Hi Guys!
I just would like to share that I was able to get a working setup using SER
as Softswitch, Asterisk as PSTN gateway and SIPROXD on my NAT Router.
SIPROXD is an open source ALG and it effectively handles sip nat traversals.
With it I dont have to run a seperate mediaproxy. When making calls from SIP
UA to PSTN, RTP is as below:
UA---NAT/SIPROXD---ASTERISK
for 2 UA behind the same NAT:
UA1--NAT---UA2
and for 2 UA behind different NATs:
UA1--NAT1----NAT2---UA2
Thus there is less latency on signals and less traffic on SER. My question
is, from the experience of other guys here, what do you think is the
drawback or advantages of using SIPROXD together with SER to solve SIP NAT
issues compared to other methods like using mediaproxy and rtpproxy?Will I
still be able to do other SER features like accounting?
Thanks!
_jeff
Hi there
I can not find where the call logs are in the ser database. What can I do to
enable call information logging (time stamps, destination, exc)? I am using
mysql.
Best regards,
Colin
Hey all you SER Guru's here is a noobie question that may seem dum, yet I
am still going to ask it.
I know SER is a SIP Proxy and a SIP Registrar and it works great.
My question is can you use SER as a full blown SIP PROXY as ASTERISK?
Does it handle call waiting?
Call Park?
Call Transfer?
3 way Conferencing?
If so what are the key strokes that activates this in SER?
Does SER Support Voicemail?
Inquiring minds want to know?
Also what is the username and password for Serweb?
I tried admin and heslo and it's telling bad username and password on the
screen.
Thanks.
Goran
Ok, so *98 seems to the standard for at least Bell South and Qwest and
also works with all my ATAs by default, so now I'm off to tackle
Caller-ID Blocking, Message Waiting Indicator (w/asterisk), and
Distinctive Ring.
Please feel free to share you comment/experiences If you've been down
this road already.
- Daryl
On 11/6/05, Patrick J. Menjor <jmenjor(a)comcast.net> wrote:
> *98
>
> -------- Original Message --------
> Subject: [Serusers] Vertical Service Codes
> Date: Sun, 6 Nov 2005 08:09:13 -0700
> From: Daryl Sanders <daryl.sanders(a)gmail.com>
> To: SER Users <serusers(a)iptel.org>, users(a)openser.org
>
>
>
> Can SER detect star codes? I see that there are many codes already
> defined in most ATAs. How would I make these work with SER?
>
> Is there a standard code for allowing users to check their voicemail
> from their SIP-phone/ATA? I was considering also using 00 as the code
> my users dial to access their voicemail? Would there any be any
> conflict with 00 and other functionality such as international
> calling? There are no DID/DDI that begin with 00 correct?
>
> Thanks!
> - Daryl
>
> _______________________________________________
> Serusers mailing list
> Serusers(a)iptel.org
> http://mail.iptel.org/mailman/listinfo/serusers
>
>
>
Hello All,
I am wanting to buy some did's from Japan, i wanted to know if there is a
Business Mailing list of SER, and if some one can point me to it.
Thank You,
Rehan
Can SER detect star codes? I see that there are many codes already
defined in most ATAs. How would I make these work with SER?
Is there a standard code for allowing users to check their voicemail
from their SIP-phone/ATA? I was considering also using 00 as the code
my users dial to access their voicemail? Would there any be any
conflict with 00 and other functionality such as international
calling? There are no DID/DDI that begin with 00 correct?
Thanks!
- Daryl
Hey All,
I have fr_inv_timer_avp pointing to an AVP named inv_timeout which
users are allowed to set and everything is working fine except upon call
forwarding from a failure route. Here's the situation...
- Call comes in for user(a)domain.com
- User at domain.com has inv_timeout set to 10s
- User at domain.com also has forward on unavailable to a PSTN number
- 10s of ringing to the user elapses
- The failure route kicks in and pushes the PSTN URI to a new branch
- avp_delete("inv_timeout"); is called
- avp_write("i:3600", "inv_timeout"); is called
- The PSTN number begins to ring
- 10s later SER sends a CANCEL to the PSTN gateway
How do I properly clear the users preset inv_timeout and reset it to 1
hour for the PSTN destination?
Thanks,
Evan
I keep coming back to that. However, a user can be a member of multiple calling scopes. for
example 1)Houston 2)LongDistance 3)Information
And I can't figure out how to do it nicely that way.. seems like maybe I'd have to create a
seperate column for each callscope? Surely there sure be a different way.. I simply want to
load a *list* of avps based on login credentials.. seems simple enough, right?
-Brett
---------- Original Message ----------------------------------
From: "Greger V. Teigre" <greger(a)teigre.com>
Date: Fri, 4 Nov 2005 08:10:04 +0100
>Brett,
>I have never used avpairs with sql (I use radius), but I believe the
>avp_db_load statement should be removed alltogether. The avps_column_str
>instructs the auth module to load the value in the callscope column ON
>SUCCESSFUL AUTHENTICATION, i.e. when the user is authenticated, the avp is
>already loaded and ready to be used.
>g-)
>
>----- Original Message -----
>From: "brett-ser-list" <brett-ser-list(a)worldcall.net>
>To: <serusers(a)lists.iptel.org>
>Sent: Friday, November 04, 2005 2:30 AM
>Subject: [Serusers] avp_db_load fails with avp as source
>
>
>> Hello All,
>> I'm trying (desperately) to use avp_db_load with an avp:
>> version: ser 0.9.3 (i386/linux)
>> flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
>> DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_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 1.197 2004/12/03 19:09:31 andrei Exp $
>> main.c compiled on 12:28:21 Aug 16 2005 with gcc 3.3
>>
>> modparam("auth_db","avps_column_str","callscope")
>> modparam("avpops","avp_aliases","callgroups=i:200")
>> modparam("avpops","avp_aliases","callscope=i:500")
>> avp_db_load("$callscope","$callgroups/avps") ;
>>
>> In watching the debug, I see the avp loaded just fine from the database
>> from the auth_db parameter:
>> 3(1735) DEBUG:avpops:print_avp: p=0xf51c20d8, flags=3
>> 3(1735) DEBUG: name=<callscope>
>> 3(1735) DEBUG: val_str=<200>
>>
>> but the above avp_db_load line is resulting in:
>> 3(1735) DEBUG:avpops:get_val_as_str: no avp found
>> 3(1735) ERROR:avpops:load_avps: failed to get uuid
>>
>> I also notice that it doesn't even bother to hit the database after the
>> auth_db hit. So unless the avp table has been cached (Which I have no idea
>> if it has..) how would it know?
>>
>> Seems like error #1: no avp found? Seems like it doesn't like my syntax
>> but according to the avpops documentation, it should be right:
>> source = (sip_uri|avp_alias|str_value)
>> ['/'('username'|'domain'|'uri'|'uuid')]
>>
>>
>> note:
>> avp_db_load("$callscope/uuid","$callgroups/avps") ;
>> gives me:
>> 0(0) ERROR:avpops:fixup_db_avp: source/flags "callscope" unknown!
>>
>> Look, what I really want to do is use the authentication credentials to
>> load an avp. Isn't that was avp is for? I don't want the FROM or the TO or
>> the [R]URI, but the actual AUTHentication username. Any ideas??
>>
>> Thanks!!
>> BTW, I've tried the newest SER DEVEL CVS, and OPENSER 1.0 with basically
>> the same results. Some custom modules I have would also make me want to
>> stick with 0.9.0.
>>
>> -Brett
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>>
>
>
Hi all,
I am working with asterisk to active voicemail for ser.
I am newbie for it.
So can you give me an example of ser.cfg for do that?
Thank you very much for your instruction.
