29 Apr
2008
29 Apr
'08
11:45 a.m.
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>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 --->
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 <sip%3A1000(a)82.98.89.134>
;tag=as4c964973
To:
<sip:017683035400@10.4.1.80 <sip%3A017683035400(a)10.4.1.80>>
Call-ID: 5e209fbb7ebdbad97f0193515c5a2982(a)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 "Noisy feedback tells: pid=459
req_src_ip=82.98.89.134 req_src_port=5060
in_uri=sip:017683035400@10.4.1.80 <sip%3A017683035400(a)10.4.1.80>
out_uri=sip:017683035400@192.168.13.102:5060 via_cnt==1"
<------------->
--- (9 headers 0 lines) ---
mg03*CLI>
<--- SIP read from 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 <sip%3A1000(a)82.98.89.134>
;tag=as4c964973
To:
<sip:017683035400@10.4.1.80 <sip%3A017683035400(a)10.4.1.80>
;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2
Call-ID: 5e209fbb7ebdbad97f0193515c5a2982(a)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 "Noisy feedback tells: pid=459
req_src_ip=82.98.89.134 req_src_port=5060
in_uri=sip:017683035400@10.4.1.80 <sip%3A017683035400(a)10.4.1.80>
out_uri=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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