Hello:
I'm interested in how this community is providing voicemail service
to IP phone users registered to a SER proxy. If you can speak to this
point I'd appreciate any feedback you can provide. Specifically I'm
interested in:
1) What product is providing voice mail service?
2) With this system can users manipulate messages via the
telephone set?
3) If the answer to #2 is yes then how is this functionality
implemented. Do you provision a lead number for users
to call into voicemail?
Thanks,Steve
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-7903
fax: 215-898-9348
sip:blairs@upenn.edu
Hi, all
I'm implementing a black list, the control access to the gateway and other
control policies and I have a problem with the call forking.
In case of a call that generates a parallel forking I want the capability to
select if accept or refuse the call for each single contact, but if the
first reply that I send to the caller is an error message 4xx, the UA caller
shuts down the call.
What would I have to do to add my reject (caused by my access control
policy) to the call forking administration and so reply an error only if all
contacts can not be reachable?
I'm trying to implement a solution to reply an error only if all the
contacts can not be reachable.
To do this thing I need to know the number of contacts that the lookup
function has found for the request-uri that is arrived ( Is it possible to
know if I have a single contact or a parallel forking?) and, if there are
more than one contact associated to the sip uri, how can I go to serve the
next contact in list without sending a reply for the destinations that I
have to block?
Any suggestion?
Thanks
Daniele
I cannot seem to get my system to work with aliases. I have the userloc
and uri modules enabled as per the manual, as well as lookup("aliases");
command. I tried entering aliases with "serctl alias add 1001
sip:bo@domain.net" as well as "serctl alias add 1001(a)domain.net
sip:bo@domain.net". Both commands successfully complete but I always
get a busy signal on the other phone when I call 1001.
0(14359) SIP Request:
0(14359) method: <INVITE>
0(14359) uri: <sip:1001@domain.net;user=phone>
0(14359) version: <SIP/2.0>
0(14359) parse_headers: flags=1
0(14359) end of header reached, state=5
0(14359) parse_headers: Via found, flags=1
0(14359) parse_headers: this is the first via
0(14359) After parse_msg...
0(14359) preparing to run routing scripts...
0(14359) DEBUG : is_maxfwd_present: searching for max_forwards header
0(14359) parse_headers: flags=128
0(14359) end of header reached, state=9
0(14359) DEBUG: get_hdr_field: <To> [50];
uri=[sip:1001@domain.net;user=phone]
0(14359) DEBUG: to body [<sip:1001@domain.net;user=phone>
]
0(14359) get_hdr_field: cseq <CSeq>: <1> <INVITE>
0(14359) DEBUG: get_hdr_body : content_length=249
0(14359) found end of header
0(14359) DEBUG: is_maxfwd_present: max_forwards header not found!
0(14359) DEBUG: add_param: tag=2562321765
0(14359) end of header reached, state=29
0(14359) parse_headers: flags=256
0(14359) find_first_route(): No Route headers found
0(14359) loose_route(): There is no Route HF
0(14359) check_self - checking if host==us: 26==13 && [domain.net] ==
[x.x.x.x]
0(14359) check_self - checking if port 5060 matches port 5060
0(14359) lookup(): '1001' Not found in usrloc
0(14359) lookup(): '1001' Not found in usrloc
0(14359) parse_headers: flags=-1
0(14359) check_via_address(x.x.x.x, x.x.x.x, 0)
0(14359) receive_msg: cleaning up
0(14359) SIP Request:
0(14359) method: <ACK>
0(14359) uri: <sip:1001@domain.net;user=phone>
0(14359) version: <SIP/2.0>
0(14359) parse_headers: flags=1
0(14359) end of header reached, state=5
0(14359) parse_headers: Via found, flags=1
0(14359) parse_headers: this is the first via
0(14359) After parse_msg...
0(14359) parse_headers: flags=4
0(14359) DEBUG: add_param: tag=4ddff14623b40c25b5f0541000aec456.33da
0(14359) end of header reached, state=29
0(14359) DEBUG: get_hdr_field: <To> [92];
uri=[sip:1001@domain.net;user=phone]
0(14359) DEBUG: to body [<sip:1001@domain.net;user=phone>]
0(14359) DEBUG: sl_filter_ACK : local ACK found -> dropping it!
0(14359) receive_msg: cleaning up
# ser -V
version: ser 0.8.12 (i386/freebsd)
flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.168 2003/10/12 15:09:08 andrei Exp $
main.c compiled on 14:15:39 Nov 21 2003 with gcc 2.95
Bo Byrd
Hi,
Has any one done using Asterisk/Digium T1 card as PSTN gateway, and SER as
the backend soft switch?
I am looking to install Asterisk boxes in a few cities to act as PSTN
gateways and using SER to accept these calls. SER will have to communicate
with other long distance carriers to allow world wide calling.
Will this combination work? I like to know your comments or suggestions on
this combination.
thanks
Tony Lum
Hello everybody,
I'm trying to configure voicemail using sems. I've
figured out that I have problem with 'vm' module in
second ser instance. In cfg file I have problem with
these two lines:
loadmodule "/usr/lib/ser/modules/vm.so"
modparam("voicemail",
"db_url","sql://root:@127.0.0.1/ser")
I recieve error:
ser[4525]: DBT:dbt_cache_get_db: database
[sql://root:@127.0.0.1/ser] does not exists!
ser[4525]: DBT:dbt_init: cannot get the link to
database
ser[4525]: ERROR; vm_init_child: could not init db
sql://root:@127.0.0.1/ser
ser[4525]: init_mod_child(): Error while initializing
module voicemail
ser[4525]: init_child failed
However I use database properly with same parameters
for other modules. I tried with another database
(created with 'ser_mysql.sh copy vm'), unfortunately I
have same results. I recompiled ser, but I didn't
found lines to uncomment to activate voicemail, so I
concluded that voicemail is enabled by default in ser
0.8.12.
Can you help me solve this problem?
Milivoje
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Hi,
I'm getting this message when my grandstream ht286 trys to register. It
seens ht286 isn't sending authorization header field to SER. Or, ht286
sending strange contact (*) disturbs SER to process REGISTER requests ?
This message starts few days ago. As I changed my ser box to last cvs
version these days I'm confuse if it's a grandstream's problem or a
ser's problem.
ngrep logs: attached
--
[]s
Carlo Pires
U 193.126.48.33:5060 -> 200.193.194.142:5060
REGISTER sip:sip.televoip.net SIP/2.0..Via: SIP/2.0/UDP 192.168.0.111;branch=z9hG4bK4977acbbc428883a..From: "Reginaldo Paula Souza" <sip:reginaldo_ptt@sip.televoip.net>;tag=307110492b802983..To: <sip:reginaldo_ptt@sip.televoip.net>..Contact: *..Call-ID: 6221319848b4afe5@192.168.0.111..CSeq: 106 REGISTER..Expires: 0..User-Agent: Grandstream HT286 1.0.4.68..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0....
#
U 200.193.194.142:5060 -> 193.126.48.33:5060
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.0.111;branch=z9hG4bK4977acbbc428883a;rport=5060;received=193.126.48.33..From: "Reginaldo Paula Souza" <sip:reginaldo_ptt@sip.televoip.net>;tag=307110492b802983..To: <sip:reginaldo_ptt@sip.televoip.net>;tag=a791a9ab0b94eb4c05eab9b15c0fec4f.f9b6..Call-ID: 6221319848b4afe5@192.168.0.111..CSeq: 106 REGISTER..P-NATed-Caller: Yes..WWW-Authenticate: Digest realm="sip.televoip.net", nonce="40d46cd7c8d155389bef79bb516a0a2b2a5524f8"..Server: Sip EXpress router (0.8.13-dev-30-db_api (i386/linux))..Content-Length: 0..Warning: 392 200.193.194.142:5060 "Noisy feedback tells: pid=11293 req_src_ip=193.126.48.33 req_src_port=5060 in_uri=sip:sip.televoip.netout_uri=sip:sip.televoip.net via_cnt==1"....
I'm starting to use the pa module, and noticed that when it gets a SUBSCRIBE, the 200 response doesn't have an expires header, as it should. (RFC 3265, 3.1.1)
SUBSCRIBE sip:vmail@192.168.0.5 SIP/2.0
Via: SIP/2.0/UDP voipdev3.zoom.com:33050;rport
From: <sip:8100100@192.168.0.5:5060>;tag=a67e989d-13c4-173-5ac26-24be
To: <sip:vmail@192.168.0.5>
CSeq: 1 SUBSCRIBE
Call-ID: b5bb80-a67e989d-13c4-173-5ac26-5440
Expires: 3600
Event: presence
Max-Forwards: 70
Supported: replaces
Contact: <sip:8100100@192.168.0.5:5060>
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP voipdev3.zoom.com:33050;rport=33050;received=192.168.0.10
From: <sip:8100100@192.168.0.5:5060>;tag=a67e989d-13c4-173-5ac26-24be
To: <sip:vmail@192.168.0.5>;tag=92462461b2a59d6c6be4b5af5841f077-b62f
CSeq: 1 SUBSCRIBE
Call-ID: b5bb80-a67e989d-13c4-173-5ac26-5440
Content-Length: 0
I haven't spent a lot of time looking at the code for pa, but should this be an easy thing to fix?
Thanks,
--
Doug
