Hi Jiri,
it works! Thank you very much!
The voice-port block is the key:
-------------------------------
voice-port 1/0/0
compand-type a-law
cptone DE
bearer-cap Speech
------------------------------
yang
----- Original Message -----
From: "Jiri Kuthan" <jiri(a)iptel.org>
To: "Yang Xiang" <yang.xiang(a)iitb.fraunhofer.de>
Sent: Friday, April 25, 2003 5:21 PM
Subject: Re: [Serusers] problem with cisco 2600 to pstn
> try to look at our config at
http://www.iptel.org/~jiri/etc/cisco/ios_2003.txt
> if it helps you. I think we had the same problem, changed some settings
and it
> worked then. I unfornantely don't remember what it was.
>
> -Jiri
>
> At 04:54 PM 4/25/2003, you wrote:
> >Hi all,
> >
> >I am expericing problem with the cisco 2600, which should function as the
> >sip2pstn gateway. If I try to complete a call from a sip phone to pstn,
the
> >router says:
> >------------------------------------------------------------------------
> > 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03
> >00:15:49: Bearer Capability i = 0x8090A2
> > ^^^^^^
> >00:15:49: Channel ID i = 0x83
> >00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may
> >have in-band info
> >00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown,
> >Type:Unknown
> >00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private,
> >Type:Subscriber(local)
> >00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83
> >00:15:49: Channel ID i = 0x89
> >00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate
now
> >available
> >00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83
> >00:15:49: Cause i = 0x80C1 - Bearer capability not implemented
> >
> >^^^^^^^^^^^^^^^^^^^^^^^^^^
> >00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate
now
> >available
>
>---------------------------------------------------------------------------
-
> >------------
> >
> >Please notice the line "Bearer Capability i = 0x8090A2", the digits "80"
> >mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and
"A2"
> >is for G.711 u-law.
> >
> >So if I call the router from a normal telephone, the debugging looks as
> >follows:
>
>---------------------------------------------------------------------------
-
> >----
> >01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01
> >01:01:58: Bearer Capability i = 0x8090A3
> > ^^^
> >01:01:58: Channel ID i = 0x89
> >01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN,
> >Type:Unknown
> >01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN,
> >Type:Subscriber(local)
> >01:01:58: High Layer Compat i = 0x9181
> >01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at
64
> >Kb/s
>
>---------------------------------------------------------------------------
-
> >---------
> >
> >whereat the bearer capability is "0x8090A3". It means that the
ISDN-switch
> >of German Telekom uses G.711 a-law.
> >
> >I am afraid that is the reason why the sip-call doesn't go through. But I
> >can't find any way to configure this.
> >
> >Has anybody in this mailinglist the same experience?
> >
> >Any hints would be very appreciated.
> >
> >Thanks
> >
> >yang
> >
> >
> >
> >
> >
> >
> >_______________________________________________
> >Serusers mailing list
> >serusers(a)lists.iptel.org
> >http://lists.iptel.org/mailman/listinfo/serusers
>
> --
> Jiri Kuthan http://iptel.org/~jiri/
>
>
Hi , Xten Networks, Inc. (www.xten.com) is pleased to announce it is making a contribution to the SER project. The Xten SER Team is comprised of 2 senior engineers and 1 project
manager who are committed full-time to the development of SERAdmin.SERAdmin is a GUI interface between SIP Express Router (SER) and the SER administrator. Project location (http://developer.berlios.de/projects/seradmin/) SERAdmin provides control over many SER tasks such as: start, stop,
pause, re-start, monitor, add user, edit user, etc. SERAdmin has an
intuitive look and feel.SERAdmin is open source, is being developed to benefit all SER
administrators, and the feature set of SERAdmin will be determined by
the iptel.org SER users' group.So please communicate with the Xten SER Team, post your comments in the public forums, and make use of the Xten SER Team as they are working for the SER community. About Xten (www.xten.com) Xten Networks, Inc. is a leading provider of high-quality SIP Voice
over Internet Protocol (VoIP) software. Xten provides IP Telephony products directly to end users, the Enterprise market, Next-Gen Service
Providers (ITSPs & Tier 2), Wireless Internet Service Providers (WISPs),
Telephone Companies (TELCOs), and Original Equipment Manufacturers (OEMs).
Regards,Xten Team
---------------------------------
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Hi all,
I am expericing problem with the cisco 2600, which should function as the
sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the
router says:
------------------------------------------------------------------------
00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03
00:15:49: Bearer Capability i = 0x8090A2
^^^^^^
00:15:49: Channel ID i = 0x83
00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may
have in-band info
00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown,
Type:Unknown
00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private,
Type:Subscriber(local)
00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83
00:15:49: Channel ID i = 0x89
00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now
available
00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83
00:15:49: Cause i = 0x80C1 - Bearer capability not implemented
^^^^^^^^^^^^^^^^^^^^^^^^^^
00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now
available
----------------------------------------------------------------------------
------------
Please notice the line "Bearer Capability i = 0x8090A2", the digits "80"
mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2"
is for G.711 u-law.
So if I call the router from a normal telephone, the debugging looks as
follows:
----------------------------------------------------------------------------
----
01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01
01:01:58: Bearer Capability i = 0x8090A3
^^^
01:01:58: Channel ID i = 0x89
01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN,
Type:Unknown
01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN,
Type:Subscriber(local)
01:01:58: High Layer Compat i = 0x9181
01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64
Kb/s
----------------------------------------------------------------------------
---------
whereat the bearer capability is "0x8090A3". It means that the ISDN-switch
of German Telekom uses G.711 a-law.
I am afraid that is the reason why the sip-call doesn't go through. But I
can't find any way to configure this.
Has anybody in this mailinglist the same experience?
Any hints would be very appreciated.
Thanks
yang
hi!
is serweb intended to just send IM one-way? i configured everything and
noticed that i can only send messages from a web-client (via serweb) to
a sip softphone, but not the other way around.
i noticed that this is due to the fact that when a user logs on to
serweb, it does not put anything in the location table in ser db. so if
a message was sent intended for that user, the sender receives 404 not
found (due to failure in lookup()).
what i am thinking as a workaround is to manipulate the php files in
serweb to insert the user into the 'location' table via fifo, the method
though crude might just work...
my problem is, before i could try to do the workaround is whether serweb
would accept the message if it indeed received one on port 5060? would
it? or should the proxy send the message through another port like 80?
I'm not aware of an affordable stress tools which covers both media
and signaling.
-Jiri
At 03:50 PM 4/24/2003, Alejandro Olchik wrote:
>I would like to start multiple simultaneous calls
>(let's say around 30), and keep calls active to
>test concurrency.
>
>Any suggestion how can I do that without requiring
>expensive software for that?
>
>I have enough ports to terminate the calls using
>G729 but I don't know what to use for originate them.
>
>Regards,
>Alejandro
>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Hi all,
I put following config in the main routing-block for forwarding INVITE-msg
to pstn-gatway:
...
route{
...
if (!t_relay()) {
sl_reply_error();
};
# forward message to PSTN Gateway
if (uri=~"^sip:[0-9]*@"){
log(5, "Forward to pstn \n");
forward("10.20.0.6");
break;
};
}
However, SER doesn't do it and sends a reply "404 not found" back instead.
Is it the correct position to put the function "forward()" there?
By the way, ser also doesn't write log message to syslog. Following is my
syslog.conf, is anything wrong in the syslog.conf?
#ident "@(#)syslog.conf 1.5 98/12/14 SMI" /* SunOS 5.0 */
#
# Copyright (c) 1991-1998 by Sun Microsystems, Inc.
# All rights reserved.
#
# syslog configuration file.
#
# This file is processed by m4 so be careful to quote (`') names
# that match m4 reserved words. Also, within ifdef's, arguments
# containing commas must be quoted.
#
*.debug;*.notice;*.info;*.crit;*.alert /var/log/debug
*.err;kern.notice;auth.notice /dev/sysmsg
*.err;kern.debug;daemon.notice;mail.crit /var/adm/messages
*.alert;kern.err;daemon.err operator
*.alert root
*.emerg *
# if a non-loghost machine chooses to have authentication messages
# sent to the loghost machine, un-comment out the following line:
#auth.notice ifdef(`LOGHOST', /var/log/authlog, @loghost)
mail.debug ifdef(`LOGHOST', /var/log/syslog, @loghost)
#
# non-loghost machines will use the following lines to cause "user"
# log messages to be logged locally.
#
user.err /dev/sysmsg
user.err /var/adm/messages
user.alert `root, operator'
user.emerg *
auth.info /var/adm/messages
local6.debug /var/adm/imapd.log
auth.debug /var/adm/auth.log
Thanks,
Yang
I would like to start multiple simultaneous calls
(let's say around 30), and keep calls active to
test concurrency.
Any suggestion how can I do that without requiring
expensive software for that?
I have enough ports to terminate the calls using
G729 but I don't know what to use for originate them.
Regards,
Alejandro
