29 Apr
2008
29 Apr
'08
11:58 a.m.
Sorry guys, I missed one important point. The call initiating asterisk
machine is using a public IP and the phone is in a private subnet as
well as the SER server. So the entire constellation is like this
phone------------->asterisk--------->SER------->asterisk---------------->phone
192.168.9.14->82.98.89.134->10.4.1.80->192.168.13.102->192.168.9.15
I know it is completely weird, but this is only a test case, it isn't
supposed to look like this in the final state.
The other way around when coming from asterisk with a private IP it
works. I tested this already.
Do you know what options I've to set to send the ringback tone from SER
to asterisk which has the public IP?
Thanks
Thorsten
Thorsten wrote:
Hi samuel,
I've already figured out the caller id issue. It was a mis configured
asterisk. I set the caller id 1000 which isn't a international format.
When I set it to
callerid="Thorsten" <6965006100>"
in sip.conf it works just fine.
But I still have that ringback tone issue.
Here comes the asterisk configuration. I've created an account on
asterisk to connect my Snom phone to it
[1000]
type=friend
username=1000
secret=mypass
regexten=1000
host=dynamic
context=toser
callerid="Thorsten" <6965006577>
qualify=yes
nat=yes
The context in extensions.conf looks like this
[toser]
exten => _X.,1,Dial(sip/${EXTEN}(a)10.4.1.80)
I've also set the "progressinband" to "yes".
When I make a call between the asterisk machines not going through SER
it works. So I guess it is a SER issue. In the SER logs I don't see
where it sends a proper 180 message, but I see it on the asterisk machine.
So I don't know if "sl_send_reply" works.
Thanks
Thorsten
samuel wrote:
Most probably your SER instance does not modify
the callerid info so I
would check both your asterisk configs and the configuration of your UAs.
The ringback tone also looks like a configuration issue of your asterisk.
I would recommend you to get some info about the asterisk
configuration to know which the problem might be.
Sam.
2008/4/29, Thorsten <serusers(a)thorko.de <mailto:serusers@thorko.de>>:
Hi guys,
I'm trying to set up a SER server between 2 asterisk machines. I run
into 2 issues.
Whenever I call someone I don't get any ringback tone even so the call
initiating asterisk machine gets the 180 message after 100.
<--- SIP read from 10.4.1.80:5060 <http://10.4.1.80:5060> --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060
From: "Thorsten" <sip:1000@82.98.89.134
<mailto:sip%3A1000@82.98.89.134>>;tag=as4c964973
To: <sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80>>
Call-ID: 5e209fbb7ebdbad97f0193515c5a2982(a)82.98.89.134
<mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134>
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.7 (i386/linux))
Content-Length: 0
Warning: 392 10.4.1.80:5060 <http://10.4.1.80:5060> "Noisy
feedback tells: pid=459
req_src_ip=82.98.89.134 <http://82.98.89.134> req_src_port=5060
in_uri=sip:017683035400@10.4.1.80
<mailto:sip%3A017683035400@10.4.1.80>
out_uri=sip:017683035400@192.168.13.102:5060
<http://sip:017683035400@192.168.13.102:5060> via_cnt==1"
<------------->
--- (9 headers 0 lines) ---
mg03*CLI>
<--- SIP read from 10.4.1.80:5060 <http://10.4.1.80:5060> --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060
From: "Thorsten" <sip:1000@82.98.89.134
<mailto:sip%3A1000@82.98.89.134>>;tag=as4c964973
To: <sip:017683035400@10.4.1.80
<mailto:sip%3A017683035400@10.4.1.80>>;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2
Call-ID: 5e209fbb7ebdbad97f0193515c5a2982(a)82.98.89.134
<mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134>
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.7 (i386/linux))
Content-Length: 0
Warning: 392 10.4.1.80:5060 <http://10.4.1.80:5060> "Noisy
feedback tells: pid=459
req_src_ip=82.98.89.134 <http://82.98.89.134> req_src_port=5060
in_uri=sip:017683035400@10.4.1.80
<mailto:sip%3A017683035400@10.4.1.80>
out_uri=sip:017683035400@192.168.13.102:5060
<http://sip:017683035400@192.168.13.102:5060> via_cnt==1
On SER I've configured to send this message:
if (method=="INVITE") {
if (uri =~ "sip:0[0-9]@*") {
route(3);
sl_send_reply("180", "Ringing");
break;
}
};
The other issue is that I don't see the caller id on the receiver
side.
I don't know if it is a asterisk or a SER issue. Only if I set the
caller id on asterisk manual in extensions.conf with
exten => _X.,1,Set(CALLERID(num)=06965006100)
I'll see the caller id on the receiver side.
I would really appreciate any help
Thanks
Thorsten
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