Hi all,
What do you recommend (and are using) as billing software freeware
and/or shareware that interacts with SER (and, if possible, with GnuGK)?
Thanks,
Marc
Just restart SER will do.
Eg.
Serctl restart
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of rupesh
Sent: Thursday, August 11, 2005 2:24 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] compilation of ser.cfg
hello ,
i am new to SER. i have made some modification in the ser.cfg
file in ser 0.9.3 version . now i want to compile this ser.cfg file so
that the changes which i have made can effect the ser, but am unable to
compile. can someone help me with this. can u tell me the compilation
steps.i have checked in the getting started document and the install
document from iptel also, but of no use. waiting for someone to help me
out. thank you,
Rupesh
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
hello ,
i am new to SER. i have made some modification in the ser.cfg file in ser 0.9.3 version . now i want to compile this ser.cfg file so that the changes which i have made can effect the ser, but am unable to compile. can someone help me with this. can u tell me the compilation steps.i have checked in the getting started document and the install document from iptel also, but of no use. waiting for someone to help me out. thank you,
Rupesh
Hi List
I am using an AudioCodes MP-104 FXS with Version ID: 4.20.354.571.
Authentication details: username/password 101/101. In End Point's Phone
Number field i am using 102 instead of 101.
The problem is....When REGISTER comes, it comes from 102 whereas it should
come from 101. Second is When SER asks for realm and other details A/C is
sending details of 101 in realm.
U 192.168.1.2:5060 -> 192.168.1.1:5060 REGISTER sip:192.168.1.1
SIP/2.0..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKacgsnejnS..From:
<sip:102@192.168.1.1>;tag=1c66110..To: <sip:102@192.168.1.1>..Call-ID :
478522552255UyxO@192.168.1.2..CSeq: 90119 REGISTER..Expires: 60..Contact:
<sip:102@192.168.1.2;user=phone>;expires=60..Content-Length: 0....#
U 192.168.1.1:5060 -> 192.168.1.2:5060 SIP/2.0 401 Unauthorized..Via:
SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKacgsnejnS;rport=5060..From:
<sip:102@192.168.1.1>;tag=1c66110..To:
<sip:102@192.168.1.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7aba..Call-ID:
478522552255UyxO@192.168.1.2..CSeq: 90119 REGISTER..WWW-Authenticate: Digest
realm="192.168.1.1",
nonce="42fad87e8fc59c528db3b626c769c8e55a3f4454"..Server: Sip EXpress router
(0.10.99-dev14-tcp (i386/linux))..Content-Length: 0..Warning: 392
192.168.1.1:5060 "Noisy feedback tells: pid=17382 req_src_ip=192.168.1.2
req_src_port=5060 in_uri=sip:192.168.1.1 out_uri=sip:192.168.1.1
via_cnt==1"....#
U 192.168.1.2:5060 -> 192.168.1.1:5060 REGISTER sip:192.168.1.1
SIP/2.0..Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKacbcphONy..From:
<sip:102@192.168.1.1>;tag=1c66110..To: <sip:102@192.168.1.1>..Call-ID :
478522552255UyxO@192.168.1.2..CSeq: 90120 REGISTER..Contact:
<sip:102@192.168.1.2;user=phone>;expires=60..Authorization:Digest
username="101",realm="192.168.1.1",nonce="42fad87e8fc59c528db3b626c769c8e55a3f4454",uri="sip:192.168.1.1",Algorithm="MD5",response="14e3a3d63687571fc152c7d2487d0314"..Max-Forwards:
70..Expires: 60..User-Agent: Audiocodes-Sip-Gateway-MP-104
FXS/v.4.20.354.571..Content-Length: 0 ....#
U 192.168.1.1:5060 -> 192.168.1.2:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.1.2;branch=z9hG4bKacbcphONy;rport=5060..From:
<sip:102@192.168.1.1>;tag=1c66110..To:
<sip:102@192.168.1.1>;tag=b27e1a1d33761e85846fc98f5f3a7e58.e20d..Call-ID:
478522552255UyxO@192.168.1.2..CSeq: 90120 REGISTER..Contact:
<sip:102@192.168.1.2:5060;user=phone>;expires=60..Server: Sip EXpress router
(0.10.99-dev14-tcp (i386/linux))..Content-Length: 0..Warning:
392192.168.1.1:5060 "Noisy feedback tells: pid=17378 req_src_ip=192.168.1.2
req_src_port=5060 in_uri=sip:192.168.1.1 out_uri=sip:192.168.1.1
via_cnt==1"....
Any idea?
_________________________________________________________________
Aamir Khan is back! http://server1.msn.co.in/SP05/Mangalpandey/index.html
See him in the mustached avatar in Mangal Pandey.
Question is ... How do u find out the current value of the rport in the
via header ?
-----Original Message-----
From: ranveer kunal [mailto:ranveerkunal@gmail.com]
Sent: Wednesday, August 10, 2005 10:09 PM
To: Sam Lee
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] ser : rport
I am sorry , i am a bit new to ser. Please tell me more elaborately, how
to make sure that ser attaches rport to the via header.
thanks,
ranveer
On 8/10/05, Sam Lee <sam.lee(a)super.net.sg> wrote:
> If it is exposed as a environmental variable, you can do a export <env
> variable> = something.
> This can be used in conjunction with the exec_msg in the ser.cfg
>
> -----Original Message-----
> From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
> On Behalf Of ranveer kunal
> Sent: Wednesday, August 10, 2005 6:42 PM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] ser : rport
>
> hi all,
> I am using ser, i want ser to add an rport to the via header,
> is this possible?
>
> Thanks
> Ranveer.
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
--
Memories : They bring Diamonds and Rust
Hi all,
I'm new to SER and was reading the admin guide and came across this:
"With a $3000 dual-CPU PC, the SIP Express Router is able to power IP
telephony services in an area as large as the Bay Area during peak
hours."
My question about this, is what is this measuring? For example, is it
only measuring SIP registrations? Maybe the same SER server is
connected to a few IPTSPs and SER is acting as both a registrar and a
proxy server? May be all of the above, while still allowing every UA
registered in the server to talk to one another? Maybe it's also
providing some more advanced features like voicemail to all those Bay
Area users?
Can someone please clarify this for me?
Thanks,
Waldo
Hi.
Reading Gettingstarted04.pdf at 1.6.2 from Onsip.org,
seems to me that I _always_ need a rtp relay when
both UA are nated.
Because NAT will not be able to know what ports
must be forwarded to the internal UA.
Is this true?
Is there any scenario where I can have both
UA nated and they can just use SIP signaling
with ser, and stablish the RTP flow between them?
Thanks in advance,
Sebastian.
(Oringinaly sent to the Sems list, thought it was relevant here as well.)
So, I'm trying to build an achitecture that acceps calls from a
third-party SIP PSTN gateway and runs them through a SEMS
auto-attendant, then out to either local users or back to the PSTN
gateway.
The gateway won't support a REFER.
Also, if I issue a REFER back to the PSTN, my SER will now be out of
the loop, accounting-wise, right?
In addition, I've got some issues with getting ivr.redirect() to work
at all. (Other thread.)
So, what is the smart road ahead here? If I need a B2BUA, can anyone
recommend a good solution?
Thanks,
A.
--
Adam Sherman
Technologist