Hello,
I prohibited posting from non-members to the list again (such mails are
held for review) due to large number of bogus messages received during
last couple of hours. It will be enabled again later.
I would like to encourage people who are using MS Outlook or similar
bullshit from microsoft to try a different mail client. Outlook is evil
and microsoft is doing nothing about that.
During the last couple of hours I received 400 reports that my emails or
emails from the mailing lists contain a virus or are undeliverable.
Please stop this madness and stop using outlook.
Jan.
PS: It's curious that Bill Gates gave a speech yesterday in Prague
regarding network security and how seriously are they taking it...
Never tried it myself - but I think it should work the following way. If the Router (NAT) has DMZ support, you can put SER on a machine in the Router's DMZ interface. This will make the SER appear with a globally routable address (that of the gateway or something else if you have multiple IP address support). So when UA3 registers itself with SER (configured with NAT support), the nathelper module will detect that UA3 is behind a NAT gateway and will enable rtp-relaying. So the trick out here is to make the SER appear to be on the Internet - rather than on the private network. Note that the router should allow communication between NATted devices and machines on the DMZ. Also note that by putting the SER on the internet, you are making it open to attacks. Add proper firewall rules.
Let me know if it works :)
Dhiraj Bhuyan
Network Security Specialist,
BT Exact Business Assurance Solutions
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Edson Gellert Schubert
Sent: 29 January 2004 15:19
To: Adrian Georgescu
Cc: Lista SER - IPTEL
Subject: Re: [Serusers] SER + Proxy + NAT
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
I thing that I didn't put myself very clear... Let me try an ASCII-Diagram... ;)
UA1 ---- public IP ---- Internet ------ ADSL/GW ------ UA2
(Win Mes) (dialup) | | (IPTables) (Win Mes)
| |
/-----/ \-----------\
| |
Router(NAT) ADSL/Win (Windows)
| |
UA3 ------------+ |
(Win Mes) | UA4 (Win Mes)
SER
What I'm looking for is a Proxy to put in the ROUTER-Machine (could be a Linux/IPTables, FreeBSD, etc):
I undestand what Jan explain, about the complications, and that's why I was asking about an "inteligent" Proxy to handle SIP traffic.
Suppose that UA3 wants to talk with UA2 (were the ADSL/GW should have SIProxd installed). The communications flow would be UA3-SER-Router-Internet-ADSL/GW-UA2. Ok, in the Router appears the first challenge (how to transverse the NAT, keeping track from the flow?). Here comes the SIP-Proxy in action. It recieves the packet from SER, make desired changes and forward it through "Internet" to ADSL/GW. There, the SIProxyd recieves the packet, apply the related changes and forward it to UA2. Great. Is what we want.
The reverse, that is, when UA2 (or UA1, or UA4) wants to talk with UA3 becomes the great challenge. How should the Proxy, in Router, knows where to send the packets that arrive from Internet? To SER? Directly to UA3? It's hard to make the decision.
The Proxy had to have many from a SIP-Server functionalities. It has to maintain flows tables with users-ID, ports and servers IP used in each communication flow (other infos could help in other tasks, but I thing that these one are the minimum), so that it could decide to whom send each packet from each flow.
So, do I make my doubts/points clear to You? If my understand is wrong, sorry and please correct me where necessary.
Edson.
P.S.: In my scenario there is no SER-2-SER communications, but another problem would be having two (or more) sites like the "Router" one. How to make than communicate each other through NAT GW/FW?
- ----- Original Message -----
From: "Adrian Georgescu" < ag(a)ag-projects.com>
To: < serusers(a)lists.iptel.org>
Sent: Thursday, January 29, 2004 8:00 AM
Subject: [Serusers] SER + Proxy + NAT
> Edson,
>
> Putting a NAT traversal solution behing NAT is a chicken and eg
> problem, isn't it?
>
> --
> Adrian
>
>
>
> Hi all...
>
> I look through the list's archives, but an not finding info to help me.
>
> The goal is use SER but not instaled in the GW/FW (it's not an
> acceptable
> option, well it's acceptable, but not for now). So I'm trying to put
> the SER
> in the Internal LAN (it could be installed in a DMZ also). So the
> question
> is if there is any proxy that could be putted on the GW/FW to handle
> incomming calls (INVITEs) and forward it correctly to the SER machine
> taken
> over the NAT issues?
>
> I already look at SIProxd and RTProxy, but the first didn't forward
> incomming calls, and the second demands that it be instaled, with SER
> on the
> GW/FW. I also am looking at SERMediaProxy (RTProxy alternative) but the
> documentations aren't sufficient detailed to answer my question. Any
> help
> would be appreciated.
>
> Edson.
>
>
- --------------------------------------------------------------------------------
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
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Edson,
Putting a NAT traversal solution behing NAT is a chicken and eg
problem, isn't it?
--
Adrian
Hi all...
I look through the list's archives, but an not finding info to help me.
The goal is use SER but not instaled in the GW/FW (it's not an
acceptable
option, well it's acceptable, but not for now). So I'm trying to put
the SER
in the Internal LAN (it could be installed in a DMZ also). So the
question
is if there is any proxy that could be putted on the GW/FW to handle
incomming calls (INVITEs) and forward it correctly to the SER machine
taken
over the NAT issues?
I already look at SIProxd and RTProxy, but the first didn't forward
incomming calls, and the second demands that it be instaled, with SER
on the
GW/FW. I also am looking at SERMediaProxy (RTProxy alternative) but the
documentations aren't sufficient detailed to answer my question. Any
help
would be appreciated.
Edson.
Hello,
it seems like that Cisco 827-V router doesn't include any from-tag within
its INVITEs...which is NOT RFC CONFORM!.
It may be due to some kind of proprietary function mode. There is nothing we
can do on the SEMS side to solve the problem.
Does anybody else have such a gateway and got it work with voicemail?
-Raphael.
----- Original Message -----
From: "Goh Sek Chye" <sekchye(a)lgatelecom.net>
To: <raphael.coeffic(a)gmx.de>
Sent: Monday, January 26, 2004 11:08 AM
Subject: sems problem
> Hi Raphael,
>
> I am using your voice mail program sems and it is very
> useful to us. Thank you very much!
>
> However, I have a problem here.
>
> I have a Cisco router 827-4V which is connected to a PBX using
> FXS port. The Cisco router is configured to point to a SIP server
> running SER and SEMS.
>
> When a PBX user calls a SIP user who is not online,
> he hears a long death silence followed by a fast busy tone.
>
> I am clueless what is the problem and would like to seek your
> help.
>
> The following is the debug message from the SER server and
> the debug message from the Cisco router. Thanks in advance for your help.
>
> =================
>
> SER debug messages
> ====================
>
> Jan 26 17:13:41 sip1 ser[25648]: SIP Request:
> Jan 26 17:13:41 sip1 ser[25648]: method: <INVITE>
> Jan 26 17:13:41 sip1 ser[25648]: uri:
<sip:10650004@192.168.0.222;user=phone>
> Jan 26 17:13:41 sip1 ser[25648]: version: <SIP/2.0>
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=1
> Jan 26 17:13:41 sip1 ser[25648]: end of header reached, state=5
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: Via found, flags=1
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: this is the first via
> Jan 26 17:13:41 sip1 ser[25648]: After parse_msg...
> Jan 26 17:13:41 sip1 ser[25648]: preparing to run routing scripts...
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG : is_maxfwd_present: searching for
max_forwards header
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=128
> Jan 26 17:13:41 sip1 ser[25648]: end of header reached, state=9
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: get_hdr_field: <To> [41];
uri=[sip:10650004@192.168.0.222;user=phone]
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: to body
[<sip:10650004@192.168.0.222;user=phone>^M ]
> Jan 26 17:13:41 sip1 ser[25648]: get_hdr_field: cseq <CSeq>: <101>
<INVITE>
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: is_maxfwd_present: value = 6
> Jan 26 17:13:41 sip1 ser[25648]: end of header reached, state=9
> Jan 26 17:13:41 sip1 ser[25648]: check_self - checking if host==us: 13==13
&& [192.168.0.222] == [192.168.0.222]
>
> Jan 26 17:13:41 sip1 ser[25648]: lookup(): '10650004' Not found in usrloc
> Jan 26 17:13:41 sip1 ser[25648]: lookup(): '10650004' Not found in usrloc
> Jan 26 17:13:41 sip1 ser[25648]: ************************ route[3]:vm:1
*************************************
> Jan 26 17:13:41 sip1 ser[25648]: route[3]:vm:2
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_addifnew: msg id=789 , global
msg id=787 , T on entrance=0xffffffff
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=-1
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: get_hdr_body : content_length=134
> Jan 26 17:13:41 sip1 ser[25648]: found end of header
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=60
> Jan 26 17:13:41 sip1 ser[25648]: t_lookup_request: start searching:
hash=43402, isACK=0
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: proceeding to pre-RFC3261
transaction matching
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_lookup_request: no transaction
found
> Jan 26 17:13:41 sip1 ser[25648]: DBG: callback type 2, id 3 entered
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=44
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: noisy_timer set for accounting
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_check: msg id=789 global id=789
T start=0x402e1070
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_check: T alredy found!
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=-1
> Jan 26 17:13:41 sip1 ser[25648]: check_via_address(203.92.75.94,
203.92.75.94, 0)
> Jan 26 17:13:41 sip1 ser[25648]: WARNING:vqm_resize: resize(0) called
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: reply sent out. buf=0x80d0cd0:
SIP/2.0 1..., shmem=0x402d8cd8: SIP/2.0 1
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_reply: finished
> Jan 26 17:13:41 sip1 ser[25648]: route[3]:method==INVITE || REFER
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=-1
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_check: msg id=789 global id=789
T start=0x402e1070
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: t_check: T alredy found!
> Jan 26 17:13:41 sip1 ser[25648]: vm: calculated route:
> Jan 26 17:13:41 sip1 ser[25648]: vm: next r-uri:
sip:41703@203.92.75.94:5060;user=phone
> Jan 26 17:13:41 sip1 ser[25648]: parse_headers: flags=-1
> Jan 26 17:13:41 sip1 ser[25648]: submit_query(): select email_address from
subscriber where username='10650004'
> Jan 26 17:13:41 sip1 ser[25648]: vm: write_to_vm_fifo: <0.1 voicemail
INVITE 10650004 sekchye(a)yahoo.com 203.92.64
> .216 192.168.0.222 5060 sip:10650004@192.168.0.222;user=phone
sip:41703@203.92.75.94:5060;user=phone "41703" <sip
> :41703@203.92.75.94>^M <sip:10650004@192.168.0.222;user=phone>
C475045A-4F1611D8-827089FC-A1038096(a)203.92.75.94 .
> . 101 43402:300376901 . . . v=0^M o=CiscoSystemsSIP-GW-UserAgent 5931
2351 IN IP4 203.92.75.94^M s=SIP Call^M c=
> IN IP4 203.92.75.94^M t=0 0^M m=audio 20416 RTP/AVP 0^M >
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: write_to_vm_fifo: write completed
> Jan 26 17:13:41 sip1 ser[25648]: DEBUG: add_to_tail_of_timer[0]:
0x402e11bc
> Jan 26 17:13:41 sip1 ser[25648]: receive_msg: cleaning up
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: version= <0.1>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.cmd= <voicemail>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.method= <INVITE>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.user= <10650004>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.email= <sekchye(a)yahoo.com>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.domain= <192.168.0.222>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.dstip= <192.168.0.222>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.port= <5060>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.r_uri=
<sip:10650004@192.168.0.222;user=phone>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.from_uri=
<sip:41703@203.92.75.94:5060;user=phone>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.from= <"41703"
<sip:41703@203.92.75.94>>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.to=
<<sip:10650004@192.168.0.222;user=phone>>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.callid=
<C475045A-4F1611D8-827089FC-A1038096(a)203.92.75.94>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.from_tag= <>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.to_tag= <>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cseq_str= <101>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cseq= <101>(101)
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.key= <43402:300376901>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.route= <>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: cmd.next_hop= <>
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: hdrs: `'
> Jan 26 17:13:41 sip1 Sems[17680]: Debug: body: `v=0
o=CiscoSystemsSIP-GW-UserAgent 5931 2351 IN IP4 203.92.75.94
> s=SIP Call c=IN IP4 203.92.75.94 t=0 0 m=audio 20416 RTP/AVP 0 '
> Jan 26 17:13:41 sip1 Sems[17680]: Error: invalid FIFO command:
cmd.from_tag is empty !!!
> Jan 26 17:13:48 sip1 ser[25685]: MSILO:clean_silo: cleaning stored
messages - 874200
> Jan 26 17:13:48 sip1 ser[25685]: MSILO:clean_silo: cleaning expired
messages
> Jan 26 17:13:48 sip1 ser[25685]: submit_query(): delete from silo where
exp_time<=1075108428
> Jan 26 17:13:55 sip1 ser[25685]: DEBUG: timer routine:0,tl=0x402e11bc
next=(nil)
> Jan 26 17:13:55 sip1 ser[25685]: DEBUG: FR_handler:stop retr. and send
CANCEL (0x402e1070)
> Jan 26 17:13:55 sip1 ser[25685]: DEBUG: FR_handler:stop retr. and send
CANCEL (0x402e1070)
> Jan 26 17:13:55 sip1 ser[25685]: ->>>>>>>>> T_code=100, new_code=408
> Jan 26 17:13:55 sip1 ser[25685]: DBG: callback type 6, id 4 entered
> Jan 26 17:13:55 sip1 ser[25685]: submit_query(): insert into missed_calls
(from_uri,to_uri,sip_method,i_uri,o_uri
> ,sip_from,sip_callid,sip_to,sip_status,username,totag,fromtag,time )
values ('sip:41703@203.92.75.94','sip:106500
>
04(a)192.168.0.222;user=phone','INVITE','sip:10650004@192.168.0.222;user=phone
','sip:10650004@192.168.0.222;user=ph
> one','"41703"
<sip:41703@203.92.75.94>','C475045A-4F1611D8-827089FC-A1038096(a)203.92.75.94'
,'<sip:10650004@203.92.
> 64.216;user=phone>','408 Request
Timeout','10650004','n/a','n/a','2004-01-26 09:13:55')
> Jan 26 17:13:55 sip1 ser[25685]: rc_get_seqnbr: couldn't open sequence
file /var/run/radius.seq: Permission denie
> d
> Jan 26 17:13:55 sip1 ser[25685]: DEBUG: relay_reply: branch=0, save=0,
relay=0
> Jan 26 17:13:55 sip1 ser[25685]: parse_headers: flags=-1
> Jan 26 17:13:55 sip1 ser[25685]: check_via_address(203.92.75.94,
203.92.75.94, 0)
> Jan 26 17:13:55 sip1 ser[25685]: DEBUG: reply relayed. buf=0x80cef08:
SIP/2.0 4..., shmem=0x402d47a8: SIP/2.0 4
> Jan 26 17:13:55 sip1 ser[25685]: DBG: callback type 7, id 1 entered
> Jan 26 17:13:55 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[4]:
0x402e1128
> Jan 26 17:13:55 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[0]:
0x402e1140
> Jan 26 17:13:55 sip1 ser[25685]: DEBUG: final_response_handler : done
> Jan 26 17:13:55 sip1 ser[25647]: SIP Request:
> Jan 26 17:13:55 sip1 ser[25647]: method: <ACK>
> Jan 26 17:13:55 sip1 ser[25647]: uri:
<sip:10650004@192.168.0.222;user=phone>
> Jan 26 17:13:55 sip1 ser[25647]: version: <SIP/2.0>
> Jan 26 17:13:55 sip1 ser[25647]: parse_headers: flags=1
> Jan 26 17:13:55 sip1 ser[25647]: end of header reached, state=5
> Jan 26 17:13:55 sip1 ser[25647]: parse_headers: Via found, flags=1
> Jan 26 17:13:55 sip1 ser[25647]: parse_headers: this is the first via
> Jan 26 17:13:55 sip1 ser[25647]: After parse_msg...
> Jan 26 17:13:55 sip1 ser[25647]: DEBUG : sl_filter_ACK: to late to be a
local ACK!
> Jan 26 17:13:55 sip1 ser[25647]: preparing to run routing scripts...
> Jan 26 17:13:55 sip1 ser[25647]: DEBUG : is_maxfwd_present: searching for
max_forwards header
> Jan 26 17:13:55 sip1 ser[25647]: parse_headers: flags=128
> Jan 26 17:13:55 sip1 ser[25647]: DEBUG: add_param:
tag=baea774f5e5e7e26323ab05f178242c9-2bd7
> Jan 26 17:13:55 sip1 ser[25647]: end of header reached, state=29
> Jan 26 17:13:55 sip1 ser[25647]: DEBUG: get_hdr_field: <To> [83];
uri=[sip:10650004@192.168.0.222;user=phone]
> Jan 26 17:13:55 sip1 ser[25647]: DEBUG: to body
[<sip:10650004@192.168.0.222;user=phone>]
> Jan 26 17:13:55 sip1 ser[25647]: DEBUG: is_maxfwd_present: value = 6
> Jan 26 17:13:55 sip1 ser[25647]: end of header reached, state=9
> Jan 26 17:13:55 sip1 ser[25647]: check_self - checking if host==us: 13==13
&& [192.168.0.222] == [192.168.0.222]
>
> Jan 26 17:13:55 sip1 ser[25647]: lookup(): '10650004' Not found in usrloc
> Jan 26 17:13:55 sip1 ser[25647]: lookup(): '10650004' Not found in usrloc
> Jan 26 17:13:55 sip1 ser[25647]: ************************ route[3]:vm:1
*************************************
> Jan 26 17:13:55 sip1 ser[25647]: receive_msg: cleaning up
> Jan 26 17:13:56 sip1 ser[25685]: DEBUG: timer routine:4,tl=0x402e1128
next=(nil)
> Jan 26 17:13:56 sip1 ser[25685]: DEBUG: retransmission_handler : reply
resending (t=0x402e1070, SIP/2.0 4 ... )
> Jan 26 17:13:56 sip1 ser[25685]: DEBUG: reply retransmitted.
buf=0x42237da0: SIP/2.0 4..., shmem=0x402d47a8: SIP/
> 2.0 4
> Jan 26 17:13:56 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[5]:
0x402e1128
> Jan 26 17:13:56 sip1 ser[25685]: DEBUG: retransmission_handler : done
> Jan 26 17:13:58 sip1 ser[25685]: DEBUG: timer routine:5,tl=0x402e1128
next=(nil)
> Jan 26 17:13:58 sip1 ser[25685]: DEBUG: retransmission_handler : reply
resending (t=0x402e1070, SIP/2.0 4 ... )
> Jan 26 17:13:58 sip1 ser[25685]: DEBUG: reply retransmitted.
buf=0x42237da0: SIP/2.0 4..., shmem=0x402d47a8: SIP/
> 2.0 4
> Jan 26 17:13:58 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[6]:
0x402e1128
> Jan 26 17:13:58 sip1 ser[25685]: DEBUG: retransmission_handler : done
> Jan 26 17:14:02 sip1 ser[25685]: DEBUG: timer routine:6,tl=0x402e1128
next=(nil)
> Jan 26 17:14:02 sip1 ser[25685]: DEBUG: retransmission_handler : reply
resending (t=0x402e1070, SIP/2.0 4 ... )
> Jan 26 17:14:02 sip1 ser[25685]: DEBUG: reply retransmitted.
buf=0x42237da0: SIP/2.0 4..., shmem=0x402d47a8: SIP/
> 2.0 4
> Jan 26 17:14:02 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[7]:
0x402e1128
> Jan 26 17:14:02 sip1 ser[25685]: DEBUG: retransmission_handler : done
> Jan 26 17:14:06 sip1 ser[25685]: DEBUG: timer routine:7,tl=0x402e1128
next=(nil)
> Jan 26 17:14:06 sip1 ser[25685]: DEBUG: retransmission_handler : reply
resending (t=0x402e1070, SIP/2.0 4 ... )
> Jan 26 17:14:06 sip1 ser[25685]: DEBUG: reply retransmitted.
buf=0x42237da0: SIP/2.0 4..., shmem=0x402d47a8: SIP/
> 2.0 4
> Jan 26 17:14:06 sip1 ser[25685]: DEBUG: add_to_tail_of_timer[7]:
0x402e1128
> Jan 26 17:14:06 sip1 ser[25685]: DEBUG: retransmission_handler : done
> Jan 26 17:14:10 sip1 ser[25685]: DEBUG: timer routine:0,tl=0x402e1140
next=(nil)
>
>
>
>
>
>
>
>
>
> Debug message from Cisco 827-V router
> =====================================
> Jan 26 17:13:41.158 SGT: Sent:
> INVITE sip:10650004@192.168.0.222;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.202:5060
> From: "41703" <sip:41703@192.168.0.202>
> To: <sip:10650004@192.168.0.222;user=phone>
> Date: Mon, 26 Jan 2004 17:13:41 GMT
> Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202
> Cisco-Guid: 3162002760-1326846424-2188282364-2701361302
> User-Agent: Cisco-SIPGateway/IOS-12.x
> CSeq: 101 INVITE
> Max-Forwards: 6
> Timestamp: 1075108421
> Contact: <sip:41703@192.168.0.202:5060;user=phone>
> Expires: 180
> Content-Type: application/sdp
> Content-Length: 136
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 5931 2351 IN IP4 192.168.0.202
> s=SIP Call
> c=IN IP4 192.168.0.202
> t=0 0
> m=audio 20416 RTP/AVP 0
>
> Jan 26 17:13:41.170 SGT: Received:
> SIP/2.0 100 Trying -- just a second
> Via: SIP/2.0/UDP 192.168.0.202:5060
> From: "41703" <sip:41703@192.168.0.202>
> To: <sip:10650004@192.168.0.222;user=phone>
> Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202
> CSeq: 101 INVITE
> Server: Sip EXpress router (0.8.11 (i386/linux))
> Content-Length: 0
>
>
>
> Jan 26 17:13:55.338 SGT: Received:
> SIP/2.0 408 Request Timeout
> Via: SIP/2.0/UDP 192.168.0.202:5060
> From: "41703" <sip:41703@192.168.0.202>
> To:
<sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9
-2bd7
> Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202
> CSeq: 101 INVITE
> Server: Sip EXpress router (0.8.11 (i386/linux))
> Content-Length: 0
>
>
>
> Jan 26 17:13:55.358 SGT: Sent:
> ACK sip:10650004@192.168.0.222;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.202:5060
> From: "41703" <sip:41703@192.168.0.202>
> To:
<sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9
-2bd7
> Date: Mon, 26 Jan 2004 17:13:41 GMT
> Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202
> Max-Forwards: 6
> Content-Length: 0
> CSeq: 101 ACK
>
>
>
> Jan 26 17:13:56.346 SGT: Received:
> SIP/2.0 408 Request Timeout
> Via: SIP/2.0/UDP 192.168.0.202:5060
> From: "41703" <sip:41703@192.168.0.202>
> To:
<sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9
-2bd7
> Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202
> CSeq: 101 INVITE
> Server: Sip EXpress router (0.8.11 (i386/linux))
> Content-Length: 0
>
>
>
>
> Jan 26 17:14:02.406 SGT: Received:
> SIP/2.0 408 Request Timeout
> Via: SIP/2.0/UDP 192.168.0.202:5060
> From: "41703" <sip:41703@192.168.0.202>
> To:
<sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9
-2bd7
> Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202
> CSeq: 101 INVITE
> Server: Sip EXpress router (0.8.11 (i386/linux))
> Content-Length: 0
>
>
>
> Jan 26 17:14:06.446 SGT: Received:
> SIP/2.0 408 Request Timeout
> Via: SIP/2.0/UDP 192.168.0.202:5060
> From: "41703" <sip:41703@192.168.0.202>
> To:
<sip:10650004@192.168.0.222;user=phone>;tag=baea774f5e5e7e26323ab05f178242c9
-2bd7
> Call-ID: C475045A-4F1611D8-827089FC-A1038096(a)192.168.0.202
> CSeq: 101 INVITE
> Server: Sip EXpress router (0.8.11 (i386/linux))
> Content-Length: 0
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> --
> Best regards
> Goh Sek Chye
> LGA Telecom Pte Ltd
> PGP KeyID: AE6D04A2
>
> PS: To import my PGP key,
> gpg --recv-keys --keyserver wwwkeys.pgp.net AE6D04A2
>
> ---------------------------------------------------------------
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After reviewing the documentation and mailing list I think I can conclude that their is no way currently to have a calls end up at a Voicemal process or server based on timeout values, at this time provided by fr_ commands, without setting this value for all calls and having all call remaining stateful.
If true has anyone found any work arounds to this problem?
I sawl mention from an email from Jan that it was a feature on the list anyone seen any update related to it?
Thanks
Hello List,
Does anyone know of any user agent - commercial or opensource - that supports symmetric RTP for both audio and video?
Thanks in advance,
Dhiraj Bhuyan
Network Security Specialist,
BT Exact Business Assurance Solutions
Tel: +44 1473 643932
Mob: +44 7962 012145
Email: dhiraj.2.bhuyan(a)bt.com
Dear List,
I have a Cisco 1700 FXO and set up a dial peer to talk to ser, on a sip session. I am trying to establish how I could identify a sip registered user in ser, based on the caller id value obtained from the sip session record passed from the Cisco to sip server. The identified user in ser will be recepiet (To:) in the INVITE message.
Appreciate yours assistance.
Deen
Hi,
I'm trying to use the CPL-C module in ser version 0.8.12 (stable).
However, I cannot get it to compile. I'm getting the following error
message:
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wall -Wa,-xarch=v8plus
-DNAME='"cpl-c.so"' -DVERSION='"0.8.12"' -DARCH='"sparc64"'
-DOS='"solaris"' -DCOMPILER='"gcc 2.95"' -D__CPU_sparc64 -D__OS_solaris
-DCFG_DIR='"/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK
-DUSE_IPV6 -DUSE_TCP -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK
-DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME
-DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -I/usr/include/libxml2 -c
cpl_run.c -o cpl_run.o
In file included from cpl_run.c:829:
cpl_proxy.h: In function `run_proxy':
cpl_proxy.h:444: structure has no member named `register_req_cb'
cpl_proxy.h:450: structure has no member named `register_req_cb'
make: *** [cpl_run.o] Error 1
The above is a compile log from Solaris, however, I'm getting the same
on Redhat 8. Any hints how to fix it?
Thanks,
Mario
--
The University of Stirling is a university established in Scotland by
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immediately if you or your employer do not consent to Internet email
for messages of this kind.
Please always CC the list. Read the admin guide or search through the
mailing lists archives. Look for t_relay_to_udp, forward or similar
functions.
If you gateway is running on IP 1.2.3.4 then you can use
t_relay_to_udp("1.2.3.4");
http://iptel.org/ser/admin.html
Jan.
On 27-01 12:46, Raymond May wrote:
> Jan,
> Sorry to bother you again but I am trying to figure
> out how to configure SER to forward digits to the
> SIP/H.323 gateway ( Forward Digits from SER to
> SIP/H.323 Gateway ). Your help is well appreciated.
>
> My Private VoIP Network -- SER -- SIP/H.323 Gateway --
> Corporate Users
>
> Brgds,
> Ray May
>
>
> --- Jan Janak <jan(a)iptel.org> wrote:
> > I guess CallManager is using Skinny protocol, isn't
> > it ? If so then you
> > will need some protocol translator (I am not aware
> > of any such).
> >
> > Jan.
> >
> > On 23-01 10:25, Raymond May wrote:
> > > Hi SER Gurus,
> > > My SER server has been running just
> > > perfect for the last 9 months and I am now trying
> > to
> > > connect my private VoIP network to my Employer's.
> > They
> > > have a Cisco Call Manager and I am trying to
> > figure
> > > out how to configure SER so I can make calls from
> > my
> > > private network to users that are registered to
> > the
> > > Corporate ( Cisco Call Manager ). Ideally I would
> > like
> > > to be able to make calls from my private network
> > to
> > > the users on the Corporate and have users on the
> > > Corporate be able to reach users on my VoIp
> > Network (
> > > SER Server ). Any suggestions would be
> > appreciated.
> > > Below is current connectivity between my private
> > > network and our corporate network.
> > >
> > > SER Users --- SER Server --- Internet --- Cisco
> > Call
> > > Manager --- Corporate Users
> > >
> > >
> > > Brgds,
> > > Ray May
> > >
> > > __________________________________
> > > Do you Yahoo!?
> > > Yahoo! SiteBuilder - Free web site building tool.
> > Try it!
> > > http://webhosting.yahoo.com/ps/sb/
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
>
>
> __________________________________
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Thank you Raphel,
What a mistake, I was blind!
I have now the voicemail working, at least answering to the caller, I didn't
yet test if it records the message and send it to the user email. I have to
work now in other things and later I will return to SEMS.
Thank very much.
Best regards,
João
-----Original Message-----
From: Raphael Coeffic [mailto:rco@iptel.org]
Sent: terça-feira, 27 de Janeiro de 2004 16:04
To: Joao Sampaio; serusers(a)lists.iptel.org
Subject: Re: [Serusers] Voicemail
Hello Joao,
forget the warning while starting SEMS, they are not so important.
Please the configuration parameter 'ser_fifo_name' within your Sems config
file.
It should point at the second Ser's fifo server, that means, the one you
configured within voicemail.cfg.
Else, Sems is trying to communicate with the first instance (the proxy one),
which didn't load the vm module.
That means, it should work if you change that value to '/tmp/vm_ser_fifo'.
-Raphael.
