripMSD, Prefix
On Wednesday 07 January 2004 06:06, Dawid Mielnik wrote:
> Hello,
>
> I can not seem to be able to get StripMSD and Prefix to work for me
> in extensions.conf. This is an example of what I have:
>
> exten => _050.,1,StripMSD,1
> exten => _50.,Prefix,01051
>
> exten => _001051.,1,Dial(${TRUNK2}/${EXTEN})
> exten => _001051.,2,Busy
> exten => _001051.,102,Busy
>
> What I want to achieve is to call 001051501657887 on TRUNK2 when
> dialing 0501657887.
Here's an idea - don't use StripMSD and Prefix anymore, as there are
better options now:
exten => _050.,1,Dial(${TRUNK2}/001051${EXTEN:1})
exten => _050.,2,Busy
-Tilghman
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] manipulating with numbers - StripMSD, Prefix
Hi again,
Sorry, completely forgot about setting the priorities - all is OK
Regards,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dawid Mielnik
Sent: Wednesday, January 07, 2004 1:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] manipulating with numbers - StripMSD, Prefix
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me in
extensions.conf. This is an example of what I have:
exten => _050.,1,StripMSD,1
exten => _50.,Prefix,01051
exten => _001051.,1,Dial(${TRUNK2}/${EXTEN})
exten => _001051.,2,Busy
exten => _001051.,102,Busy
What I want to achieve is to call 001051501657887 on TRUNK2 when dialing
0501657887.
dialing 0501657887:
-- Executing StripMSD("SIP/3267915-7e9a", "1") in new stack
...this is the only thing I get on the console, stuck ?
dialing 501657887:
-- Executing StripMSD("SIP/3267915-7e9a", "1") in new stack
-- Executing StripMSD("SIP/3267915-4297", "1") in new stack
-- Executing Prefix("SIP/3267915-285a", "001051") in new stack
-- Prepended prefix, new extension is 001051501657887
-- Executing Busy("SIP/3267915-285a", "") in new stack
== Spawn extension (demo, 001051501657887, 2) exited non-zero on
'SIP/3267915-285a'
...the call fails
dialing 0010515016571887:
-- Executing Dial("SIP/3267915-43af", "Zap/g2/001051501657887") in new stack
-- Called g2/001051501657887
-- Zap/32-1 is making progress passing it to SIP/3267915-43af
-- Hungup 'Zap/32-1'
- the call gets through.
My question, how to manipulate with 05016576887 to obtain 001051657887 ? -
seems like very trivial but doesent want to work for me.
Thanks
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] manipulating with numbers - StripMSD, Prefix
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me in
extensions.conf. This is an example of what I have:
exten => _050.,1,StripMSD,1
exten => _50.,Prefix,01051
exten => _001051.,1,Dial(${TRUNK2}/${EXTEN})
exten => _001051.,2,Busy
exten => _001051.,102,Busy
What I want to achieve is to call 001051501657887 on TRUNK2 when dialing
0501657887.
dialing 0501657887:
-- Executing StripMSD("SIP/3267915-7e9a", "1") in new stack
...this is the only thing I get on the console, stuck ?
dialing 501657887:
-- Executing StripMSD("SIP/3267915-7e9a", "1") in new stack
-- Executing StripMSD("SIP/3267915-4297", "1") in new stack
-- Executing Prefix("SIP/3267915-285a", "001051") in new stack
-- Prepended prefix, new extension is 001051501657887
-- Executing Busy("SIP/3267915-285a", "") in new stack
== Spawn extension (demo, 001051501657887, 2) exited non-zero on
'SIP/3267915-285a'
...the call fails
dialing 0010515016571887:
-- Executing Dial("SIP/3267915-43af", "Zap/g2/001051501657887") in new stack
-- Called g2/001051501657887
-- Zap/32-1 is making progress passing it to SIP/3267915-43af
-- Hungup 'Zap/32-1'
- the call gets through.
My question, how to manipulate with 05016576887 to obtain 001051657887 ? -
seems like very trivial but doesent want to work for me.
Thanks
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ring tone
Hi ! I have a small problem. When switching a call (pstn -> sip user), I get the sip phone ringing - ie. everything is OK, but I do not get a ringtone in the handset on the pstn side. Can anyone help me out in how to make * play tones ? My setup: E1 IP pstn -- Asterisk -- sip phone Regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] one way choppy sound problem !
Oh btw. I am using a P4, 500Mb RAM regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Monday, January 05, 2004 5:22 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] one way choppy sound problem ! Michael Van Donselaar wrote: >On Mon, 5 Jan 2004 13:29:06 +0100, "Dawid Mielnik" <[EMAIL PROTECTED]> >wrote: > > > >>Hi Again, >> >>Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and >>some others. I have tried different codecs - GSM, aLAW uLAW. They all give >>the same result. In the direction PSTN user ---> Softphone user the sound is >>crystal clear (also tried on a dial-up connection), in the other direction >>however the sound is a bit choppy. The chops occur at regular intervals of >>time, at about 1-2 seconds !? >> >> > >Are the PSTN interface and a network card sharing an interrupt? I had similar >problems with my X100P and a thunderlan dual ethernet card shring IRQs (also >would make one of the ethernet ports fails until reboot) > >Are you still using the P133? I tried using a P120, but it wouldn't do the >trick with GSM conversion. DIAX and iaxComm, since they use the iaxclient >library, need to use GSM. > > I have the same choppy sound problem on my server, my card is not sharing an interrupt and I am using G711 which is not hittng the P2 400 at all.. It seems there is a gremlin.. :) ___ 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] one way choppy sound problem !
Michael, The PSTN cards and the network card are not sharing an interrupt, the PSTN interface is sharing an interrupt with an audio controller and an smbus, the network card is not sharing an IRQ with anything though. Best regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Van Donselaar Sent: Monday, January 05, 2004 4:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] one way choppy sound problem ! On Mon, 5 Jan 2004 13:29:06 +0100, "Dawid Mielnik" <[EMAIL PROTECTED]> wrote: > >Hi Again, > >Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and >some others. I have tried different codecs - GSM, aLAW uLAW. They all give >the same result. In the direction PSTN user ---> Softphone user the sound is >crystal clear (also tried on a dial-up connection), in the other direction >however the sound is a bit choppy. The chops occur at regular intervals of >time, at about 1-2 seconds !? Are the PSTN interface and a network card sharing an interrupt? I had similar problems with my X100P and a thunderlan dual ethernet card shring IRQs (also would make one of the ethernet ports fails until reboot) Are you still using the P133? I tried using a P120, but it wouldn't do the trick with GSM conversion. DIAX and iaxComm, since they use the iaxclient library, need to use GSM. > >When analyzing *'s ethernet interface with tcpdump (raw tcpdump -i eth0) I >have noticed that the scrolling slows down during the times when chops occur >in the sound. > >I have tested things using different softphones and different internet >connections (user side) - always yelding the same result. In other words >this is probably a problem on asterisk, either the hardware (ehternet >interface/E100p) or a swoftware bug, incoming RTP buffering maybe ? > >Has anyone actually obtained a good quality sound in a similar setup ? > > Internet 2 x E1 > x-lite <---> Asterisk ---> PSTN > > >Any help appreciated ! > >Best regards, > >Dave > > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Nicolas >Gudino >Sent: Friday, January 02, 2004 6:35 PM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] one way choppy sound problem ! > > >I have a similar problem, with GS phones, X-Lite or Kphone. I tried all >the codecs with the same result. Choppy sound in the direction SIP-Phone >-> pstn, but crystal clear sound the other way around. The only >difference in my case is that I have two asterisks servers connected >together via IAX2, the PSTN call is received in one asterisk, while the >sip phones are in the other asterisk. Ex: > >pstn -> * --iax2--> * ->sip phone (GS, Xlite or Kphone) > >If I use an Xlite in the same asterisk as the pstn line, the sound is >perfect in both ways. But when I answer the call in the second asterisk, >the sound from the sip phone to pstn is choppy, with or without silence >detection, and the sound from pstn to sip phone is perfect. > >The asterisk server with the pstn line is an old pentium 133, maybe >thats the problem, I will try with a better machine and see how it goes. > > >On Fri, 2004-01-02 at 06:23, Dawid Mielnik wrote: >> Hi all, >> >> I have my asterisk setup as following: >> >> IP 2 x E1 >> x-lite <---> Asterisk ---> PSTN >> >> >> When I place a call from x-lite to PSTN, the quality of the sound in the >> direction x-lite -> PSTN is very bad. That is, the voice of the x-lite >user, >> heard by the PSTN user is choppy and makes communication not very >pleasant. >> The sound is choppy as if bits of data were lost. The strange thing is >that >> the x-lite user hears the PSTN user fine ! >> >> In x-lite, I have swithed off sience detection (transmit silence - yes), >> this has improved the sound quality but did not eliminated the problem. I >> have fed a countinious sound into the microphone and still got chops in >the >> sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get >the >> same problem with all of them. Maybe the problem lies somewhere in audio >> buffering settings on x-lite ? >> >> Has anyone ever had this sort of problem and managed to deal with it ? I >> would greatly appreciate your help ! >> >> Best regards, >> >> Dave >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >___ >A
RE: [Asterisk-Users] one way choppy sound problem !
Hi Again, Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and some others. I have tried different codecs - GSM, aLAW uLAW. They all give the same result. In the direction PSTN user ---> Softphone user the sound is crystal clear (also tried on a dial-up connection), in the other direction however the sound is a bit choppy. The chops occur at regular intervals of time, at about 1-2 seconds !? When analyzing *'s ethernet interface with tcpdump (raw tcpdump -i eth0) I have noticed that the scrolling slows down during the times when chops occur in the sound. I have tested things using different softphones and different internet connections (user side) - always yelding the same result. In other words this is probably a problem on asterisk, either the hardware (ehternet interface/E100p) or a swoftware bug, incoming RTP buffering maybe ? Has anyone actually obtained a good quality sound in a similar setup ? Internet 2 x E1 x-lite <---> Asterisk ---> PSTN Any help appreciated ! Best regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nicolas Gudino Sent: Friday, January 02, 2004 6:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] one way choppy sound problem ! I have a similar problem, with GS phones, X-Lite or Kphone. I tried all the codecs with the same result. Choppy sound in the direction SIP-Phone -> pstn, but crystal clear sound the other way around. The only difference in my case is that I have two asterisks servers connected together via IAX2, the PSTN call is received in one asterisk, while the sip phones are in the other asterisk. Ex: pstn -> * --iax2--> * ->sip phone (GS, Xlite or Kphone) If I use an Xlite in the same asterisk as the pstn line, the sound is perfect in both ways. But when I answer the call in the second asterisk, the sound from the sip phone to pstn is choppy, with or without silence detection, and the sound from pstn to sip phone is perfect. The asterisk server with the pstn line is an old pentium 133, maybe thats the problem, I will try with a better machine and see how it goes. On Fri, 2004-01-02 at 06:23, Dawid Mielnik wrote: > Hi all, > > I have my asterisk setup as following: > > IP 2 x E1 > x-lite <---> Asterisk ---> PSTN > > > When I place a call from x-lite to PSTN, the quality of the sound in the > direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, > heard by the PSTN user is choppy and makes communication not very pleasant. > The sound is choppy as if bits of data were lost. The strange thing is that > the x-lite user hears the PSTN user fine ! > > In x-lite, I have swithed off sience detection (transmit silence - yes), > this has improved the sound quality but did not eliminated the problem. I > have fed a countinious sound into the microphone and still got chops in the > sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get the > same problem with all of them. Maybe the problem lies somewhere in audio > buffering settings on x-lite ? > > Has anyone ever had this sort of problem and managed to deal with it ? I > would greatly appreciate your help ! > > Best regards, > > Dave > > > ___ > 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] one way choppy sound problem !
Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <---> Asterisk ---> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data were lost. The strange thing is that the x-lite user hears the PSTN user fine ! In x-lite, I have swithed off sience detection (transmit silence - yes), this has improved the sound quality but did not eliminated the problem. I have fed a countinious sound into the microphone and still got chops in the sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get the same problem with all of them. Maybe the problem lies somewhere in audio buffering settings on x-lite ? Has anyone ever had this sort of problem and managed to deal with it ? I would greatly appreciate your help ! Best regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E100P configuration
Scott,
Thanks a lot ! this is exaclty what I wanted. Both my E1's came up without
problems.
Best regards,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel
Sent: Tuesday, December 30, 2003 3:42 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] E100P configuration
Hi-
Not sure that I understand your question about grouping, but here is what I
use for 2 E1's connected to a private switch (in addition to the other
parameters) Note that I use the pri_cpe ("customer premise equipment")
setting. The defined channels act as one big group of 60 channels, if
that's what you mean. Your telephone company will define the call
distribution for your incoming calls:
pridialplan=unknown
context=incoming
usecallerid=yes
group=1
signalling=pri_cpe
channel => 1-15,17-31
channel => 32-46,48-62
Regards,
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email: scott "at" evtmedia.com
URL:www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dawid Mielnik
Sent: Tuesday, December 30, 2003 11:50 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] E100P configuration
Hi !
I am trying to configure two E100P cards, but I am a bit confused with
zapta.conf in what I am trying to achieve.
The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines
will be used for incoming calls as well as outgoing calls.
My problem now is what to put in zapta.conf, I would like to group all
channels from both cards together (if that's possible). Does this make sense
?
context=default
switchtype=euroisdn
signalling=pri_net
;pridialplan=national
overlapdial=yes
group=1
channel => 1-15,17-31,32-46,48-62
what does "channel include" ? all the channels d and b ?
Thanks for your help.
Dave
___
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] E100P configuration
Hi ! I am trying to configure two E100P cards, but I am a bit confused with zapta.conf in what I am trying to achieve. The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines will be used for incoming calls as well as outgoing calls. My problem now is what to put in zapta.conf, I would like to group all channels from both cards together (if that's possible). Does this make sense ? context=default switchtype=euroisdn signalling=pri_net ;pridialplan=national overlapdial=yes group=1 channel => 1-15,17-31,32-46,48-62 what does "channel include" ? all the channels d and b ? Thanks for your help. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem)
Hi all, Clipcomm - looks interesting, you get NAT/Ethernet/Analog line (to SIP) D-link DVG-1120M/H/S - this is also on the lines of what I'm looking for, lets you connect standard analog phone directly to it, has NAT and an Ethernet port - anyone ever tried this with * ? What I am exactly trying to find is an easy to setup/'elegent'/cheap (I know I might be asking for too much) solution to provide SIP connectivity (while maintaining Internet connectivity from PC) considering two scenarios: 1. Cable ISP user with single IP over DHCP (the user already has a network card in his PC): this is quite easy, the D-link DVG-1120 will do the trick - PC goes to to the Ethernet plug, analogue phone goes directly to the RJ-11 plug (nice) 2. ADSL (PPPoA) user with a USB connected modem, single IP over DHCP (no network card in his PC): this is more tricky - Intertex IX66+PF may do the trick although is quite expensive, I've also found "SMC Barricade ADSL Modem Router" with similar features at a lesser price. This still calls for a ATA286 analog phone adaptor if the guy is to use his normal telephone. The ideal case would be a a small cheab box with an Ethernet/ADSL WAN plugs on one end and an Ethernet/USB/Analog (from SIP) plugs on the other end ;-) Thanks, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Friday, December 19, 2003 1:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem) Hi! > I was wondering whether any of you have experience/info on Cable and/or ADSL > modems that would come together with a SIP phone adaptor. What I am > interested in is something that would plug directly into you ISP's cable (be > it ethernet or adsl/phoneline), would combine a modem/router/nat such that > on the other you could simply plug in your RJ-45 cable for your PC and a > RJ-11 cable for the telephone. Something that would combine the > functionality of a (adsl modem+) router and a SIP telephone adaptor in one > box. This is not exactly what you are asking for, but its getting close: http://www.voip-info.org/tiki-index.php?page=VOIP+Phones Look at the "Clipcomm" devices (Korea). I'd be interested in any reports & experiences. Next to that the newer Grandstream firmware now supports PPPoE (but now routing or port forwarding). Not sure what the CISCO devices offer in this respect. I wouldn't want to have ADSL modem & router coupled in one device, that'll make your router useless if you move to cable modem etc. So better not integrate too many task into one piece of hardware. Cheers, Philipp ___ 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] nat router + sip phone adaptor (+adsl modem)
Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for the telephone. Something that would combine the functionality of a (adsl modem+) router and a SIP telephone adaptor in one box. I would appreciate any info that you might have on this. regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk - scalable ?
Hi all, How scalable is asterisk ? I am considering using asterisk as a VoIP platform/gateway between Internet and PSTN (switches) to offer services to home customers. What goes along with it is eventually a lot of users - upto thousands probably. Is load balancing possible with multiple asterisk boxes ? Does anyone have any sort of info/experience with such projects ? How would asterisk cope with such load. Asterisk offers conferencing (meetme). The conference rooms as such dont have any user verification, setting up passwords etc. Has anyone implemented anything like that ? regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail as an attachement
Hi again, sorry for the spam but, I have tried connecting * to a different internet connection (isp) and the mail attachments work ok now - dont know why. Best regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dawid Mielnik Sent: Tuesday, December 16, 2003 10:30 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail as an attachement It works with "attachment=no" ! [EMAIL PROTECTED] is my masqueraded (by sendmail) [EMAIL PROTECTED] Any ideas ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Thompson Sent: Monday, December 15, 2003 11:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail as an attachement - Original Message ----- From: "Dawid Mielnik" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, December 15, 2003 4:16 PM Subject: [Asterisk-Users] voicemail as an attachement > > Hi, > > I can not send voicemails as an attachement. When setting the "attach=yes" > option in voicemail.conf the mails get rejected from the mail server: > >- Transcript of session follows - > 451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection timed > out > with higgs.elka.pw.edu.pl. > 451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection > timed > out with elektron.elka.pw.edu.pl. > ... while talking to mion.elka.pw.edu.pl.: > >>> MAIL From:<[EMAIL PROTECTED]> SIZE=21382 > <<< 553 5.7.1 For MAIL FROM address <[EMAIL PROTECTED]> access is denied by the > pol > icy analysis functions. > 501 5.6.0 Data format error > > Can anyone help me out with this ? Why is there a data format error. You're not reading enough of the error message... The mail is failing because it doesn't like your sending email address. If it works with attach=no, I'd be suprised. Do the notifications come from the same email address([EMAIL PROTECTED]) when the attachments are disabled? - Andrew Thompson http://aktzero.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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail as an attachement
It works with "attachment=no" ! [EMAIL PROTECTED] is my masqueraded (by sendmail) [EMAIL PROTECTED] Any ideas ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Thompson Sent: Monday, December 15, 2003 11:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail as an attachement - Original Message ----- From: "Dawid Mielnik" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, December 15, 2003 4:16 PM Subject: [Asterisk-Users] voicemail as an attachement > > Hi, > > I can not send voicemails as an attachement. When setting the "attach=yes" > option in voicemail.conf the mails get rejected from the mail server: > >- Transcript of session follows - > 451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection timed > out > with higgs.elka.pw.edu.pl. > 451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection > timed > out with elektron.elka.pw.edu.pl. > ... while talking to mion.elka.pw.edu.pl.: > >>> MAIL From:<[EMAIL PROTECTED]> SIZE=21382 > <<< 553 5.7.1 For MAIL FROM address <[EMAIL PROTECTED]> access is denied by the > pol > icy analysis functions. > 501 5.6.0 Data format error > > Can anyone help me out with this ? Why is there a data format error. You're not reading enough of the error message... The mail is failing because it doesn't like your sending email address. If it works with attach=no, I'd be suprised. Do the notifications come from the same email address([EMAIL PROTECTED]) when the attachments are disabled? - Andrew Thompson http://aktzero.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] voicemail as an attachement
I have tried sending mail notifications to different mail servers. Again, each time I attache the voicemail - they fail. regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher Sent: Monday, December 15, 2003 11:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail as an attachement On Monday 15 December 2003 15:16, Dawid Mielnik wrote: > I can not send voicemails as an attachement. When setting the > "attach=yes" option in voicemail.conf the mails get rejected from > the mail server: This is not a problem with Asterisk. Have a talk with your mail admin about this destination mail server problem. -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] voicemail as an attachement
Hi, I can not send voicemails as an attachement. When setting the "attach=yes" option in voicemail.conf the mails get rejected from the mail server: - Transcript of session follows - 451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection timed out with higgs.elka.pw.edu.pl. 451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection timed out with elektron.elka.pw.edu.pl. ... while talking to mion.elka.pw.edu.pl.: >>> MAIL From:<[EMAIL PROTECTED]> SIZE=21382 <<< 553 5.7.1 For MAIL FROM address <[EMAIL PROTECTED]> access is denied by the pol icy analysis functions. 501 5.6.0 Data format error Can anyone help me out with this ? Why is there a data format error. thanks. best regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail.conf email notification
Jeremy, So the only way is to change the voicemailcode and recompile ? the flexible - uncomment these lines does will never work ? thnx. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Monday, December 15, 2003 5:33 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail.conf email notification Dawid Mielnik wrote: >Hello, > >I am trying to change the email body and the from string sent in the >voicemail notifocation mail. > >I have changed the entries in the voicemail.conf but I still receive the >standard email template from "Asterisk PBX" (instead of my from) and [PBX]: >in the subject. Can anyone help me out in customizing the email notification >? > > cd /path/to/asterisk vi apps/app_voicemail2.c Jeremy McNamara ___ 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] voicemail.conf email notification
Hello, I am trying to change the email body and the from string sent in the voicemail notifocation mail. I have changed the entries in the voicemail.conf but I still receive the standard email template from "Asterisk PBX" (instead of my from) and [PBX]: in the subject. Can anyone help me out in customizing the email notification ? Thanks in advace. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
