I am having some trouble with avpops and the from variable. I am
doing something like
avp_db_load("$to","$disabled");
which produces a query like
select value,attribute,type from usr_preferences where
username='+15183207486' AND attribute='50'
the problem is I need a query like select value,attribute,type from
usr_preferences where username='15183207486' AND
attribute='50' ,without the "+"
is there any way I can strip the 1st digit from the $from variable? I
have already tried using strip(1);
As always any help greatly appreciated.
Hi,
When I use X-pro to do a call test,the radius server debugs:
rad_recv: Access-Request packet from host 127.0.0.1:33407, id=111, length=256
User-Name = "<A href="mailto:003@159.226.58.100">003(a)159.226.58.100</A>"
Digest-Attributes = 0x0a05303033
Digest-Attributes = 0x01103135392e3232362e35382e313030
Digest-Attributes = 0x022a34333465343837383763373365376536316365356238636663636135333562643837303161633263
Digest-Attributes = 0x04147369703a3135392e3232362e35382e313030
Digest-Attributes = 0x030a5245474953544552
Digest-Attributes = 0x050661757468
Digest-Attributes = 0x090a3030303030303139
Digest-Attributes = 0x08224244303044334532333841363432363341434337414135414643444646454635
Digest-Response = "f4ac6b69ae532a2feb1b8b9e32e0fe86"
Service-Type = Sip-Session
Sip-Uri-User = "003"
NAS-IP-Address = 127.0.0.1
NAS-Port = 5060
modcall: entering group authorize
modcall[authorize]: module "preprocess" returns ok
modcall[authorize]: module "chap" returns noop
rlm_eap: EAP-Message not found
modcall[authorize]: module "eap" returns noop
rlm_digest: Converting Digest-Attributes to something sane...
Digest-User-Name = "003"
Digest-Realm = "159.226.58.100"
Digest-Nonce = "434e48787c73e7e61ce5b8cfcca535bd8701ac2c"
Digest-URI = "sip:159.226.58.100"
Digest-Method = "REGISTER"
Digest-QOP = "auth"
Digest-Nonce-Count = "00000019"
Digest-CNonce = "BD00D3E238A64263ACC7AA5AFCDFFEF5"
rlm_digest: Adding Auth-Type = DIGEST
modcall[authorize]: module "digest" returns ok
rlm_realm: Looking up realm "159.226.58.100" for User-Name = "<A href="mailto:003@159.226.58.100">003(a)159.226.58.100</A>"
rlm_realm: No such realm "159.226.58.100"
modcall[authorize]: module "suffix" returns noop
users: Matched DEFAULT at 152
modcall[authorize]: module "files" returns ok
modcall[authorize]: module "mschap" returns noop
modcall: group authorize returns ok
rad_check_password: Found Auth-Type DIGEST
auth: type "digest"
modcall: entering group authenticate
rlm_digest: Configuration item "User-Password" is required for authentication.
modcall[authenticate]: module "digest" returns invalid
modcall: group authenticate returns invalid
auth: Failed to validate the user.
Delaying request 34 for 1 seconds
Finished request 34
Going to the next request
Sending Access-Reject of id 110 to 127.0.0.1:33406
"003" is the user name and I have set the secret in the configure files of both radius server and radius client.
Anyone can give me an instruction?Many Thanks.
The last week__s I tried to insall webser on a machine other than my
asterisk box. I cannot make it!!! My next attempt is to use my Asterisk
box for openSER and Asterisk.
I will install openser and webser first with listening to port 5062
instead of the usually 5060 SIP port.
Than I will test openSER and serweb.
So far, I believe I can make it, but for the next steps I need some help.
What is the easiest way to get the Asterisk users into SER?
I want to swap in Asterisk the port 5060 to 5061 and from openSER the
port 5062 to 5060 AND forward all calls to Asterisk. What do I need
to do for that? If the user is registered in Asterisk AND in SER, than
he will get a different context, while if only in SER, than he should
only get [from_ser] which means only local calls.
In addition I want to allow for SER users ENUM and DUNDI. I found a
module ENUM, but I did not find a DUNDI module.
Any ideas, suggestions or recommendations for my path?
BTW, I read many docs about SER and it mentions iptel.cfg in the etc
directory, but I do not have that example. Where can I get this one?
bye
Ronald Wiplinger
I am using SER is the primary SIP server in front of four Asterisk PBX's
each located in a different city(e.g., City 1: 899xxxx, City 2: 598xxxx,
City 3: 356xxxx, City 4: 829xxxx). Users are identified by a 7 digit number
based on their home city. For PSTN-bound destinations, I want to be able to
direct SIP calls to the appropriate Asterisk server based on who is calling.
Using the src IP address isn't sufficient as users move around the network -
in one city one day and another city another day.
For example, I want a call from user 8991234(a)domain.com to
91234567(a)domain.com to be sent to the server in City 1 because user 8991234
resides in City 1. I can't use the src ip address because users move around
in the network. I have wondered about using a city-based db in addition to
a global db. Being a newcomer to this, I'm hoping that there is a simple
solution to this problem.
Your help and insights would be greatly appreciated,
Duncan
---
<mailto:Duncan.Glendinning@cox.net> Duncan.Glendinning(a)cox.net
Is there anyone that can help me with my following syntax? I am
basically trying to have all calls come in from an IP and if that IP
fails for some reason, it will roll to another IP.
if (src_ip== xxx.xxx.xxx.xxx){
t_on_failure("1");
rewritehost("xxx.xxx.xxx.xxx");
};
route[1]{
rewritehost("xxx.xxx.xxx.xxx");
break;
t_relay();
};
Hi,
I was just thinking.....
I have a multi-domain setup and using domain in lookup().
However, some UAC sends the proxy address in INVITE.
If all my contacts are registered with <username> and <domain>, lookup
("location") will return false since the domain in request will be
<proxy ip address>.....
Is there a way of reverse dns handle this or is just textops the way to
go?
--
mvh/best regards
Helge Waastad
System Engineer
Smartnet
Hello,
I read a previous post by Iqbal (missed caller call back) where a
subscriber can ring a pre-defined number which calls back their last
missed call...
Is my logic correct as follows?:
In the missed call section (i.e. when theres a 408/487 message) do:
avp_db_store("$ruri", "i:1000");
Then when the caller dials the callback number e.g.1471 this code is
invoked:
if (uri=~"^sip:1471@x.x.x.x"){
log(1, "in missed call return section");
avp_db_load("$from/username", "i:1000");
avp_pushto("i:1000", "$ruri");
// route as normal
};
I am getting an error when restarting SER:
ERROR: avops: fixup_pushto_avp: bad param 1; expected :
$[ruri|hdr_name|..]
Is it correct to push the AVP i:1000 to the ruri? Do I need to define
aliases in the modparam perhaps?
Any help would be appreciated.
Many thanks,
Aisling.
p.s. In case anyone was following my other post regarding asterisk
voicemail - It's now working using avpops, the problem was with my
asterisk
config. There is still a problem playing back the messages though.
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I would like to know why does my BYE method are always replied with a
'Call Leg/Transaction does not exist' . How do they compare whether the
transaction in the BYE method exist or not ? ( tag? ftag ? ) Are there
any thing in the config that might cause this kind of problem ? Just
want to highlight that all the calls are made in a good condition,
everything except when the call is ending. I have checked that the
phones have not received a prior BYE. Any other idea what is wrong ?
Here's a more detailed situation :-
Caller (PSTN) --> Voice Gateway --> OPENSER --> Callee (UA)
When Callee (UA) tried to end the call , OPENSER will forward a copy of
the BYE to Voice Gateway to inform him of the BYE.
The Gateway , somehow , replied with a 'Call Leg/Transaction Does Not
Exist' . The strange thing is, the Caller (PSTN) was somehow informed of
the BYE method and terminate the session . Anyone has any idea how to
handle these errors ? I will be glad to provide a ngrep for more
reference.
Please let me know if you dont understand.
Regards,
Sam
Hi,
We have a polycom sound point ip 500, sip app ver 1.5 (from
voip-info.org<http://voip-info.org>).
It turns out that it won't display MESSAGE ... but it does receive them and
it replies with a 200 OK. But no message on the screen or anywhere within
the phone.
The configuration has not been changed, it is pretty much the standard stuff
that comes from voip-info.org <http://voip-info.org>.
Anyone has ever came across the same problem?
Regards,
Cesc
PS - Getting the sip software from polycom resellers is a mess ... well, i
would say almost impossible. Next time we'll go for a costumer frendlier
phone manufacturer i guess.
I'm sorry but I have never played with these models so I can not help you much.....
In order to extend the INVITE retransmission, there must be somewhere in the web interface of these phones (if they have) something like T1 and T2, these values determine the retransmission period.
I guess you should check the Internet for information about your phones.
Hope somebody else with more experience can help you,
Samuel.
Unclassified.
>>> Alex <alexandergav(a)gmail.com> 10/11/05 05:45PM >>>
I am using cisco 7960 and grandstream. 286,486
On 10/11/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote:
>
> Which UA are you using??? It looks like all your issues comes from the
> implementation you are using as end-point and has nothing to do with SER.
>
> Samuel.
>
>
> Unclassified.
> >>> Alex <alexandergav(a)gmail.com> 10/11/05 11:06AM >>>
> Hi all
> I would like to explain some problem.
> When my UAC initiates INVITE the server response with 100 trying
> After receiving 100 trying the UAC sending again INVITE. ( HERE is the
> problem).
> RFC - 3261 ("After receiving a 1xx response, any retransmissions cease
> altogether ");
> Any help on that will be appreciated.
> In addition if there any way to extend the timeout between the INVITES.
> Thanks
>
>