Hello.
I read about this parameter in the registrar module on :
http://lists.iptel.org/pipermail/serusers/2004-November/013373.html
I would like to know if there is any update about the new SER
release.
Is too dagerous use the CVS version for a "production" enviroment?
Thanks in advance,
Best regards
Ricardo Martinez.-
Yes, yes, and maybe.
> I would like to know if SER can notify by anyway to other application of the start and end of
> VoIP transmissions in order to maintain a control in real-time of calls made by the users.
This one maybe a little more tricky, I'm looking for such an app myself. I suppose this could be done via the acc module.
-----Original Message-----
From: Alberto Martínez [mailto:amartinez@astrasoft.es]
Sent: Thursday, December 16, 2004 9:21 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Does SER do that?
-----BEGIN PGP SIGNED MESSAGE-----
Hash: MD5
Hello,
I am new in SER. I would like to ask you if SER is able to do somethings I am looking for. I would like it to check if the users who try to connect with it by SIP are correct, checking the login info and, if they are, redirect the connection to the VoIP provider. If the user who is trying to connect is not set up in SER it must be rejected.
I would like to know if SER can notify by anyway to other application of the start and end of VoIP transmissions in order to maintain a control in real-time of calls made by the users.
Thank you.
Best regards,
Alberto
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_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hello,
I would like to restrict registrations to use username, auth. username
and password fields all together.
My problem:
If the user sets in his device the auth. username and password, it's
enough for registration. However he can set whatever username, which I
don't want to allow, while I would like to use numbers and don't want
that anybody misuse this.
Any idea how to do that?
Kind regards,
Tamas
I have browsed the mailing lists for information
relating to conferencing and I still do not have any
authoritative answers to the following questions:
1) I realize that the SEMS has a conferencing module -
But from what I understand, each user who wants to
participate in a conference would be "dialing" a
specific conferencing URI and they would all be
bridged and conferencing can take place. Is this
correct?
2) A convenient and more realistic scenario is when I
am on a call with someone and I'd like to conference
in someone. Normally, I would put the first user on
hold, Dial a key sequence followed by the phone
number of the person I'd like to conference in. Once
the person picks up, I can bring the other person
on-line and the conference can take place.
What does one require to make this happen? From what I
read, the User Agents have to support this
functionality (mixing?). Can this be done via an ATA
or do I need IP Phones? Which ATA/IP Phones supports
this? Secondly, I assume I'd still require SEMS to act
as a media server? Is this correct?
Finally, has someone ever implemented such a solution
and would someone be willing to share some thought
about it.
Thanks in advance
__________________________________
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
-----BEGIN PGP SIGNED MESSAGE-----
Hash: MD5
Hello,
I am new in SER. I would like to ask you if SER is able to do
somethings I am looking for. I would like it to check if the users who
try to connect with it by SIP are correct, checking the login info
and, if they are, redirect the connection to the VoIP provider. If the
user who is trying to connect is not set up in SER it must be
rejected.
I would like to know if SER can notify by anyway to other application
of the start and end of VoIP transmissions in order to maintain a
control in real-time of calls made by the users.
Thank you.
Best regards,
Alberto
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Hello,
I have two SERs: one on local subnet (192.168.1.x), the other remote. I also
have two exact phones: one with local SER, the other is on the same subnet
with remote SER. Both of them can register correctly to the SER on the same
subnet. However, my local IPphone fails to register to the remote SER. I am
pretty sure the username and password are valid. To verify the username
setup, I have xlite installed on the local network 192.168.1, it can
register to both local and remote server without any problem by the same
account.
I notice that the successful registration always has port information in the
header:
Via: SIP/2.0/UDP
192.168.1.103:5060;rport=5060;branch=z9hG4bK8F05E1B1000941AC83B8205023AD0B8F
The failed registration doesn't:
Via: SIP/2.0/UDP
192.168.1.102;branch=z9hG4bKc0f9cc1144b12ca2;received=24.22.117.57
Can someone shed some light on it?
thanks!
steven
I have SER version 0.8.14 running as a proxy, registrar, and location
server. It works great, but I have encountered one problem. When I
call from my PingTel SIP phone to an IP enabled IVR, and the IVR
responds with "404 Not found", SER sends an ACK to the IVR and forwards
the 404 to the PingTel phone. All that is fine, but when the PingTel
phone sends back its ACK, SER does not find the transaction to associate
it with. It forwards the ACK to the IVR, then a timer fires and the 404
is resent to the PingTel because SER does not think it has gotten the
ACK. I have determined that the problem is in modules/tm/t_lookup.c,
around line 450, where it says "if (! EQ_LEN(from)) continue;"
The original INVITE contained a From value of
"sip:7216@wic.west.com;tag=1c8140"
While the ACK contains a From value of
"<sip:7216@wic.west.com>;tag=1c8140"
According to RFC3261, Section 20.20, these should be considered
identical, but SER does not see it that way because the angle brackets
make the length different (the SIP stack on the IVR adds them in its
responses). I have only seen this behavior when a failure response code
is returned, if I get a "200 Ok", everything works fine.
I've tried attaching a log of the SIP messages recorded by the PingTel
phone, and the log from SER, but the list seems to have a problem with
attachments.
Dave Wearne
I have SER version 0.8.14 running as a proxy, registrar, and location
server. It works great, but I have encountered one problem. When I
call from my PingTel SIP phone to an IP enabled IVR, and the IVR
responds with "404 Not found", SER sends an ACK to the IVR and forwards
the 404 to the PingTel phone. All that is fine, but when the PingTel
phone sends back its ACK, SER does not find the transaction to associate
it with. It forwards the ACK to the IVR, then a timer fires and the 404
is resent to the PingTel because SER does not think it has gotten the
ACK. I have determined that the problem is in modules/tm/t_lookup.c,
around line 450, where it says "if (! EQ_LEN(from)) continue;"
The original INVITE contained a From value of
"sip:7216@wic.west.com;tag=1c8140"
While the ACK contains a From value of
"<sip:7216@wic.west.com>;tag=1c8140"
According to RFC3261, Section 20.20, these should be considered
identical, but SER does not see it that way because the angle brackets
make the length different (the SIP stack on the IVR adds them in its
responses). I have only seen this behavior when a failure response code
is returned, if I get a "200 Ok", everything works fine.
I've attached a log of the SIP messages recorded by the PingTel phone,
and the log from SER.
Dave Wearne
<<pingtelcall.log.gz>> <<ser.log.gz>>
Hello!
How can I restrict unregistered clients to make calls through my SER
proxy? And how can I enable some unregistered endpoints (like incoming
PSTN gateway)?
I'm still newbie in this field :(
Thanks in advance,
Tamas
Hi,
I have encountered the following error messages in /var/log/messages
using ser-0.8.14.
Dec 16 14:26:35 /usr/local/sbin/ser[4788]: ERROR: open_reply_pipe: open
error (/tmp/ser_receiver_5067): No such file or directory
Dec 16 14:26:35 /usr/local/sbin/ser[4788]: ERROR: fifo_reply: no reply
pipe /tmp/ser_receiver_5067
Dec 16 14:33:27 /usr/local/sbin/ser[4788]: ERROR: open_reply_pipe: open
error (/tmp/ser_receiver_5084): No such file or directory
Dec 16 14:33:27 /usr/local/sbin/ser[4788]: ERROR: fifo_reply: no reply
pipe /tmp/ser_receiver_5084
What is this kind of problem? But I can run ser as usual.
Thomas
