Someone correct me if I'm wrong, but is the typical setup for Nathelper and
Portaone RTP proxy work on seperate servers with dual network cards? then
talking to another SER box doing registrations and call routing? and the
Portaone RTP taking the voice packets?
Thanks,
- Darren
Wangji,
You need a packet sniffer, like www.ethereal.com, and probably follow more
carefully the sample script:
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/nathelper.cfg?…
Good luck!
Jaime
From: jimmy way <jimway71(a)yahoo.com> on 01/12/2003 10:04
To: Jaime GIL/EN/HTLUK@HTLUK
cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] nathelper question
Jaime,
I have do it, buf still fail.
By the way, I really want to see the SIP packets SER
sended, recieved and the packets after changed.
maybe can see what wrong with it. But I can't do it.
Wangji
--- jaime.gil(a)orange.co.uk wrote:
>
> Wangji,
>
> Not sure, but normally you need to force_rport and
> fix_nated_contact at
> registration, before you save to DB.
>
> Jaime
>
>
>
>
*******************************************************************************
Important.
Confidentiality: This communication is intended for the above-named person and
may be confidential and/or legally privileged. Any opinions expressed in this
communication are not necessarily those of the company. If it has come to you
in error you must take no action based on it, nor must you copy or show it to
anyone; please delete/destroy and inform the sender immediately.
Monitoring/Viruses
Orange may monitor all incoming and outgoing emails in line with current
legislation. Although we have taken steps to ensure that this email and
attachments are free from any virus, we advise that in keeping with good
computing practice the recipient should ensure they are actually virus free.
Orange PCS Limited is a subsidiary of Orange SA and is registered in England No
2178917, with its address at St James Court, Great Park Road, Almondsbury Park,
Bradley Stoke, Bristol BS32 4QJ.
*******************************************************************************
HI,
My problem is unknown authenfication with radius
digest authorization work properly with x-lite
but doesnt work with kerio softphone and dlink hardware gateway
in debug messages i have
where is problem ?
7(18572) SIP Request:
7(18572) method: <REGISTER>
7(18572) uri: <sip:195.138.96.157>
7(18572) version: <SIP/2.0>
7(18572) parse_headers: flags=1
7(18572) end of header reached, state=5
7(18572) parse_headers: Via found, flags=1
7(18572) parse_headers: this is the first via
7(18572) After parse_msg...
7(18572) preparing to run routing scripts...
7(18572) DEBUG : is_maxfwd_present: searching for max_forwards header
7(18572) parse_headers: flags=128
7(18572) DEBUG: is_maxfwd_present: value = 70
7(18572) parse_headers: flags=8
7(18572) DEBUG: add_param: tag=62db317e3f8f6cd7
7(18572) end of header reached, state=29
7(18572) parse_headers: flags=256
7(18572) end of header reached, state=9
7(18572) DEBUG: get_hdr_field: <To> [36]; uri=[sip:1002@evgeniy.riscom.net]
7(18572) DEBUG: to body [1002 <sip:1002@evgeniy.riscom.net>]
7(18572) get_hdr_field: cseq <CSeq>: <276> <REGISTER>
7(18572) DEBUG: get_hdr_body : content_length=0
7(18572) found end of header
7(18572) find_first_route(): No Route headers found
7(18572) loose_route(): There is no Route HF
7(18572) receive_msg: cleaning up
hi,
I have set the debug=9 in ser.cfg and build with mode=debug, but i still can not get much info from /var/log/messages.
Can anybody give a hint ?
Many thanks,
hong
--------------------------
debug=9 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
#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"
--------------------------------
Dec 2 18:32:09 localhost ./ser[15797]: INFO: udp_init: SO_RCVBUF is initially 6
5535
Dec 2 18:32:09 localhost ./ser[15797]: INFO: udp_init: SO_RCVBUF is finally 131
070
Dec 2 18:32:09 localhost ./ser[15797]: INFO: udp_init: SO_RCVBUF is initially 6
5535
Dec 2 18:32:09 localhost ./ser[15797]: INFO: udp_init: SO_RCVBUF is finally 131
070
Dec 2 18:32:09 localhost ./ser[15806]: INFO: fifo process starting: 15806
Dec 2 18:32:09 localhost ./ser[15806]: SER: open_uac_fifo: fifo server up at /t
mp/ser_fifo...
Sorry I attached the wrong config file in my previous email. Correct one attached -
Regards,
Dhiraj
-----Original Message-----
From: dhiraj.2.bhuyan(a)bt.com [mailto:dhiraj.2.bhuyan@bt.com]
Sent: 02 December 2003 10:48
To: serusers(a)lists.iptel.org
Subject: [Serusers] SIP URL with IP Address Problem
Hello everyone,
I am facing the following problem -
eniac.alien.bt.co.uk = 132.146.196.91
A Grandstream BudgeTone 100 phone is registered to the SER registrar and proxy running on eniac.alien.bt.co.uk as dhiraj(a)eniac.alien.bt.co.uk
When I try
serctl ping sip:dhiraj@eniac.alien.bt.co.uk
I get back a "200 OK"
But when I do
serctl ping sip:dhiraj@132.146.196.91
I get back a "404".
How to overcome this? I am using SER from the CVS. My ser.cfg is attached.
Thanks,
Dhiraj Bhuyan
Network Security Specialist,
BT Exact
Tel: +44 1473 643932
Mob: +44 7962 012145
Email: dhiraj.2.bhuyan(a)bt.com
FYI,
The Grandstream phones work great with SER provided the numeric aliases
can be used to reach endpoints. The Grandstream can only dial numeric
numbers and not generic sip URI/URLs. It would be great if there was a
lookup available underneath the http://www.iptel.org/user/index.php
(MyAccount link)->(phone book tab)->(find user link) to find numeric
aliases to reach iptel.org users, provided you got the correct non
numeric SIP URIs of iptel.org users.
Also, the peering prefix for an iptel.org user to reach and
fwd.pulver.com user apears to be in error at
http://www.fwd.pulver.com/index.php?section_id=78&PHPSESSID=06a451f4079f56c…,
a prefix of 21111 works great and **393 does not work.
Bye,
Scott Holben
sip iptel.org URIs: skaht AT iptel.org or 90931 AT iptel.org
Just found an IETF draft on how to implement PBX features in SIP
http://www.voip-info.org/tiki-index.php?page=SIP+PBX+functions
Maybe we can try to implement these in SER and document these functions
on the Wiki?
There's some of these in the source examples and in the howto's, but it
might be good to document functions one by one.
# Call Hold
# Music on Hold
# Unattended Transfer
# Consultation Hold
# Unconditional Call Forwarding
# Attended Transfer
# No Answer Call Forwarding
# Busy Call Forwarding
# Single-Line Extension
# 3-way Call
# Incoming Call Screening
# Find-Me
# Call Pickup
# Call Park
# Outgoing Call Screening
# Automatic Redial
Anyway, it's good reading material!
/Olle
Hello,
After having worked with ser for a few days, I have SER running with
authentication, accounting , voicemail and conferencing, which is
working great (Thanks for this great product). However, I have a problem
connecting Cisco 7960 phones.
Ser server: colo.foo.bar.
Domain/authentication realm/alias: foo.net.
In my grandstream phones I can configure a different domain then the sip
proxy. In the cisco phones, I can only configure the sip proxy. The
cisco's are registering with the username(a)colo.foo.bar and not with
username(a)foo.net. The phones will not authenticate, because there only
exists a username(a)foo.net in the database.
What is the best to solve this problem. I can think of the following:
- match the uri and do: rewritehost("foo.net")
- make a srv entry for foo.net pointing to colo.foo.bar and set the
proxy in the cisco phone to foo.net
- configure multidomain support and put in both domains.
I'm not sure what the best solution is, does anyone have experience with
this issue?
Furthermore, I have a question of which I could not find an answer for
in the archives or google. Is it possible to do call pick-up with ser?
My first thought would be no (and use a timeout with forwarding),
because Ser is not call statefull (and has no b2bua). But maybe I'm wrong.
Kind regards and thanks for reading,
Geert Nijpels
Hello I have tryed installing SER on two Debian(I386) servers, on both
mashines i get the following error in the syslog, while running "serctl
moni"
Nov 30 13:38:45 *host* /usr/sbin/ser[24949]: ERROR: open_reply_pipe:
open error (/tmp/ser_receiver_25028): Permission denied
Nov 30 13:38:45 *host* /usr/sbin/ser[24949]: ERROR: fifo_reply: no reply
pipe /tmp/ser_receiver_25028
what am I missing.
thanks for now
Thrane
Have you tried to use DO= ip-address of ser-server instead of ati.com?
I use authentication so I add AUTH-USERNAME=38140299 and
AUTH-PASSWD=xxxxxxxx to the "create sip" command.
Here is one of my configs:
S SWITCH AGET=NORMAL
S SWITCH PORT=WAN IFLT=ON
S SWITCH PORT=LAN1 IFLT=ON
S SWITCH PORT=LAN2 IFLT=ON
S SWITCH PORT=LAN3 IFLT=ON
A VLAN=1 PORT=WAN FRAME=TAGGED
C VLAN=iptnett VID=5
A VLAN=5 PORT=VOIP
A VLAN=5 PORT=WAN FRAME=TAGGED
C VLAN=tvnett VID=10
A VLAN=10 PORT=LAN1,LAN3
A VLAN=10 PORT=WAN FRAME=TAGGED
SET IP INTERFACE=eth0 CONFIGURATION=DHCP
ENABLE TELNET
ENABLE NTP
SET NTP UTCOFFSET=1
ADD NTP SERVER=10.100.10.31 DEFAULT
SET PASSWORD=CIEEHOBB@HHILCHBNADNHA@NGL@B@HIF
ENABLE SIP
CREATE SIP PORT=0 PHNO=38140299 AUTH-USERNAME=38140299
AUTH-PASSWD=xxxxxxxx DO=ipt-server1.ivisjon.no LS=ipt-server1.ivisjon.no
PS=ipt-server1.ivisjon.no CAP=PCMU;PCMA TOS=5
SET PHONE PORT=0 DCALL=VOIP VAD=OFF
I have "ipt-server1.ivisjon.no" in my dns.
Tor.
andyv(a)sympatico.ca wrote:
> The RG's are 213
>
> Here are the *.cfg files
>
> #1
> SET IP INTERFACE=eth0 IPADDRESS=192.168.20.3 MASK=255.255.255.0
> GATEWAY=192.168.20.1
> SET LOADER SERVER=192.168.20.10
> ENABLE SIP
> CREATE SIP PORT=0
> PHNO=1001 DO=ati.com LS=192.168.20.4 PS=192.168.20.4 CAP=PCMU;PCMA;G723 TOS=0
> SET
> PHONE PORT=0 DCALL=VOIP
>
> #2
> SET IP INTERFACE=eth0 IPADDRESS=192.168.20.2 MASK=255.255.255.0
> GATEWAY=192.168.20.1
> SET LOADER SERVER=192.168.20.4
> SET DOMAIN=ati.com
> ENABLE
> SIP
> CREATE SIP PORT=0 PHNO=1000 DO=ati.com LS=192.168.20.4 PS=192.168.20.4
> CAP=PCMU;PCMA;G723 TOS=0
> ENABLE TELNET
> SET PHONE PORT=0 DCALL=VOIP
> SET
> PHONE PORT=1 DCALL=VOIP
>
>
>
>>From: Tor Setane <tor.setane(a)ella.no>
>>Date: Mon, 01 Dec 2003 13:19:00 +0100
>>To: serusers(a)lists.iptel.org
>>Subject: [Serusers] Re: Residential Gateways
>>
>>What kind of gateways are you using? 213 or 613?
>>I'm using both without any problems. Please post your rg config..
>>
>>Tor.
>>
>>
>>>Date: Sun, 30 Nov 2003 19:59:37 -0500
>>>From: "Andy Vander Woude" <andyv(a)sympatico.ca>
>>>Subject: [Serusers] Residential Gateways
>>>To: <serusers(a)lists.iptel.org>
>>>Message-ID: <000001c3b7a6$68560d70$8adaacce@generator>
>>>Content-Type: text/plain; charset="us-ascii"
>>>
>>>
>>>Running Redhat V9 and have installed the SIP Express Router (ser)
>>>packages. Have two Allied Telesyn Residential Gateways (192.168.20.2 and
>>>192.168.20.3) and laptop (192.168.20.4) running ser connecting to a
>>>managed switch IP address (192.168.20.1 (gateway))
>>>The RG's are configured to access the Proxy server, & Domain server at
>>>192.168.20.4 and the Gateway as 192.168.20.1 The Domain is called
>>>ati.com.
>>>
>>>When connected all together I get a dial tone on the phones, however
>>>when I dial the # configured on the RG's I do not get a ring and they do
>>>not register with the ser server. When I type in the "ser start" command
>>>I see the alias entries as "127.0.0.1 localhost localdomain localhost"
>>>and "192.168.20.4 localhost.ati.com localhost"
>>>
>>>The entries in the ser.cfg relating to the localdomain have been changed
>>>to ati.com
>>>I have added the 192.168.20.4 to the hosts.cfg relating to the ati.com
>>>domain.
>>>
>>>Any idea as to why this configuration is not working. This is a lab
>>>situation with no outside connections.
>>>Thank you.
>>>
>>>
>>
>>
>>---
>>[This E-mail was scanned for viruses by Declude Virus]
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
> ---
> [This E-mail was scanned for viruses by Declude Virus]
>
>
>
---
[This E-mail was scanned for viruses by Declude Virus]
