Topic: DID Incoming calls from PortaOne system

[compsos]

Hi
We have an account with World Dial Point who use a PortaOne system. We can place calls and have done for quite a while. We have now taken out a DID service but it is always answered (very quickly) by a message at their system that "The person at...is unavailable". I do not think we are seeing anything of the call

Their support has suggested that it is the useragent string of "Asterisk PBX" needs to be changed to something like PAP2. The trunk stanza shows useragent of PAP2T. But the 'sip show peer 0740840650' returns a blank useragent field. Changing the stanza does not change this test and the support still says we are passing the "Asterisk PBX" through to them.

Googling shows that problems between Asterisk and PortaOne is not uncommon. PortaOne have released a Radius client for Asterisk. Is that what we need to receive incoming calls from the DID?

This I think is the only evidence of the incoming call from my mobile.

Code: [Select]

<--- SIP read from 203.176.185.10:5060 --->
INVITE sip:61740840650@220.245.107.242 SIP/2.0
Record-Route: <sip:203.176.185.10;ftag=fa503ca7f97694c630eb69dab4b58782;lr>
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4bde.503177c818797d1f1278a8ae92d06f43.0
Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK8d66d0bae6730ce3459072f64b76585e;rport=5061
Max-Forwards: 16
From: <sip:0429338896@203.176.185.10>;tag=fa503ca7f97694c630eb69dab4b58782
To: <sip:61740840650@203.176.185.10>
Call-ID: call-71605C32-370F-2C10-0308-DEDA@203.176.186.10~1o
CSeq: 200 INVITE
Contact: Anonymous <sip:203.176.185.10:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 876176738-909730608-1717986918-1684366392
h323-conf-id: 876176738-909730608-1717986918-1684366392
H323-credit-time: 7200
Content-disposition: session
Content-Length: 286
Content-Type: application/sdp

v=0
o=Sippy 151550124 0 IN IP4 203.176.185.10
s=VoipCall
t=0 0
m=audio 56392 RTP/AVP 18 4 8 0 101
c=IN IP4 203.176.185.10
a=rtpmap:18 g729/8000/1
a=abcde:20
a=rtpmap:4 g723/8000/1
a=rtpmap:8 pcma/8000/1
a=rtpmap:0 pcmu/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv

<--- Reliably Transmitting (no NAT) to 203.176.185.10:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4bde.503177c818797d1f1278a8ae92d06f43.0;received=203.176.185.10
Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK8d66d0bae6730ce3459072f64b76585e;rport=5061
From: <sip:0429338896@203.176.185.10>;tag=fa503ca7f97694c630eb69dab4b58782
To: <sip:61740840650@203.176.185.10>;tag=as178f3fd7
Call-ID: call-71605C32-370F-2C10-0308-DEDA@203.176.186.10~1o
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48ad6d39"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'call-71605C32-370F-2C10-0308-DEDA@203.176.186.10~1o' in 6400 ms (Method: INVITE)
*CLI>
<--- SIP read from 203.176.185.10:5060 --->
ACK sip:61740840650@220.245.107.242 SIP/2.0
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4bde.503177c818797d1f1278a8ae92d06f43.0
From: <sip:0429338896@203.176.185.10>;tag=fa503ca7f97694c630eb69dab4b58782
Call-ID: call-71605C32-370F-2C10-0308-DEDA@203.176.186.10~1o
To: <sip:61740840650@203.176.185.10>;tag=as178f3fd7
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
--- (14 headers 0 lines) ---
Really destroying SIP dialog '06f12004537182670d67405e7e984fb9@192.168.36.1' Method: OPTIONS
Really destroying SIP dialog 'call-71605C32-370F-2C10-0308-DEDA@203.176.186.10~1o' Method: ACK


We have reset the type from peer to friend but 'sip reload' returns

Code: [Select]

[Apr 20 09:41:00] WARNING[12209]: chan_sip.c:17983 reload_config: Section '0740840650' lacks type
[Apr 20 09:41:00] WARNING[12209]: chan_sip.c:17983 reload_config: Section 'wdp_out' lacks type

'sip show peer 0740840650' returns

Code: [Select]

  * Name       : 0740840650
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : mainmenu
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  FromUser     : 61740840650
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : sip.bbvoice.com.au
  Addr->IP     : 203.176.185.10 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username: 61740840650
  SIP Options  : (none)
  Codecs       : 0x10c (ulaw|alaw|g729)
  Codec Order  : (g729:20,alaw:20,ulaw:20)
  Auto-Framing:  No
  Status       : OK (88 ms)
  Useragent    :
  Reg. Contact :


Has anyone else got a working DID with World Dial Point?
TIA

[SARK devs]

OK...

two things;

you need to set insecure=very (or insecure=port,invite) in your sip definition (you don't need user=friend by the way).

Next you need to create a PTT_DiD_Group trunk entry of 61740840650 to receive and route your inbound call.

Best

S

[compsos]

Hi S

Thank you for the reply.

Our User Stanza for the trunk definition already had insecure=port,invite. I have now changed the type back to type=peer. But still the providers recorded message. It is so quick I am suspicious.

The current sip set debug

Code: [Select]

<--- SIP read from 203.176.185.10:5060 --->
INVITE sip:61740840650@220.245.107.242 SIP/2.0
Record-Route: <sip:203.176.185.10;ftag=92441ef3e2d601aba09469b66b513d7e;lr>
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKb661.cb2160f0a5b038ab66eefb5f53580251.0
Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5bb892668911967d45e0349badbb9df8;rport=5061
Max-Forwards: 16
From: <sip:0429338896@203.176.185.10>;tag=92441ef3e2d601aba09469b66b513d7e
To: <sip:61740840650@203.176.185.10>
Call-ID: call-7182373C-710F-2C10-1402-F08F@203.176.186.10~1o
CSeq: 200 INVITE
Contact: Anonymous <sip:203.176.185.10:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 876176738-1664563555-1717986918-1714436196
h323-conf-id: 876176738-1664563555-1717986918-1714436196
H323-credit-time: 7200
Content-disposition: session
Content-Length: 286
Content-Type: application/sdp

v=0
o=Sippy 140536396 0 IN IP4 203.176.185.10
s=VoipCall
t=0 0
m=audio 35676 RTP/AVP 18 4 8 0 101
c=IN IP4 203.176.185.10
a=rtpmap:18 g729/8000/1
a=abcde:20
a=rtpmap:4 g723/8000/1
a=rtpmap:8 pcma/8000/1
a=rtpmap:0 pcmu/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv

<------------->
--- (18 headers 13 lines) ---
Sending to 203.176.185.10 : 5060 (no NAT)
Using INVITE request as basis request - call-7182373C-710F-2C10-1402-F08F@203.176.186.10~1o
Found peer 'peerwdp_out'
server-sos*CLI>
<--- Reliably Transmitting (no NAT) to 203.176.185.10:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKb661.cb2160f0a5b038ab66eefb5f53580251.0;received=203.176.185.10
Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5bb892668911967d45e0349badbb9df8;rport=5061
From: <sip:0429338896@203.176.185.10>;tag=92441ef3e2d601aba09469b66b513d7e
To: <sip:61740840650@203.176.185.10>;tag=as366733b4
Call-ID: call-7182373C-710F-2C10-1402-F08F@203.176.186.10~1o
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09be00ce"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'call-7182373C-710F-2C10-1402-F08F@203.176.186.10~1o' in 6400 ms (Method: INVITE)
server-sos*CLI>
<--- SIP read from 203.176.185.10:5060 --->
ACK sip:61740840650@220.245.107.242 SIP/2.0
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKb661.cb2160f0a5b038ab66eefb5f53580251.0
From: <sip:0429338896@203.176.185.10>;tag=92441ef3e2d601aba09469b66b513d7e
Call-ID: call-7182373C-710F-2C10-1402-F08F@203.176.186.10~1o
To: <sip:61740840650@203.176.185.10>;tag=as366733b4
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---


[SARK devs]

Did you create the DiD entry?

[compsos]

This is the definition.
Trunk DID or IP Name: 61740840650    
Carrier Name: PTT_DiD_Group
Technology: DiD
   Host:

db selintra show 61740840650
61740840650=lineIO
    active=YES
    alertinfo=
    carrier=PTT_DiD_Group
    closedisa=NO
    closegreet=None
    closeivr=NO
    closequeue=None
    closeroute=Operator
    cluster=default
    desc=WDPDID
    disa=NO
    faxdetect=NO
    forceivr=NO
    host=
    inprefix=
    lcl=NO
    opengreet=None
    openroute=Operator
    peername=
    queue=None
    remotenum=61740840650
    swoclip=YES
    transform=
    zzeor=EOR

Is this correct?

[SARK devs]

send your sip.conf to admin@aelintra.com

cheers

S

[bbialy]

Gentelmen,
just set in sip headers
useragent=portasipfriendly


simple isn't it??
i was looking this setting for 3 weekes.

[compsos]

bbialy

just set in sip headers
useragent=portasipfriendly

That did not work for us. How is your system setup?

[gippsweb]

I use Franks normal system and have had my DID working until his pear shaped software upgrade 2 weeks back. DID hasn't worked since.

The way mine used to work was by changing the registration string to
username:password@ipaddress/DIDnumber

This worked for us until 2 weeks ago

[compsos]

Hi Gippsweb
Do you mean "Franks" as in WDP? It might be interesting to excahnge sip debug reports of incoming calls.
We are currently on Sail 2.3.1-3 with Dahdi. What is your setup? Is there a way we can contact you?

[gippsweb]

Hi Gordon,
yes as in WDP, but I'm not on the business service so my sip debug won't be of any help to you....
I'm only on his residential servive which has changed recently.
The business service uses Comvergences service from memory...

[compsos]

We log in the same way and it does sound like we have the same problem with the same provider. So doubt if the grade of service is the issue. Selintra & Frank suggested there was a problem with the proxy-authentication but so far have not found a way of correcting it.

We did a session with Frank and modified the user-agent in the general section of the sip.conf file. He said he dialled me on the DiD but we have never been able to dial from outside in. Has frank been able to dial you since the DiD stopped? Also when we do a sip show peer "WDP_DiD" it does not show any useragent string. Are you seeing the same issues?

[gippsweb]

Mine stopped working after they did the software upgrades a week or two back. I wasn't even getting anything off sip debug, which I found very strange.
My registration string changes worked perfectly until the changes at his end..

I've since had some help from his engineers, we've now at least got calls being rejected by my box. But they are USA based and only available late at night AUS time.

[compsos]

Software updates being on Sail or WDP?

[gippsweb]

The residential service was had the billing system upgraded and asterisk upgraded to 1.6