I was looking at an old thread regarding this.
http://lists.iptel.org/pipermail/serusers/2003-July/002003.html
1) How do I know when to direct it to this portion of the code? After x
rings?
2) Am I mistaken in thinking that there is a voicemail system built in?
Do I need to install a separate voicemail system? If not, how do I
configure the voice mail system? If so, can anyone recommend one?
Thanks,
Stephen
I am using a Cisco ATA 186 and a Cisco AS5350 in conjunction with a SER
SIP server. When I try to place calls from the ATA to the PSTN, I am
getting the following error. Is there a command that I can issue on the
Gateway to provide more information for debug?
Thanks,
Stephen
Total call-legs: 2
1284 : 46480131hs.94 +-1 +0 pid:1 Originate 13125551212
dur 00:00:00 tx:0/0 rx:0/0 41 (bearer capability not implemented)
Telephony 3/0:D (180): tx:0/0/0ms None noise:0dBm acom:0dBm
1284 : 46480129hs.95 +-1 +3 pid:700 Answer 256
dur 00:00:00 tx:0/0 rx:0/0 41 (bearer capability not implemented)
IP 209.242.10.153:16384 rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms
g711ulaw
U 66.155.138.5:5060 -> 209.242.10.153:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
209.242.10.153;branch=z9hG4bK7b35.fb17
329.0,SIP/2.0/UDP 216.145.234.84:5060..From:
<sip:256@209.242.10.153;user=p
hone>;tag=741484739..To:
<sip:18475551212@209.242.10.153;user=phone>;tag=1B
C0FF10-3C3..Date: Thu, 06 Jan 2000 09:20:30 GMT..Call-ID:
1667923238(a)216.14
5.234.84..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 1
INVITE..Content-Length
: 0....
#
U 66.155.138.5:5060 -> 209.242.10.153:5060
SIP/2.0 501 Not Implemented..Via: SIP/2.0/UDP
209.242.10.153;branch=z9hG4bK
7b35.fb17329.0,SIP/2.0/UDP 216.145.234.84:5060..From:
<sip:256@209.242.10.1
53;user=phone>;tag=741484739..To:
<sip:18475551212@209.242.10.153;user=phon
e>;tag=1BC0FF10-3C3..Date: Thu, 06 Jan 2000 09:20:30 GMT..Call-ID:
16679232
38@216.145.234.84..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 1
INVITE..Conte
nt-Length: 0....
while attempting logon at
http://mydomain.com/admin/index,php with username and
password , i get an "error in SQL query, line: 24"
following the HowTo guide,
i ran /usr/sbin/ser_mysql.sh create (version 1.38.2.6)
when i query "select * from user;" i got error "no
database selected"
but when i connect ser;
show tables; i can see all the table fields.
on /var/www/html/config.php, i have configured the
database connection user to "root" password
"mysqlpassword" DB name "ser".
any insights greatly appreciated.
tim
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Hi all,
Im replacing an old analog PBX with cisco ATA's, ser and a vega 50 BRI.
Currently, on the old system, when some one rings the PSTN number, it
goes through and rings two different phones (one upstairs, one
downstairs), and the first person to answer the phone gets the call.
Now, I need to replicate the behavior with ser, and the ata 186's. Has
any one attempted this type of behavior? Is it within the capability of sip?
Thanks,
-Jev
Hi everybody,
I'm having troubles with the Serweb 0.8.11 /user_interface/login.
I have no problem with the standard admin login (after some corrections...)
but urls in the user_interface folder don't return any result. I just got
endless timer pointer.
Any help?
thank you
Philippe
Hi,
Adding record_route(); to my routing block solved this problem, calls
work great now! thank you for the pointer :)
-Jev
Greg Fausak wrote:
> Howdy,
>
> I had this problem as well. I believe it was due to the fact
> that I didn't to record routing, and the ACK didn't get back.
> My calls lasted exactly 9 seconds.
>
> Anyway, it is really handy to do call tracing to debug
> these sorts of things. If you have a unix machine, tcpdump -w,
> or a windows machine can use ethereal. You save the sniffer
> session and then process it with 'sipscenario'. It makes a very
> easy to look at call flow, which will probably highlight the
> problem.
>
> Of course to do sniffing you have to have a machine that sees all
> of the packets.
>
> ---greg
> Greg Fausak
> Addaline.com, Inc.
>
>
>
>>Hi,
>>
>>I'm setting up a vega 50 BRI with ser, and I can make in/out bound
>>calls, quality is great!
>>
>>I have a problem, when I dial from a sip phone to a land line, my call
>>gets set up, and works great, but after approx 15 seconds it drops the
>>call. Below is what I get in the logs from ser.
>>
>>
>> 0(29026) DEBUG: is_maxfwd_present: max_forwards header not found!
>> 0(29026) DEBUG: add_param: tag=4063815580
>> 0(29026) end of header reached, state=29
>> 0(29026) parse_headers: flags=256
>> 0(29026) find_first_route(): No Route headers found
>> 0(29026) loose_route(): There is no Route HF
>> 0(29026) check_self - checking if host==us: 9==9 && [10.0.0.11] ==
>>[10.0.0.11]
>> 0(29026) lookup(): '0868594416' Not found in usrloc
>> 0(29026) Warning: sl_send_reply: I won't send a reply for ACK!!
>> 0(29026) receive_msg: cleaning up
>>
>>
>>My Version of ser is:
>># ser -V
>>version: ser 0.8.11 (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 1.162.2.5 2003/08/27 16:04:35 andrei Exp $
>>main.c compiled on 11:29:10 Sep 5 2003 with gcc 2.95
>>
>>Does anyone know hwta may be going wrong here? I'm working through docs
>>right now, but some guidance would be gratly appreciated;
>>
>>
>>My vega 50 version details:
>>Vega50(ISDN) Runtime System
>>Version: 06.02.05.1
>>Built: Aug 19 2003 12:21:46 T018
>>Serial #:0050580106bd
>>
>>Bootstrap System
>>Version: 1.09(0ws)
>>
>>ISDN Interface
>>Version: Not known
>>
>>FLASH Partition Information:
>>Partition 1: H.323 Firmware
>> Version: 06.01.05.1
>> Built: Jul 17 2003 12:28:38 T014
>>
>>Partition 2: SIP Firmware (ACTIVE)
>> Version: 06.02.05.1
>> Built: Aug 19 2003 12:21:46 T018
>>
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
Hi,
I'm setting up a vega 50 BRI with ser, and I can make in/out bound
calls, quality is great!
I have a problem, when I dial from a sip phone to a land line, my call
gets set up, and works great, but after approx 15 seconds it drops the
call. Below is what I get in the logs from ser.
0(29026) DEBUG: is_maxfwd_present: max_forwards header not found!
0(29026) DEBUG: add_param: tag=4063815580
0(29026) end of header reached, state=29
0(29026) parse_headers: flags=256
0(29026) find_first_route(): No Route headers found
0(29026) loose_route(): There is no Route HF
0(29026) check_self - checking if host==us: 9==9 && [10.0.0.11] ==
[10.0.0.11]
0(29026) lookup(): '0868594416' Not found in usrloc
0(29026) Warning: sl_send_reply: I won't send a reply for ACK!!
0(29026) receive_msg: cleaning up
My Version of ser is:
# ser -V
version: ser 0.8.11 (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 1.162.2.5 2003/08/27 16:04:35 andrei Exp $
main.c compiled on 11:29:10 Sep 5 2003 with gcc 2.95
Does anyone know hwta may be going wrong here? I'm working through docs
right now, but some guidance would be gratly appreciated;
My vega 50 version details:
Vega50(ISDN) Runtime System
Version: 06.02.05.1
Built: Aug 19 2003 12:21:46 T018
Serial #:0050580106bd
Bootstrap System
Version: 1.09(0ws)
ISDN Interface
Version: Not known
FLASH Partition Information:
Partition 1: H.323 Firmware
Version: 06.01.05.1
Built: Jul 17 2003 12:28:38 T014
Partition 2: SIP Firmware (ACTIVE)
Version: 06.02.05.1
Built: Aug 19 2003 12:21:46 T018
Hello,
I am having trouble with the textops' subst method. I am trying to
rewrite the Contact header field for certain requests - and according to
the debug output it seems to work (apart from the fact the the log
output is scrambled):
--
textops: subst_f: replacing at offset 341 [Contact:
<sip:felix@192.168.0.66:5064;transport=udp>;methods="INVIT] with
[Contact:
<sip:felix@beluga.homeunix.org:5064;transport=udp>;methods="INVITE,
MESSAGE, INFO, SUBSCRIBE, OPTIONS, B]E, CANCEL, NOTIFY, ACK"
---
However, if I dump out the whole request directly after calling subst(),
using 'exec_msg("echo REQUEST: ; cat - ; echo");', I get this:
---
REQUEST:
REGISTER sip:beluga.homeunix.org SIP/2.0
Via: SIP/2.0/UDP 192.168.0.66:5064
CSeq: 7656 REGISTER
To: "Felix Schmid" <sip:felix@beluga.homeunix.org>
Expires: 900
From: "Felix Schmid" <sip:felix@beluga.homeunix.org>
Call-ID: 742144068(a)192.168.0.66
Content-Length: 0
User-Agent: KPhone/3.11
Event: registration
Allow-Events: presence
Contact: <sip:felix@192.168.0.66:5064;transport=udp>;methods="INVITE,
MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
---
Why isn't the request in a rewritten state after subst() returns?
regards,
felix
--
PGP-Key: http://belugalounge.net/~felix/Felix%20Schmid.asc
Hello,
I remember having read somewhere that the 'ser.cfg' file used at
'iptel.org' can be downloaded somewhere in order to be used as an
example for many of the functionalities ser offers. Unfortunately I
can't remember where I read this, nor did I find it on the ftp server...
:(
Is this config file still available for download?
I am mainly asking because I am looking for an answer for the following
phenomenon:
I have SER running on my home network (on the gateway). When I try to
send an IM to my account at iptel.org using kphone and I use my gateway
SER as an outbound proxy, everything runs smoothly; I get a message back
from iptel.org that the IM will be delivered to me as soon as I login
the next time (what will not happen until I solved the NAT problem ;)).
Now, when I try the same without using my gateway as an outbound proxy,
I get the beloved message from iptel.org that it doesn't like my private
Contact address.
The only way the two requests differ is that the one using the outbound
proxy uses a record-route
I have attached the ngrep logfile, if anyone can explain me this
behaviour I'd be glad.
regards,
felix
--
PGP-Key: http://belugalounge.net/~felix/Felix%20Schmid.asc
Hello all,
I should be taking delivery of a vega 50 BRI gateway unit at the end of
this week. I plan to use this unit with ser, and a couple of cisco
ata-186's.
While I am waiting for the unit, I thought it might be a good idea to
ask here if anyone has past experience using the vega 50 with ser? I
know it's a pretty non-specific question, but it might help me get a
headstart when I begin to set everything up :)
Cheers!
-Jev
