Hi,
Is anyone aware of any SIP user agent that can record a SIP call and store
it to a .wav file?
Is it possible to configure the SIP Express Router to record the SIP voice
call and save it to a file rather than sending it to the other end?
If yes, please let me know the details.
Thanks,
-Darshan.
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Quick Question. Can I put SER onto multiple servers with load balancing and share the userloc tables between the servers (ie. With MySQL or Postgres SQL)
Is this possible? Or does one SER instance only recognize registrations for phones that register with that particular instance/daemon?
Thanks!
Darren Nay - dnay(a)libertyisp.com
At 07:39 PM 8/14/2003, Chad Brown wrote:
>Perfect,
>
>Let me ask 2 quick follow-on questions...
>
>1. Can I go back to http://www.iptel.org/ser/tarball/ser_8_11_stable.tgz
>to get the latest patched and STABLE builds?
That's the latest, most stable source, you need to compile it myself.
There will be a new complete distribution by end of this month -- we are now
waiting to run SER through the upcoming SIPIT to release it.
>2. What location / version of the modules should I use when running
>builds for this location? (Serweb, mysql, jabber, etc)
All SER modules are included there.
uptodate SERWEB is now in the http://www.iptel.org/ser/tarball/ directory too.
-jiri
hello,
i am testing SER.
But i have a problem:
Am firt i use serctl alias 12345 sip:chen@testdomain:
It shows:
200 Added to table
('12345', 'sip:chen@testdomain) to 'aliases'
But i use then serctl alias show 12345:
It shows:
404 Username 12345 in table alises not found.
i have checked also mysql database. There is no my record in
table "aliases".
Why?
regards
Chen
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Hi !
After specific time I redirect (revert_uri and append_branch)
call to another sip address. Everythig is ok for UA like ATA, Kphone
and C7960). When the call is started from Grandstream
after the pick up second site (Asterisk IVR- after redirection),
connection is terminated afer a few seconds.
This situations takes place (only for GS ) also when I redirect calls from
one sip domain to another depends on prefix call (for client
doesn't support URL sip addresses like GS, ATA)
In logs ACK message directed to ser I see differences between UA.
Originated destination sip address is 3000, when no answer, call is
redirected to 4000
For ATA I have:
192.168.0.83:5060 -> 192.168.0.1:5060
ACK sip:3000@192.168.0.1 SIP/2.0..
Route: <sip:4000@192.168.0.81;branch=0>,<sip:4000@192.168.0.1:6060>..
Via: SIP/2.0/UDP 192.168.0.83:5060..
From: radan <sip:3100@sip.router.pl;user=phone>;tag=3207317092..
To: <sip:3000@sip.router.pl;user=phone>;tag=as2d60db53..
Call-ID: 3934861712(a)192.168.0.83..
CSeq: 1 ACK..
User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A)..
Content-Length:
0....
For GS I have:
192.168.0.84:5060 -> 192.168.0.1:5060
ACK sip:4000@192.168.0.1:6060 SIP/2.0..
Via: SIP/2.0/UDP 192.168.0.84..
Route:<sip:3000@192.168.0.1;ftag=f6e3b058-8afd-fac2-e60b-e493a7d83844;lr>..
Route: <sip:4000@192.168.0.81;branch=0>..
From: "radan - grandstream" <sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844..
To: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868..
Contact:<sip:3102@192.168.0.84;user=phone>..
Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84..
CSeq: 65090 ACK..
User-Agent: Grandstream SIP UA 1.0.3.81..
Max-Forwards: 70..
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE..
Content-Length: 0....
Two different calls are confirmed.
for GS I have then following info a few times (5 or 6)
192.168.0.1:5060 -> 192.168.0.84:5060
SIP/2.0 200 OK..
Via: SIP/2.0/UDP 192.168.0.84..
Record-Route: <sip:4000@192.168.0.81;branch=0>..
Record-Route: <sip:3000@192.168.0.1;ftag=f6e3b058-8afd-fac2-e60b-e493a7d83844;lr>..
From: "radan - grandstream" <sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844..
To: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868..
Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84..
CSeq: 65090 INVITE..
User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact:
<sip:4000@192.168.0.1:6060>..
Content-Type: application/sdp..
Probably GS is not able to send ACK
After them the ser sends BYE to the GS
192.168.0.1:5060 -> 192.168.0.84:5060
BYE sip:3102@192.168.0.84;user=phone SIP/2.0..
Record-Route: <sip:3000@192.168.0.1;ftag=as18e54868;lr>..
Max-Forwards: 9 ..
Via: SIP/2.0/UDP 192.168.0.1;branch=z9hG4bK2743.08055687.0..
Via: SIP/2.0/UDP 192.168.0.81;branch=z9hG4bKcc8e.cc7088e2.0..
Via: SIP/2.0/UDP 192.168.0.1:6060;branch=z9hG4bK0ff02add..
From: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868..
To: "radan - grandstream" <sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844..
Contact: <sip:4000@192.168.0.1:6060>..
Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84..
CSeq: 102 BYE..User-Agent: Asterisk PBX
Content-Length: 0....
a GS talks that this connection doesn't exist
192.168.0.84:5060 -> 192.168.0.1:5060
SIP/2.0 481 ..
Via: SIP/2.0/UDP 192.168.0.1;branch=z9hG4bK2743.08055687.0..
Via: SIP/2.0/UDP 192.168.0.81;branch=z9hG4bKcc8e.cc7088e2.0..
Via: SIP/2.0/UDP 192.168.0.1:6060;branch=z9hG4bK0ff02add..
Record-Route: <sip:3000@192.168.0.1;ftag=as18e54868;lr>..
From: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868..
To: "radan - grandstream" <sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844..
Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84..
CSeq: 102 BYE..
User-Agent: Grandstream SIP UA 1.0.3.81..
Content-Length: 0....
It is a some bug in soft for GS, or do I have to
add something special in configuration file for GS ?
Thanks
Andrzej
Hi,
I had ser-0.8-11 on linux 8.0. I downloaded the 0.8.12 version. When i
run the rpm pakage i get the following error:
file /usr/lib/ser/modules/*.so from install of ser-0.8.12-0 conflicts with
file from package ser-0.8.11-0.
I had deleted all the ser-0.8.11-0 files before i did this installation.
Where is this error coming from?
Also, i want to work with the dbtext module which needs that i work with
the source module only but when i do a rpm -i the installation is done.
What is the way around this? The command ./ser -f ser.cfg comes up with
errors, where can i view these errors?
Any pointers will be of great help.
Thanks,
Annie
Hi,
has anyone an example config for using ser with sems and the isdn4linux
package to use ser as SIP<>ISDN GW for outgoing and maybe routing incoming
calls?
best regards,
Arnd
Hi
i've downloaded lastest tarball but when I start ser after compiling it
gave me segmentation fault.
any idea?
andrea
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