[Serusers] Ringback tone on SER

[Thorsten]

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}@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 at thorko.de <mailto:serusers at 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 at 82.98.89.134
>     <mailto:sip%3A1000 at 82.98.89.134>>;tag=as4c964973
>     To: <sip:017683035400 at 10.4.1.80 <mailto:sip%3A017683035400 at 10.4.1.80>>
>     Call-ID: 5e209fbb7ebdbad97f0193515c5a2982 at 82.98.89.134
>     <mailto:5e209fbb7ebdbad97f0193515c5a2982 at 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 at 10.4.1.80
>     <mailto:sip%3A017683035400 at 10.4.1.80>
>     out_uri=sip:017683035400 at 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 at 82.98.89.134
>     <mailto:sip%3A1000 at 82.98.89.134>>;tag=as4c964973
>     To: <sip:017683035400 at 10.4.1.80
>     <mailto:sip%3A017683035400 at 10.4.1.80>>;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2
>     Call-ID: 5e209fbb7ebdbad97f0193515c5a2982 at 82.98.89.134
>     <mailto:5e209fbb7ebdbad97f0193515c5a2982 at 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 at 10.4.1.80
>     <mailto:sip%3A017683035400 at 10.4.1.80>
>     out_uri=sip:017683035400 at 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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