Need help for outgoing calls from Asterisk to IMS network

Asterisk Asterisk Support
1

Hello!

First, sorry for my very bad English!!!

I’m a Professional Degree trainee in the field of VOIP Network and System Management into Alcatel-Lucent firm.
During my placement, I have to connect Asterisk (Installed on Ubuntu 9.10 OS) with an Alcatel-Lucent IMS platform.
The way to achieved this integration is the following:
I have 3 users A, B, C.
A and B are both IMS subscribers, and C is an Asterisk user (unknown for IMS network).
I’m using the subscriber B as a “gateway” between Asterisk user C and IMS subscriber A.
So, Asterisk performs a registration on subscriber B, exactly as it was a voip provider profile.
Then, calls from subscriber A to B are forwarded to Asterisk user C (Incoming calls), and calls from C to A are forwarded to subscriber B to reach A (Outgoing calls).

The problem:
Asterisk registration and incoming calls works fine, but on outgoing calls I always get the message from the p-cscf (proxy server of IMS network):
“407 Proxy Authentication required” (in wireshark);
“Fail to authenticate on invite” (in Asterisk CLI).

sip.conf

[general]
realm=ims.perros.bzh ;my domain name (ims.perros.bzh) for digest md5 authentication
defaultexpiry=3600
pedantic=yes ;if not present registration doesn't works
allow=ulaw
allow=alaw
allow=gsm
language=fr
deny=0.0.0.0/0.0.0.0
permit=192.168.135.7/255.255.255.255

register => 0296000001@ims.perros.bzh:0296000001:0296000001@192.168.135.7:5060/0296000001 ;registration of user B known of IMS network

[ims_in]
type=friend
context=appels_entrants
host=192.168.135.7

[ims_out]
type=peer
username=0296000001
fromuser=0296000001
secret=0296000001
fromdomain=192.168.135.7
insecure=very
host=192.168.135.7
qualify=yes
nat=no
allow=ulaw
allow=alaw
allow=gsm
outboundproxy=192.168.135.7

[0296000001] ;my asterisk user C which carry the same name as IMS subscriber B for transparency reasons between IMS and Asterisk worlds
type=friend  ;However if it has a different name the problem stays the same
username=0296000001
fromuser=0296000001
callerid="0296000001"
secret=0296000001
context=interne
host=dynamic

extensions.conf

[appels_entrants]
exten => 0296000001,1,Dial(SIP/${EXTEN})

[appels_sortants]
exten => _9.,1,Dial(SIP/${EXTEN:1}@ims_out)

[interne]
include => appels_sortants
exten => _1.,1,Dial(SIP/${EXTEN:1})

I searched during hours on the Internet and I tried a lot of things but nothing solved my problem.
I have noticed that IMS calls between IMS subscribers are based on domain name (invite user@domain,) whereas calls from Asterisk user to IMS subscriber are based on the P-CSCF IP address (invite user@192.168.135.7).
I tried to use SRV records in asterisk with bind9 to based calls on domain name but it doesn’t works.

How can I provide this authentication during outgoing calls?
If it’s not really an authentication problem what’s wrong between IMS network and Asterisk.

I also heard this “SIP signaling from IMS is a little bit different from the Asterisk SIP stack.”.
here: http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg22020.html
What can you tell me about that?

Thank you in advance.

François Bourdoulous.

2

Hello,

Please post a SIP trace (sip set debug) of an outgoing call.

BR,
Ioan.

3

Hellow,

First, Thanks for your very fast reply, and sorry for the slowness of mine.

Many problems appeared on the IMS platform and finally we have entirely reinstalled it.
I think my problem is still the same but now I can not make outgoing and incoming calls due to “400 Bad Request”.
Something has changed in the new configuration, and i’m trying to solved it.

However, I post a SIP trace (sip set debug) of an outgoing call.

<--- SIP read from UDP:192.168.135.90:5061 --->
INVITE sip:9jerome@ims.perros.bzh;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.135.90:5061;branch=z9hG4bK-d8754z-0ee23e9b3b575f63-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:0296000001@192.168.135.90:5061;transport=UDP>
To: <sip:9jerome@ims.perros.bzh;transport=UDP>
From: "0296000001"<sip:0296000001@ims.perros.bzh;transport=UDP>;tag=4944de53
Call-ID: NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rev.5415
Content-Length: 308

v=0
o=Z 0 0 IN IP4 192.168.135.90
s=Z
c=IN IP4 192.168.135.90
t=0 0
m=audio 8000 RTP/AVP 3 110 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 15 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 192.168.135.90 : 5061 (no NAT)
Using INVITE request as basis request - NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.
Found peer '0296000001' for '0296000001' from 192.168.135.90:5061

<--- Reliably Transmitting (no NAT) to 192.168.135.90:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.135.90:5061;branch=z9hG4bK-d8754z-0ee23e9b3b575f63-1---d8754z-;received=192.168.135.90;rport=5061
From: "0296000001"<sip:0296000001@ims.perros.bzh;transport=UDP>;tag=4944de53
To: <sip:9jerome@ims.perros.bzh;transport=UDP>;tag=as4fb18ce4
Call-ID: NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="ims.perros.bzh", nonce="426ca3e6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.135.90:5061 --->
ACK sip:9jerome@ims.perros.bzh;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.135.90:5061;branch=z9hG4bK-d8754z-0ee23e9b3b575f63-1---d8754z-;rport
Max-Forwards: 70
To: <sip:9jerome@ims.perros.bzh;transport=UDP>;tag=as4fb18ce4
From: "0296000001"<sip:0296000001@ims.perros.bzh;transport=UDP>;tag=4944de53
Call-ID: NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.135.90:5061 --->
INVITE sip:9jerome@ims.perros.bzh;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.135.90:5061;branch=z9hG4bK-d8754z-d23f12ffce37a7b1-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:0296000001@192.168.135.90:5061;transport=UDP>
To: <sip:9jerome@ims.perros.bzh;transport=UDP>
From: "0296000001"<sip:0296000001@ims.perros.bzh;transport=UDP>;tag=4944de53
Call-ID: NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rev.5415
Authorization: Digest username="0296000001",realm="ims.perros.bzh",nonce="426ca3e6",uri="sip:9jerome@ims.perros.bzh;transport=UDP",response="5e80d932851125388d548cd696ed60db",algorithm=MD5
Content-Length: 308

v=0
o=Z 0 0 IN IP4 192.168.135.90
s=Z
c=IN IP4 192.168.135.90
t=0 0
m=audio 8000 RTP/AVP 3 110 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 192.168.135.90 : 5061 (no NAT)
Using INVITE request as basis request - NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.
Found peer '0296000001' for '0296000001' from 192.168.135.90:5061
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.135.90:8000
Looking for 9jerome in interne (domain ims.perros.bzh)
list_route: hop: <sip:0296000001@192.168.135.90:5061;transport=UDP>

<--- Transmitting (no NAT) to 192.168.135.90:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.135.90:5061;branch=z9hG4bK-d8754z-d23f12ffce37a7b1-1---d8754z-;received=192.168.135.90;rport=5061
From: "0296000001"<sip:0296000001@ims.perros.bzh;transport=UDP>;tag=4944de53
To: <sip:9jerome@ims.perros.bzh;transport=UDP>
Call-ID: NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:9jerome@192.168.135.90>
Content-Length: 0


<------------>
    -- Executing [9jerome@interne:1] Dial("SIP/0296000001-00000008", "SIP/jerome@ims_out") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 192.168.135.90 port 17294
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.135.7:5060:
INVITE sip:jerome@192.168.135.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.135.90:5060;branch=z9hG4bK04890892;rport
Max-Forwards: 70
From: "0296000001" <sip:0296000001@192.168.135.7>;tag=as4569878d
To: <sip:jerome@192.168.135.7>
Contact: <sip:0296000001@192.168.135.90>
Call-ID: 08a8144d7ade734d0f4da36c4b385a20@192.168.135.7
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Remote-Party-ID: "0296000001" <sip:0296000001@192.168.135.7>;privacy=off;screen=no
Date: Fri, 21 May 2010 09:38:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 93591572 93591572 IN IP4 192.168.135.90
s=Asterisk PBX 1.6.2.2
c=IN IP4 192.168.135.90
t=0 0
m=audio 17294 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called jerome@ims_out

<--- SIP read from UDP:192.168.135.7:5060 --->
SIP/2.0 100 Trying
Call-ID: 08a8144d7ade734d0f4da36c4b385a20@192.168.135.7
Via: SIP/2.0/UDP 192.168.135.90:5060;received=192.168.135.90;branch=z9hG4bK04890892;rport=5060
To: <sip:jerome@192.168.135.7>
From: "0296000001" <sip:0296000001@192.168.135.7>;tag=as4569878d
CSeq: 102 INVITE
Date: Fri, 21 May 2010 09:38:53 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

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

<--- SIP read from UDP:192.168.135.7:5060 --->
SIP/2.0 400 Bad Request
Call-ID: 08a8144d7ade734d0f4da36c4b385a20@192.168.135.7
Via: SIP/2.0/UDP 192.168.135.90:5060;received=192.168.135.90;branch=z9hG4bK04890892;rport=5060
To: <sip:jerome@192.168.135.7>;tag=4be2b15f-1274434733758645
From: "0296000001" <sip:0296000001@192.168.135.7>;tag=as4569878d
CSeq: 102 INVITE
Contact: <sip:pcsf-stdn.imsgroup0-000.ics.ims.perros.bzh:5060>
Date: Fri, 21 May 2010 09:38:53 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Got SIP response 400 "Bad Request" back from 192.168.135.7
Transmitting (no NAT) to 192.168.135.7:5060:
ACK sip:jerome@192.168.135.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.135.90:5060;branch=z9hG4bK04890892;rport
Max-Forwards: 70
From: "0296000001" <sip:0296000001@192.168.135.7>;tag=as4569878d
To: <sip:jerome@192.168.135.7>;tag=4be2b15f-1274434733758645
Contact: <sip:0296000001@192.168.135.90>
Call-ID: 08a8144d7ade734d0f4da36c4b385a20@192.168.135.7
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.2
Remote-Party-ID: "0296000001" <sip:0296000001@192.168.135.7>;privacy=off;screen=no
Content-Length: 0


---
    -- SIP/ims_out-00000009 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/0296000001-00000008' status is 'CONGESTION'

<--- Reliably Transmitting (no NAT) to 192.168.135.90:5061 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.135.90:5061;branch=z9hG4bK-d8754z-d23f12ffce37a7b1-1---d8754z-;received=192.168.135.90;rport=5061
From: "0296000001"<sip:0296000001@ims.perros.bzh;transport=UDP>;tag=4944de53
To: <sip:9jerome@ims.perros.bzh;transport=UDP>;tag=as30947c31
Call-ID: NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


<------------>

<--- SIP read from UDP:192.168.135.90:5061 --->
ACK sip:9jerome@ims.perros.bzh;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.135.90:5061;branch=z9hG4bK-d8754z-d23f12ffce37a7b1-1---d8754z-;rport
Max-Forwards: 70
To: <sip:9jerome@ims.perros.bzh;transport=UDP>;tag=as30947c31
From: "0296000001"<sip:0296000001@ims.perros.bzh;transport=UDP>;tag=4944de53
Call-ID: NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '08a8144d7ade734d0f4da36c4b385a20@192.168.135.7' Method: INVITE
Really destroying SIP dialog 'NmNiMDY0Mjc3ZTE1ZWNkMGQzMzhjMjUwOWY2YjdjOTU.' Method: ACK
Reliably Transmitting (no NAT) to 192.168.135.7:5060:
OPTIONS sip:192.168.135.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.135.90:5060;branch=z9hG4bK08a21976;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.135.90>;tag=as01a0bd91
To: <sip:192.168.135.7>
Contact: <sip:asterisk@192.168.135.90>
Call-ID: 789e513760d5f343578d1fc53226bec8@192.168.135.90
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.2
Date: Fri, 21 May 2010 09:38:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.135.7:5060 --->
SIP/2.0 200 OK
Call-ID: 789e513760d5f343578d1fc53226bec8@192.168.135.90
Via: SIP/2.0/UDP 192.168.135.90:5060;received=192.168.135.90;branch=z9hG4bK08a21976;rport=5060
To: <sip:192.168.135.7>;tag=4be2b15f-1274434741635899
From: "asterisk" <sip:asterisk@192.168.135.90>;tag=as01a0bd91
CSeq: 102 OPTIONS
Accept: application/sdp, application/xml, application/reginfo+xml, application/ISUP
Accept-Language: en
Date: Fri, 21 May 2010 09:39:01 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: 100rel, precondition, timer, sec-agree
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '789e513760d5f343578d1fc53226bec8@192.168.135.90' Method: OPTIONS

<--- SIP read from UDP:192.168.135.90:5061 --->

Aserisk files are still the same as shown previously, so only the response from the P-CSCF is different for an other reason.

Thank you in advance.

François Bourdoulous.

4

Hi François,

Did u managed to succeed this issue? if yes i would be needing your help to achieve the same in my project.

Request you just send a hi to my mail id - santhoshc86@yahoo.co.in else please suggest in same thread.

Thanks in advance!!

Santhosh