[Asterisk-Users] Some asterisk ser problems

Alex alexander_gav at yahoo.com
Tue Mar 1 09:19:40 MST 2005 

In ser.cfg
------------------------------------------------------------------------------
if (method == "INVITE") { 
    if (uri =~ "sip:1[0-9]{10}@*"){ 
        log(1, "Forwarding to Asterisk\n"); 
        rewritehostport("xxx.xxx.xxx.xxx:5061"); 
        t_relay(); 
        break; 
    }     
 }       
 
 
In  sip.conf
---------------------------------------------------------------------------------------
[ser]
type=friend
host=xxx.xxx.xxx.xxx
context=from-ser
 
In extension.conf
----------------------------------------------------------------------------------------
[from-ser]
exten => _1,1,Dial(SIP/sipphonenumber at xxx.xxx.xxx.xxx,20,r)
 
Sip Debug from Asterisk
----------------------------------
 
Sip read: 
INVITE sip:1xxxxxxx at xxxx.xxxx.xx.xxx:5061 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKe189.20ca26d7.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK45dda3d0
From: "Alex" <sip:phonefromwhich at xxx.xxx.xxx.xxx>;tag=00036b09607e003b16a3f758-1d78797a
To: <sip:1xxxxxxxx at xxx.xxx.xxx.xxx>
Call-ID: 00036b09-607e003b-552c14b9-021cab1d at xxx.xxx.xxx.xxx
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact: <sip:callingfrom at xxx.xxx.xxx.xxx:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 248
Accept: application/sdp
v=0
o=Cisco-SIPUA 7329 20490 IN IP4 numbercallingfrom
s=SIP Call
c=IN IP4 numbercallingfrom
t=0 0
m=audio 26274 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 11 lines
Using latest request as basis request
Sending to xxx.xxx.xxx.xxx : 5060 (non-NAT)
Found peer 'ser'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port numbercallingfrom:26274
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 1xxxxxxxxx in from-ser
Reliably Transmitting (no NAT):
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKe189.20ca26d7.0
Via: SIP/2.0/UDP numbercallingfrom:5060;branch=z9hG4bK45dda3d0
From: "Alexg" <sip:numbercallingfrom at xxx.xxx.xxx.xxx>;tag=00036b09607e003b16a3f758-1d78797a
To: <sip:1xxxxxxxx at xxx.xxx.xxx.xxx>;tag=as6a19e3f4
Call-ID: 00036b09-607e003b-552c14b9-021cab1d at numbercallingfrom
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1xxxxxxxxxx at xxx.xxx.xxx.xxx:5061>
Content-Length: 0

 to xxx.xxx.xxx.xxx:5060

Sip read: 
ACK sip:1xxxxxxxxxxx at xxx.xxx.xxx.xxx:5061 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKe189.20ca26d7.0
From: "Alexg" <sip:numbercallingfrom at xxx.xxx.xxx.xxx>;tag=00036b09607e003b16a3f758-1d78797a
Call-ID: 00036b09-607e003b-552c14b9-021cab1d at ipcallingfrom
To: <sip:1xxxxxxxxxxxx at xxx.xxx.xxx.xxx>;tag=as6a19e3f4
CSeq: 101 ACK
User-Agent: Sip EXpress router(0.8.14 (i386/linux))
Content-Length: 0

8 headers, 0 lines
Destroying call '00036b09-607e003b-552c14b9-021cab1d at ipcallingfrom'
 
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 .
 
The help is more than welcome.
Thanks.
 
 
Alistair Cunningham <acunningham at integrics.com> wrote:
Alex,

If you are forwarding calls in SER based on URI patterns rather than the 
location database, you don't need to register Asterisk with SER. Instead 
of the register line, you should have a peer for SER; something like this:

[ser]
type = friend
host = 
context = 

There are lots more options for the peer, but this should get you started.

If you'd like more detailed support, my company, Integrics Ltd, does 
support for both Asterisk and SER. We can also write the IVRs for you.

Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/


Alex wrote:
> I have some simple questions and i need your help guys.
> 
> I have ser server which working fine, between users.
> I am trying to add some more features to the ser. Most important is the IVR.
> 
> I installed Asterisk and i am trying to register user in asterisk with 
> no success.
> Part of ser.cfg file where i am trying to redirect the call to the asterisk.
> ---------------------------------------------------------------------------------------------------------
> if (method == "INVITE") {
> if (uri =~ "sip:1[0-9]{4}@*"){
> log(1, "Forwarding to Asterisk\n");
> rewritehostport("xx.xx.xx.xx:xxxx");
> t_relay();
> break;
> }
> }
> -----------------------------------------------------------------------------------------------------------
> 
> 
> inside sip.conf i have
> -----------------------------------------------------------------------------------------------------
> register => sipphonenumber:password at siphostname/3333
> 
> 
> error
> ---------------------------------------------------------------------------------------------------------
> chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to 
> ';tag=as12200854'
> 
> I need some help with configuring asterisk to work with ser.
> Let's say i am calling from sip phone to number 12345 , i would like to 
> enter into IVR system where i can configure which number to press, what 
> kind of music to play etc.....
> 
> The main goal is to create IVR system for identical phone number for SIP 
> users.
> 
> Thanks for any help.
> 
> ------------------------------------------------------------------------
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> 
> ------------------------------------------------------------------------
> 
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Content preview: Alex, If you are forwarding calls in SER based on URI 
patterns rather than the location database, you don't need to register 
Asterisk with SER. Instead of the register line, you should have a 
peer for SER; something like this: [...] 

Content analysis details: (0.1 points, 5.0 required)

pts rule name description
---- ---------------------- --------------------------------------------------
0.1 FORGED_RCVD_HELO Received: contains a forged HELO


		
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