1 May
2008
1 May
'08
7:27 p.m.
Hi Thorsten, I have seen your sip trace and I couldn't find SDP information
in the 180 messages that you pasted. If you can check the " Content-Length:
0" it indicates if the SDP has or not information about media capabilities.
If I understand your test environment I supposed you have subscribers behind
the Asterisk boxes, so the CPE subscriber should have the capability to
generate a ringback inbound or outbound depends on if it receives or not a
180/183 message with SDP information (early media).
Regards
Alberto Cruz
-----Original Message-----
From: serusers-bounces(a)lists.iptel.org
[mailto:serusers-bounces@lists.iptel.org] On Behalf Of Thorsten
Sent: Tuesday, April 29, 2008 4:58 AM
To: serusers(a)thorko.de
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Ringback tone on SER
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---------------->pho
ne
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:
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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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