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@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 ---> 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@82.98.89.134
;tag=as4c964973
To: <sip:017683035400@10.4.1.80 sip%3A017683035400@10.4.1.80> Call-ID: 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 "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@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@82.98.89.134
;tag=as4c964973
To: <sip:017683035400@10.4.1.80 sip%3A017683035400@10.4.1.80
;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2
Call-ID: 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 "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@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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