Hi
iam runing SER
and my client is registering with Asterisk, running call center
he keep getting the following error
chan_sip.c: Forbidden - wrong password on authentication for INVITE to
'"M0601001458000013456" <sip:user@domain.com>;tag=as5bbc2eec'
and voice going choppy, when he call from single user the voice go clear
when he intiates the calls from number of people voice is very choppy
and some time not able to hear
all are public IP, not NAT here
any suggestions
ram
Hi Users,
I'm setup SERWeb 0.9.6 with Openser 1.2.1.
At first, when login with correct username/password, there is a "sorry --
cannot open write fifo" on the my_account.php.
Referred from google, I changed the access right of openser_fifo to 666;
that cause the my_account.php does not show.
Please kindly advise, thanks in advance.
Regards,
Lesley
Hi All,
A test case with SUBSCRIBE winfo event message have reached values of
hundreds of messages per second in a load test scenario.
But a test case with SUBSCRIBE presence event message have only reached
values of 20 messages per second.
Is there any explainable reason for this big difference between the
results of both test cases?
Thanks,
Toni
Okay, thanks.
- Jeremy
>-----Original Message-----
>From: Ovidiu Sas [mailto:sip.nslu@gmail.com]
>Sent: Thursday, May 31, 2007 10:42 AM
>To: 'Jeremy George'
>Cc: users(a)openser.org
>Subject: Re: [Users] adding users
>
>Hi Jeremy,
>
>Have you installed the serweb tables?
>If yes, you have to set HAS_SERWEB="yes" in your openserctlrc. Check
>the mailing list archive, this issue was already discusses a few
>times.
>
>
>Regards,
>Ovidiu Sas
>
>On 5/31/07, Jeremy George <jeremy(a)georgeco.us> wrote:
>>
>> I upgraded to 1.2.1 and completely re-created the db. Now I can add
>> exactly one user to the db. Also, trying to delete that one user from
>> openserctl hangs. Thoughts?
>>
>> Thanks much.
>>
>> - Jeremy
>>
>> Breathless-FC6# openserctl add jeremy xxxx murph(a)the.surf
>> MySql password for user 'openser@localhost':
>> new user 'jeremy' added
>> Breathless-FC6# openserctl add helen xxxx murph(a)the.surf
>> MySql password for user 'openser@localhost':
>> ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3
>> ERROR: introducing the new user 'helen' to the database failed
>> Breathless-FC6#
>> Breathless-FC6# openserctl rm jeremy
>>
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>>
>
I upgraded to 1.2.1 and completely re-created the db. Now I can add
exactly one user to the db. Also, trying to delete that one user from
openserctl hangs. Thoughts?
Thanks much.
- Jeremy
Breathless-FC6# openserctl add jeremy xxxx murph(a)the.surf
MySql password for user 'openser@localhost':
new user 'jeremy' added
Breathless-FC6# openserctl add helen xxxx murph(a)the.surf
MySql password for user 'openser@localhost':
ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3
ERROR: introducing the new user 'helen' to the database failed
Breathless-FC6#
Breathless-FC6# openserctl rm jeremy
Hi to all.
I've configured my polycom ip500 IPphone to use it with ser.
If I try a call to users registred to ser all works fine.
If I try to call PSTN Number through my gateway I've the follow sip message:
INVITE sip:0672028405@10.28.19.202:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD.
From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84.
To: <sip:0672028405@10.28.19.202;user=phone>.
CSeq: 2 INVITE.
Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143.
Contact: <sip:0660522015@10.28.19.143>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1.
Supported: 100rel,replace.
Allow-Events: talk,hold,conference.
Proxy-Authorization: Digest username="0660522015",
realm="10.28.19.202",
nonce="465ef44bbb09c0155dc8555314519a20cac896f8",
uri="sip:0672028405@10.28.19.202:5060;user=phone",
response="e6eed258e095ee6be5be6c92210f9d99", algorithm=MD5.
Max-Forwards: 70.
Content-Type: application/sdp.
Content-Length: 237.
.
v=0.
o=- 1180620549 1180620549 IN IP4 10.28.19.143.
s=Polycom IP Phone.
c=IN IP4 10.28.19.143.
t=0 0.
m=audio 2234 RTP/AVP 18 8 0 101.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
#
U 2007/05/31 16:09:03.527834 10.28.19.202:5060 -> 10.28.19.143:5060
SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL).
Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD.
From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84.
To: <sip:0672028405@10.28.19.202;user=phone>;tag=979d95a734c13f6db8b9e3a72b9f44a0.14e7.
CSeq: 2 INVITE.
Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143.
Server: Sip EXpress router (0.9.6 (i386/linux)).
Content-Length: 0.
Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=8314
req_src_ip=10.28.19.143 req_src_port=5060
in_uri=sip:0672028405@10.28.19.202:5060;user=phone
out_uri=sip:0672028405@10.28.52.105:5060:5060;user=phone via_cnt==1".
Have you any suggestion about?
I use my Polycom with Asterisk and BroadSoft without any problem.
Thanks for your support and great patience :D
Bye,
F.
Hello all,
I am having trouble getting openserct ping to work (version 1.2). The
server is running but I get
[1]+ Done cat <$path | filter_fl
400
I modified openserct at the options_ping() function to do
echo $RET
and I get
[1]+ Done cat <$path | filter_fl
400 Too few or too many arguments
I tried playing around with the arguments according to the documentation
for tm's function t_uac_dlg, but no luck. Can someone help?
thank you
George
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Does any one know what modules(and corresponding functions if possible) that I need to use in order to make the SER do the Redirection by initiating a 302( after the reception of INVITE) that contains the contact info/uri that the caller will need to use to call the forwarded-to client?
Can this be achieved without the need to use CPL language?
Thanks in advance.
Regards,
Cyp.
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Hi,
is there a way in openser to get it work like this:
1. SIP-Request comes in
2. openser detects a special service for dialed number
3. openser sets that original sip session on hold and forwards the
request to a remote service node
4. remote service node does it job and sends back a result message via
network to openser
5. openser checks that result, get the session from hold and continues
it's work on the session
In the end I 'm looking for a way to let openser handles a sip session
like a workflow, where I can inject some value added services by REMOTE
service nodes (not by modules or plugins) without forwarding the
complete session to a remote service node.
I hope u understand what I mean.
regards
Helmut
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