<br>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.<br>The ringback tone also looks like a configuration issue of your asterisk.<br>
<br>I would recommend you to get some info about the asterisk configuration to know which the problem might be.<br><br>Sam.<br><br><br><div><span class="gmail_quote">2008/4/29, Thorsten <<a href="mailto:serusers@thorko.de">serusers@thorko.de</a>>:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
Hi guys,<br> I'm trying to set up a SER server between 2 asterisk machines. I run<br> into 2 issues.<br> Whenever I call someone I don't get any ringback tone even so the call<br> initiating asterisk machine gets the 180 message after 100.<br>
<--- SIP read from <a href="http://10.4.1.80:5060">10.4.1.80:5060</a> ---><br> SIP/2.0 100 trying -- your call is important to us<br> Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060<br> From: "Thorsten" <<a href="mailto:sip%3A1000@82.98.89.134">sip:1000@82.98.89.134</a>>;tag=as4c964973<br>
To: <<a href="mailto:sip%3A017683035400@10.4.1.80">sip:017683035400@10.4.1.80</a>><br> Call-ID: <a href="mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134">5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134</a><br> CSeq: 102 INVITE<br>
Server: Sip EXpress router (0.9.7 (i386/linux))<br> Content-Length: 0<br> Warning: 392 <a href="http://10.4.1.80:5060">10.4.1.80:5060</a> "Noisy feedback tells: pid=459<br> req_src_ip=<a href="http://82.98.89.134">82.98.89.134</a> req_src_port=5060<br>
in_uri=<a href="mailto:sip%3A017683035400@10.4.1.80">sip:017683035400@10.4.1.80</a><br> out_uri=<a href="http://sip:017683035400@192.168.13.102:5060">sip:017683035400@192.168.13.102:5060</a> via_cnt==1"<br><br><br> <-------------><br>
--- (9 headers 0 lines) ---<br> mg03*CLI><br> <--- SIP read from <a href="http://10.4.1.80:5060">10.4.1.80:5060</a> ---><br> SIP/2.0 180 Ringing<br> Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060<br>
From: "Thorsten" <<a href="mailto:sip%3A1000@82.98.89.134">sip:1000@82.98.89.134</a>>;tag=as4c964973<br> To: <<a href="mailto:sip%3A017683035400@10.4.1.80">sip:017683035400@10.4.1.80</a>>;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2<br>
Call-ID: <a href="mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134">5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134</a><br> CSeq: 102 INVITE<br> Server: Sip EXpress router (0.9.7 (i386/linux))<br> Content-Length: 0<br>
Warning: 392 <a href="http://10.4.1.80:5060">10.4.1.80:5060</a> "Noisy feedback tells: pid=459<br> req_src_ip=<a href="http://82.98.89.134">82.98.89.134</a> req_src_port=5060<br> in_uri=<a href="mailto:sip%3A017683035400@10.4.1.80">sip:017683035400@10.4.1.80</a><br>
out_uri=<a href="http://sip:017683035400@192.168.13.102:5060">sip:017683035400@192.168.13.102:5060</a> via_cnt==1<br><br> On SER I've configured to send this message:<br> if (method=="INVITE") {<br> if (uri =~ "sip:0[0-9]@*") {<br>
route(3);<br> sl_send_reply("180", "Ringing");<br> break;<br> }<br> };<br>
<br> The other issue is that I don't see the caller id on the receiver side.<br> I don't know if it is a asterisk or a SER issue. Only if I set the<br> caller id on asterisk manual in extensions.conf with<br> exten => _X.,1,Set(CALLERID(num)=06965006100)<br>
I'll see the caller id on the receiver side.<br><br> I would really appreciate any help<br> Thanks<br> Thorsten<br><br> _______________________________________________<br> Serusers mailing list<br> <a href="mailto:Serusers@lists.iptel.org">Serusers@lists.iptel.org</a><br>
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