[Serusers] Ringback tone on SER
Thorsten
serusers at thorko.de
Tue Apr 29 11:58:15 CEST 2008
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}@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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>>
>>
>>
>
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