<DIV>In ser.cfg</DIV>
<DIV>------------------------------------------------------------------------------<BR>if (method == "INVITE") { <BR> if (uri =~ "sip:1[0-9]{10}@*"){ <BR> log(1, "Forwarding to Asterisk\n"); <BR> rewritehostport("xxx.xxx.xxx.xxx:5061"); <BR> t_relay(); <BR> break; <BR> } <BR> } </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>In sip.conf</DIV>
<DIV>---------------------------------------------------------------------------------------</DIV>
<DIV>[ser]<BR>type=friend<BR>host=xxx.xxx.xxx.xxx<BR>context=from-ser</DIV>
<DIV> </DIV>
<DIV>In extension.conf</DIV>
<DIV>----------------------------------------------------------------------------------------</DIV>
<DIV>[from-ser]<BR>exten => _1,1,Dial(<A href="mailto:SIP/sipphonenumber@xxx.xxx.xxx.xxx,20,r">SIP/sipphonenumber@xxx.xxx.xxx.xxx,20,r</A>)</DIV>
<DIV> </DIV>
<DIV>Sip Debug from Asterisk</DIV>
<DIV>----------------------------------</DIV>
<DIV> </DIV>
<DIV>Sip read: <BR>INVITE sip:1xxxxxxx@xxxx.xxxx.xx.xxx:5061 SIP/2.0<BR>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKe189.20ca26d7.0<BR>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK45dda3d0<BR>From: "Alex" <sip:phonefromwhich@xxx.xxx.xxx.xxx>;tag=00036b09607e003b16a3f758-1d78797a<BR>To: <sip:1xxxxxxxx@xxx.xxx.xxx.xxx><BR>Call-ID: <A href="mailto:00036b09-607e003b-552c14b9-021cab1d@xxx.xxx.xxx.xxx">00036b09-607e003b-552c14b9-021cab1d@<FONT color=#000000>xxx.xxx.xxx.xxx</FONT></A></DIV>
<DIV>CSeq: 101 INVITE<BR>User-Agent: CSCO/6<BR>Contact: <sip:callingfrom@xxx.xxx.xxx.xxx:5060><BR>Expires: 180<BR>Content-Type: application/sdp<BR>Content-Length: 248<BR>Accept: application/sdp</DIV>
<DIV>v=0<BR>o=Cisco-SIPUA 7329 20490 IN IP4 numbercallingfrom<BR>s=SIP Call<BR>c=IN IP4 numbercallingfrom<BR>t=0 0<BR>m=audio 26274 RTP/AVP 0 8 18 101<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:18 G729/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-15</DIV>
<DIV>13 headers, 11 lines<BR>Using latest request as basis request<BR>Sending to xxx.xxx.xxx.xxx : 5060 (non-NAT)<BR>Found peer 'ser'<BR>Found RTP audio format 0<BR>Found RTP audio format 8<BR>Found RTP audio format 18<BR>Found RTP audio format 101<BR>Peer audio RTP is at port numbercallingfrom:26274<BR>Found description format PCMU<BR>Found description format PCMA<BR>Found description format G729<BR>Found description format telephone-event<BR>Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)<BR>Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)<BR>Looking for 1xxxxxxxxx in from-ser<BR>Reliably Transmitting (no NAT):<BR>SIP/2.0 484 Address Incomplete<BR>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKe189.20ca26d7.0<BR>Via: SIP/2.0/UDP numbercallingfrom:5060;branch=z9hG4bK45dda3d0<BR>From: "Alexg"
<sip:numbercallingfrom@xxx.xxx.xxx.xxx>;tag=00036b09607e003b16a3f758-1d78797a<BR>To: <sip:1xxxxxxxx@xxx.xxx.xxx.xxx>;tag=as6a19e3f4<BR>Call-ID: <A href="mailto:00036b09-607e003b-552c14b9-021cab1d@numbercallingfrom">00036b09-607e003b-552c14b9-021cab1d@numbercallingfrom</A><BR>CSeq: 101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>Contact: <sip:1xxxxxxxxxx@xxx.xxx.xxx.xxx:5061><BR>Content-Length: 0</DIV>
<DIV><BR> to xxx.xxx.xxx.xxx:5060</DIV>
<DIV><BR>Sip read: <BR>ACK sip:1xxxxxxxxxxx@xxx.xxx.xxx.xxx:5061 SIP/2.0<BR>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKe189.20ca26d7.0<BR>From: "Alexg" <sip:numbercallingfrom@xxx.xxx.xxx.xxx>;tag=00036b09607e003b16a3f758-1d78797a<BR>Call-ID: <A href="mailto:00036b09-607e003b-552c14b9-021cab1d@ipcallingfrom">00036b09-607e003b-552c14b9-021cab1d@ipcallingfrom</A><BR>To: <sip:1xxxxxxxxxxxx@xxx.xxx.xxx.xxx>;tag=as6a19e3f4<BR>CSeq: 101 ACK<BR>User-Agent: Sip EXpress router(0.8.14 (i386/linux))<BR>Content-Length: 0</DIV>
<DIV><BR>8 headers, 0 lines<BR>Destroying call <A href="mailto:'00036b09-607e003b-552c14b9-021cab1d@ipcallingfrom'">'00036b09-607e003b-552c14b9-021cab1d@ipcallingfrom'</A></DIV>
<DIV> </DIV>
<DIV>After call i hear busyvoice on the line. I have to configure it to use some IVR system in order to be abble to choose numbers (extensions) and depend on the extension to play some kind of music .</DIV>
<DIV> </DIV>
<DIV>The help is more than welcome.</DIV>
<DIV>Thanks.</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV><B><I>Alistair Cunningham <acunningham@integrics.com></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Alex,<BR><BR>If you are forwarding calls in SER based on URI patterns rather than the <BR>location database, you don't need to register Asterisk with SER. Instead <BR>of the register line, you should have a peer for SER; something like this:<BR><BR>[ser]<BR>type = friend<BR>host = <IP SER of hostname or address><BR>context = <CONTEXT calls incoming for><BR><BR>There are lots more options for the peer, but this should get you started.<BR><BR>If you'd like more detailed support, my company, Integrics Ltd, does <BR>support for both Asterisk and SER. We can also write the IVRs for you.<BR><BR>Alistair Cunningham,<BR>Integrics Ltd,<BR>Telephony, Database, Unix consulting worldwide<BR>+44 (0)7870 699 479<BR>http://integrics.com/<BR><BR><BR>Alex wrote:<BR>> I have some simple questions and i need your help guys.<BR>> <BR>> I have ser server which working fine, between users.<BR>>
I am trying to add some more features to the ser. Most important is the IVR.<BR>> <BR>> I installed Asterisk and i am trying to register user in asterisk with <BR>> no success.<BR>> Part of ser.cfg file where i am trying to redirect the call to the asterisk.<BR>> ---------------------------------------------------------------------------------------------------------<BR>> if (method == "INVITE") {<BR>> if (uri =~ "sip:1[0-9]{4}@*"){<BR>> log(1, "Forwarding to Asterisk\n");<BR>> rewritehostport("xx.xx.xx.xx:xxxx");<BR>> t_relay();<BR>> break;<BR>> }<BR>> }<BR>> -----------------------------------------------------------------------------------------------------------<BR>> <BR>> <BR>> inside sip.conf i have<BR>> -----------------------------------------------------------------------------------------------------<BR>> register => sipphonenumber:password@siphostname/3333<BR>> <BR>> <BR>> error<BR>>
---------------------------------------------------------------------------------------------------------<BR>> chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to <BR>> '<sip:sipphonenumber@siphostname>;tag=as12200854'<BR>> <BR>> I need some help with configuring asterisk to work with ser.<BR>> Let's say i am calling from sip phone to number 12345 , i would like to <BR>> enter into IVR system where i can configure which number to press, what <BR>> kind of music to play etc.....<BR>> <BR>> The main goal is to create IVR system for identical phone number for SIP <BR>> users.<BR>> <BR>> Thanks for any help.<BR>> <BR>> ------------------------------------------------------------------------<BR>> Do you Yahoo!?<BR>> Take Yahoo! Mail with you! <BR>> <HTTP: maildemo mobile.yahoo.com *http: mobile taglines mail_us us.rd.yahoo.com><BR>> Get it on your mobile phone.<BR>> <BR>> <BR>>
------------------------------------------------------------------------<BR>> <BR>> _______________________________________________<BR>> Asterisk-Users mailing list<BR>> Asterisk-Users@lists.digium.com<BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> To UNSUBSCRIBE or update options visit:<BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>_______________________________________________<BR>Asterisk-Users mailing list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR><BR>Spam detection software, running on the system "zeus.avanzada7.com", has<BR>identified this incoming email as possible spam. The original message<BR>has been attached to this so you can view it (if it isn't spam) or label<BR>similar future email. If you have any questions, see<BR>the administrator of that system for
details.<BR><BR>Content preview: Alex, If you are forwarding calls in SER based on URI <BR>patterns rather than the location database, you don't need to register <BR>Asterisk with SER. Instead of the register line, you should have a <BR>peer for SER; something like this: [...] <BR><BR>Content analysis details: (0.1 points, 5.0 required)<BR><BR>pts rule name description<BR>---- ---------------------- --------------------------------------------------<BR>0.1 FORGED_RCVD_HELO Received: contains a forged HELO<BR><BR></BLOCKQUOTE></sip:sipphonenumber@siphostname><p>
<hr size=1>Do you Yahoo!?<br>
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