31 Oct
2005
31 Oct
'05
11:23 a.m.
Hi all,
I am working with asterisk to active voicemail for ser.
I am newbie for it.
So can you give me an example of ser.cfg for do that?
Thank you very much for your instruction.
31 Oct
31 Oct
8:08 p.m.
Julien,
Have a look at the call forwarding example ser.cfg from onsip.org. I
just added the IP address of my Asterisk box to the user_preferences
table and it worked without problems.
- Daryl
On 10/31/05, julien bos <julientoseruser(a)gmail.com> wrote:
Hi all,
I am working with asterisk to active voicemail for ser.
I am newbie for it.
So can you give me an example of ser.cfg for do that?
Thank you very much for your instruction.
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serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
1 Nov
1 Nov
6:52 a.m.
Daryl Sanders writes:
Have a look at the call forwarding example ser.cfg
from onsip.org. I
just added the IP address of my Asterisk box to the user_preferences
table and it worked without problems.
one on my users had tried to make asterisk to register to ser and it
turned out that asterisk included tag into to header of register request
that it send after receiving 401. how can such a software can ever be
made to work with any standards compliant sip proxy?
-- juha
5 Nov
5 Nov
10:50 p.m.
Hi,
I have one question.
I intend to use asterisk and ser in the same server.
I use Ser like a SIP server, while Asterisk is only a voicemail.
After SIPphone register with SER, if the callee don't reply
after few seconds. I forward the call to Asterisk voicemail.
I use some function in avpops.so to make the forward, example:
+avp_db_load: to look in table preference the user who i set
"fwdonbusy" or "fwdnoanswer"
+avp_check
+avp_pushto: to push AVP into SIP message, i can
foward the message to Asterisk.
The problem is that if i can't change the port of message to
the Asterisk. So if i use SER+Asterisk in the same server.
Both will use port 5060, then it can't run at all.
Do you have some solutions for this problem? Thank you so
much.
P/s: if i use "forward", i will miss the problem with NAT or
after that Asterisk can't talk directly to SIPphone.
On 11/1/05, Juha Heinanen <jh(a)tutpro.com> wrote:
Daryl Sanders writes:
Have a look at the call forwarding example
ser.cfg from onsip.org<http://onsip.org>.
I
just added the IP address of my Asterisk box to
the user_preferences
table and it worked without problems.
11:10 p.m.
On 10/31/05, Juha Heinanen <jh(a)tutpro.com> wrote:
Daryl Sanders writes:
In very limited capacity for certain. I don't have asterisk
registering w/SER so I've not experienced this, but I can see how it
would definitely be a problem. From what I've read regarding asterisk,
its SIP channel is the least developed, but I understand there are
improvements on the way.
- Daryl
Have a look at the call forwarding example
ser.cfg from onsip.org. I
just added the IP address of my Asterisk box to the user_preferences
table and it worked without problems.
one on my users had tried to make asterisk to register to ser and it
turned out that asterisk included tag into to header of register request
that it send after receiving 401. how can such a software can ever be
made to work with any standards compliant sip proxy?
-- juha
6758
days inactive
6763
days old
4 comments
3 participants
participants (3)
-
Daryl Sanders -
Juha Heinanen -
julien bos