Hi,
In "SIP Express Router RADIUS HOWTO" by Jan Janak I've read that by adding
the Sip-Rpid field in my freeradius user file "SIP Express Router can be
configured to put the phone number into Remote-Party-ID header field of the
SIP message"
Well, my question is, can I use the Rpid field and put it in the Caller id
field and how.?
If anybody has done this or even thinks he has a clue, please help.
Thanks
Hello,
I introduced the reminder functionality submitted by your patch in the
development version of msilo. I changed the patch to integrate better
with the module and I introduced several other functionalities. Here is
the description.
There is a new column "snd_time" in the table silo which specifies the
time after which the message can be delivered. If the value is 0, then
the message is not delivered based on timer.
If the message is not successfully delivered at the specified time, then
the snd_time is set to 0, to avoid flooding the network, if the
destination is not online. It will be delivered first time when he
registers.
The time for delivery can be set via AVP, the AVP must contain a value
in the format YYYYMMDDHHMMSS. The id of the avp can be set via module
parameter.
Here is a cfg example which can be used to set a reminder service:
modparam("msilo","reminder","sip:reminder@test.org")
modparam("msilo","send_time",10)
modparam("msilo","userid_avp",34)
modparam("msilo","snd_time_avp",35)
route {
....
if(method=="MESSAGE")
{
if(uri=~"sip:4442[0-9]{13}@")
{
strip(3);
avp_write("$ruri/username", "i:35");
avp_write("$from", "i:34");
m_store("0", "0" /*do not send info message back*/);
sl_send_reply("200", "ok");
exit;
};
};
.....
}
The documentation has been updated:
http://openser.org/docs/modules/1.1.x/msilo.html
Please note that you have to add the snd_time column to table silo (see
the definition in readme) and increase the version to 4 in table 'version'.
Testing and feedback are welcome.
Cheers,
Daniel
I am running ser 0.9.4 on FC4 with MySQL and all is working.
I want to implement forking based on SIP URI.
When someone dials 004888(which is a SIP URI just used for this matching), I
want to fork the call to 004990 and 004100 in parallel.
So I put this in my ser.cfg:
if (uri==myself)
{
if (uri=~"^sip:004888@~*")
{
rewriteuri("sip:004990@sip.tcto.net:49300");
append_branch("sip:004100@sip.tcto.net:49200");
route(1);
break;
};
};
Route[1]
{
if (!t_relay())
{
sl_reply_error();
};
}
Can anyone tell me what I am doing wrong? Or point me to some documentation
on forking. I have searched ONSIP and module documentation already.
Leo Papadopoulos
leo(a)ltcjp.com
Well, actually, ethereal supports Real Time. There is an option when you
start doing the capture that allows to show the packets in real time. Try to
look at that.
Jose Simoes
On 1/30/06, Aisling <ashling.odriscoll(a)cit.ie> wrote:
>
> Hi Jose,
>
>
>
> I looked at ethereal but I don't think it's possible to use that in real
> time i.e. you have to start the capture and when you think the call is
> over stop the capture and examine the results. I would like something in
> real-time…that makes me think tethereal perhaps might be an option….
>
>
>
> Thanks,
>
> Aisling.
>
>
>
> -----Original Message-----
> *From:* Voipers Portugal [mailto:voipers@gmail.com]
> *Sent:* 30 January 2006 11:47
> *To:* Aisling
> *Cc:* John Todd; serusers(a)iptel.org; clona(a)cyberhouse.no
> *Subject:* Re: [Serusers] SER RTP QoS Monitoring
>
>
>
> Ethereal does that automatically, so I think you can start there. As for
> RTPproxy, as far as I remember, it does not support RTCP (at least it didn´t
> when I tested a few months ago).
>
>
>
> Jose Simoes
>
>
>
> On 1/30/06, *Aisling* <ashling.odriscoll(a)cit.ie> wrote:
>
>
> Hi,
>
> Thanks to John, Nils and Atle for the replies. So if I am to proceed
> with extracting QoS info from RTCP it is now my understanding that I
> would either have to have my QoS gathering application at:
>
> 1) the media proxy (used primarily for nat situations)
> 2) where the voice is going peer to peer, it would need to be placed at
> the end points. However what I'm wondering about here is would it be
> possible to have a QoS collector application running on another machine
> on the same network as the two endpoints (obviously presuming they're
> all on the same network) which could intercept the sender and receiver
> reports? And if so does anyone have any idea if there's another way to
> distinguish RTCP packets without knowing the IP address or port?
>
> Many thanks,
> Aisling.
>
> -----Original Message-----
> From: John Todd [mailto: jtodd(a)loligo.com]
> Sent: 29 January 2006 22:33
> To: Aisling O'Driscoll; serusers(a)iptel.org
> Subject: Re: [Serusers] SER RTP QoS Monitoring
>
> Hello -
> Your method #1 (RTCP) is the best solution, it would seem. Method
> #2 (INFO) is sort-of supported in some devices, like the Cisco 79xx
> series with SIP software - they support insertion of some basic call
> element data into the BYE message. Method #3 is complex, as it would
> require the SNMP collector to have full knowledge of all call states
> on the devices that were queried (though possibly this is not more
> complex than the RTCP collection process, either.) In any of these
> three cases, SER does not collect the call quality data - the
> endpoints or media proxies are the only places that see this data.
> If you run a mediaproxy in conjunction with SER, then it may be
> possible to collect and report on that data, but there are drawbacks
> to a design of that nature.
>
> For the long term, I would suggest taking a look at this draft,
> which describes the use of RTCP-XR (or "vq-rtcpxr") as set of data to
> be transmitted via SIP messages between media-handling entities and
> other reporting systems:
>
>
> http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary-
> 08.txt
>
> While we are still a while away from seeing vendor (or open source)
> implementations, this seems like a reasonable proposal. It allows
> for SIP-based publication or subscription of call quality events by
> various network elements, and it will permit arbitrary querying of
> this data at any time interval (not just at the end of the call.)
>
> JT
>
>
>
> >Hi,
> >
> >I want to be able to extract QoS parameters such as jitter, delay and
> >packet loss from VoIP calls made using my SER(both peer to peer for
> >my public clients and also for clients where the voice will be routed
> >with mediaproxy). I would like to develop something as opposed to
> >purchasing a software package.
> >
> >>From searching the Internet the last few days I have seen that the
> >following options seem to be available:
> >
> >1) RTCP
> >RTCP stream will contain this information. The problem is that
> >sniffers cannot clearly distinguish RTCP packets from other UDP
> >packets unless something like an IP address or specific UDP port is
> >supplied. I had considered using TCPDUMP but then where do I position
> >this in the network when voice is going peer to peer between client
> >and I'm not sure how this can work to monitor all calls when I might
> >not be aware of the end user IP addresses.
> >
> >2) SIP INFO extension
> >This also seem to be an option but most off the shelf phones don't
> >seem to support this and this would required modification of a SIP ua.
> >
> >3) SNMP
> >I thought maybe SNMP might be an option but the SER snmp module no
> >longer exists...
> >
> >Does anyone have any comments on the above? Are the statements that I
> >made correct or can anyone think of other ways to monitor the voice
> >QoS? I am trying to understand how commercial applications have
> >accomplished this.
> >
> >Many thanks in advance for advice.
> >Aisling.
> >
> >>---- Original Message ----
> >>From: clona(a)cyberhouse.no
> >>To: ashling.odriscoll(a)cit.ie
> >>Subject: Re: [Serusers] Status of SER SNMP module
> >>Date: Sun, 29 Jan 2006 20:30:02 +0100
> >>
> >>>
> >>>Hi Aisling,
> >>>
> >>>The SNMP module is dead as a killed turtle :(
> >>>
> >>>-Atle
> >>>* Aisling O'Driscoll < ashling.odriscoll(a)cit.ie> [060129 20:26]:
> >>>> Hello,
> >>>>
> >>>> I'm just wondering what the current status of the SER SNMP module
> >>is?
> >>>> Is it currently supported and is there somewhere I can find
> >>>> documentation on it?
> >>>>
> >>>> Many thanks,
> > >>> Aisling.
> >
> >_______________________________________________
> >Serusers mailing list
> >Serusers(a)iptel.org
> >http://mail.iptel.org/mailman/listinfo/serusers
>
>
> -------------------Legal
> Disclaimer---------------------------------------
>
> The above electronic mail transmission is confidential and intended only
> for the person to whom it is addressed. Its contents may be protected by
> legal and/or professional privilege. Should it be received by you in
> error please contact the sender at the above quoted email address. Any
> unauthorised form of reproduction of this message is strictly
> prohibited. The Institute does not guarantee the security of any
> information electronically transmitted and is not liable if the
> information contained in this communication is not a proper and complete
> record of the message as transmitted by the sender nor for any delay in
> its receipt.
>
>
> -------------------Legal Disclaimer---------------------------------------
>
>
> The above electronic mail transmission is confidential and intended only
> for the person to whom it is addressed. Its contents may be protected by
> legal and/or professional privilege. Should it be received by you in error
> please contact the sender at the above quoted email address. Any
> unauthorised form of reproduction of this message is strictly prohibited.
> The Institute does not guarantee the security of any information
> electronically transmitted and is not liable if the information contained in
> this communication is not a proper and complete record of the message as
> transmitted by the sender nor for any delay in its receipt.
>
>
> _______________________________________________
> Serusers mailing list
> Serusers(a)iptel.org
> http://mail.iptel.org/mailman/listinfo/serusers
>
>
> -------------------Legal Disclaimer---------------------------------------
> The above electronic mail transmission is confidential and intended only for
> the person to whom it is addressed. Its contents may be protected by legal
> and/or professional privilege. Should it be received by you in error please
> contact the sender at the above quoted email address. Any unauthorised form
> of reproduction of this message is strictly prohibited. The Institute does
> not guarantee the security of any information electronically transmitted and
> is not liable if the information contained in this communication is not a
> proper and complete record of the message as transmitted by the sender nor
> for any delay in its receipt.
> -------------------Legal Disclaimer---------------------------------------
> The above electronic mail transmission is confidential and intended only for
> the person to whom it is addressed. Its contents may be protected by legal
> and/or professional privilege. Should it be received by you in error please
> contact the sender at the above quoted email address. Any unauthorised form
> of reproduction of this message is strictly prohibited. The Institute does
> not guarantee the security of any information electronically transmitted and
> is not liable if the information contained in this communication is not a
> proper and complete record of the message as transmitted by the sender nor
> for any delay in its receipt.
>
Hi Jose,
I looked at ethereal but I dont think its possible to use that in real
time i.e. you have to start the capture and when you think the call is
over stop the capture and examine the results. I would like something in
real-time that makes me think tethereal perhaps might be an option .
Thanks,
Aisling.
-----Original Message-----
From: Voipers Portugal [mailto:voipers@gmail.com]
Sent: 30 January 2006 11:47
To: Aisling
Cc: John Todd; serusers(a)lists.iptel.org; clona(a)cyberhouse.no
Subject: Re: [Serusers] SER RTP QoS Monitoring
Ethereal does that automatically, so I think you can start there. As for
RTPproxy, as far as I remember, it does not support RTCP (at least it
didn´t when I tested a few months ago).
Jose Simoes
On 1/30/06, Aisling <ashling.odriscoll(a)cit.ie> wrote:
Hi,
Thanks to John, Nils and Atle for the replies. So if I am to proceed
with extracting QoS info from RTCP it is now my understanding that I
would either have to have my QoS gathering application at:
1) the media proxy (used primarily for nat situations)
2) where the voice is going peer to peer, it would need to be placed at
the end points. However what I'm wondering about here is would it be
possible to have a QoS collector application running on another machine
on the same network as the two endpoints (obviously presuming they're
all on the same network) which could intercept the sender and receiver
reports? And if so does anyone have any idea if there's another way to
distinguish RTCP packets without knowing the IP address or port?
Many thanks,
Aisling.
-----Original Message-----
From: John Todd [mailto: jtodd(a)loligo.com]
Sent: 29 January 2006 22:33
To: Aisling O'Driscoll; serusers(a)lists.iptel.org
Subject: Re: [Serusers] SER RTP QoS Monitoring
Hello -
Your method #1 (RTCP) is the best solution, it would seem. Method
#2 (INFO) is sort-of supported in some devices, like the Cisco 79xx
series with SIP software - they support insertion of some basic call
element data into the BYE message. Method #3 is complex, as it would
require the SNMP collector to have full knowledge of all call states
on the devices that were queried (though possibly this is not more
complex than the RTCP collection process, either.) In any of these
three cases, SER does not collect the call quality data - the
endpoints or media proxies are the only places that see this data.
If you run a mediaproxy in conjunction with SER, then it may be
possible to collect and report on that data, but there are drawbacks
to a design of that nature.
For the long term, I would suggest taking a look at this draft,
which describes the use of RTCP-XR (or "vq-rtcpxr") as set of data to
be transmitted via SIP messages between media-handling entities and
other reporting systems:
http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary-
08.txt
While we are still a while away from seeing vendor (or open source)
implementations, this seems like a reasonable proposal. It allows
for SIP-based publication or subscription of call quality events by
various network elements, and it will permit arbitrary querying of
this data at any time interval (not just at the end of the call.)
JT
>Hi,
>
>I want to be able to extract QoS parameters such as jitter, delay and
>packet loss from VoIP calls made using my SER(both peer to peer for
>my public clients and also for clients where the voice will be routed
>with mediaproxy). I would like to develop something as opposed to
>purchasing a software package.
>
>>From searching the Internet the last few days I have seen that the
>following options seem to be available:
>
>1) RTCP
>RTCP stream will contain this information. The problem is that
>sniffers cannot clearly distinguish RTCP packets from other UDP
>packets unless something like an IP address or specific UDP port is
>supplied. I had considered using TCPDUMP but then where do I position
>this in the network when voice is going peer to peer between client
>and I'm not sure how this can work to monitor all calls when I might
>not be aware of the end user IP addresses.
>
>2) SIP INFO extension
>This also seem to be an option but most off the shelf phones don't
>seem to support this and this would required modification of a SIP ua.
>
>3) SNMP
>I thought maybe SNMP might be an option but the SER snmp module no
>longer exists...
>
>Does anyone have any comments on the above? Are the statements that I
>made correct or can anyone think of other ways to monitor the voice
>QoS? I am trying to understand how commercial applications have
>accomplished this.
>
>Many thanks in advance for advice.
>Aisling.
>
>>---- Original Message ----
>>From: clona(a)cyberhouse.no
>>To: ashling.odriscoll(a)cit.ie
>>Subject: Re: [Serusers] Status of SER SNMP module
>>Date: Sun, 29 Jan 2006 20:30:02 +0100
>>
>>>
>>>Hi Aisling,
>>>
>>>The SNMP module is dead as a killed turtle :(
>>>
>>>-Atle
>>>* Aisling O'Driscoll < <mailto:ashling.odriscoll@cit.ie>
ashling.odriscoll(a)cit.ie> [060129 20:26]:
>>>> Hello,
>>>>
>>>> I'm just wondering what the current status of the SER SNMP module
>>is?
>>>> Is it currently supported and is there somewhere I can find
>>>> documentation on it?
>>>>
>>>> Many thanks,
> >>> Aisling.
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
-------------------Legal
Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only
for the person to whom it is addressed. Its contents may be protected by
legal and/or professional privilege. Should it be received by you in
error please contact the sender at the above quoted email address. Any
unauthorised form of reproduction of this message is strictly
prohibited. The Institute does not guarantee the security of any
information electronically transmitted and is not liable if the
information contained in this communication is not a proper and complete
record of the message as transmitted by the sender nor for any delay in
its receipt.
-------------------Legal
Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only
for the person to whom it is addressed. Its contents may be protected by
legal and/or professional privilege. Should it be received by you in
error please contact the sender at the above quoted email address. Any
unauthorised form of reproduction of this message is strictly
prohibited. The Institute does not guarantee the security of any
information electronically transmitted and is not liable if the
information contained in this communication is not a proper and complete
record of the message as transmitted by the sender nor for any delay in
its receipt.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
-------------------Legal
Disclaimer--------------------------------------- The above electronic
mail transmission is confidential and intended only for the person to
whom it is addressed. Its contents may be protected by legal and/or
professional privilege. Should it be received by you in error please
contact the sender at the above quoted email address. Any unauthorised
form of reproduction of this message is strictly prohibited. The
Institute does not guarantee the security of any information
electronically transmitted and is not liable if the information
contained in this communication is not a proper and complete record of
the message as transmitted by the sender nor for any delay in its
receipt.
-------------------Legal Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.
Hello!
Two cases of beer or two bottles of whiskey to first developer that release
a simple snmp module that provide equal status equal to the info accessible
through the fifo.
Deadline for first sneak preview is next Friday! :-)
-
Cheers
Thomas
Hi all,
I wish to set ser (outbound proxy) with failover
support :
SER1
sip agents ======= ======= Asterisk Farm
==sip/pstn gateway
SER2
SER1 and SER2 use SQL DB if one of the SERs is down
how can we get the other available for the sip agents
for example we need to reboot a SER when the ser.cfg
has changed .
I look at dispatcher module !
Regards
Harry
___________________________________________________________________________
Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international.
Téléchargez sur http://fr.messenger.yahoo.com
I was able to successfully install SER and I can make calls. But I have two
problems with it, they are
1) Even unregistered Users are able to call out.
2) We have few computers connected via LAN and we are connected to a router. We
have a UNIX box somewhere else on which we installed SER. My problem is from
computers which have indiviual IP are able to make and receive calls but in my
internal network we are having one way problems or sometimes we are unable to
make or recieve calls.
I am attaching my ser.cfg, please check it and help me to solve my problems.
Thanks,
Manoj.
