Problem with calling "normal" numbers via SIP

reindje · #11 (**permalink**) August 14th, 2005, 07:28 PM

Sorry but even when option 4 is like 2 and 3 the same problems appear.

Maybe this helps a bit:

When i call to the other budgetphone number (option 2 and 3) sip debug say this:

--------------------------

We're at 192.168.0.1 port 18898
Answering/Requesting with root capability 0x400 (ilbc)
Answering with preferred capability 0x200 (speex)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 14 lines
Reliably Transmitting:
INVITE sip:0307110319@budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0962f4a8
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as120ae01b
To: <sip:0307110319@budgetphone.nl>
Contact: <sip:0647506476@192.168.0.1>
Call-ID: 632938914d8823f407cfe7a30e11499c@192.168.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 14 Aug 2005 19:15:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 17092 17092 IN IP4 192.168.0.1
s=session
c=IN IP4 192.168.0.1
t=0 0
m=audio 18898 RTP/AVP 97 110 8 3 0 101
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
(no NAT) to 81.23.228.150:5060
-- Called 0307110319@budgetphone.nl
linuxbak*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0962f4a8;rport=1028 ;received=145.97.216.214
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as120ae01b
To: <sip:0307110319@budgetphone.nl>
Call-ID: 632938914d8823f407cfe7a30e11499c@192.168.0.1
CSeq: 102 INVITE
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0

8 headers, 0 lines
linuxbak*CLI>

Sip read:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0962f4a8;rport=1028 ;received=145.97.216.214
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as120ae01b
To: <sip:0307110319@budgetphone.nl>
Call-ID: 632938914d8823f407cfe7a30e11499c@192.168.0.1
CSeq: 102 INVITE
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0

8 headers, 0 lines
linuxbak*CLI>

Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.1:5060;rport=1028;received=145.97.216.21 4;branch=z9hG4bK0962f4a8
Record-Route: <sip:0307110319@81.23.228.150;ftag=as120ae01b;lr=o n>
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as120ae01b
To: <sip:0307110319@budgetphone.nl>;tag=01e5fff67f9b78 b4
Call-ID: 632938914d8823f407cfe7a30e11499c@192.168.0.1
CSeq: 102 INVITE
User-Agent: Grandstream HT286 1.0.6.7
Content-Length: 0

---------------------------------------

When i let asterisk try to call my mobile number (exten => 4,1,Dial(SIP/0647506476@budgetphone.nl,40), sip debug reports this:

----------------------------------

We're at 192.168.0.1 port 15958
Answering/Requesting with root capability 0x2 (gsm)
Answering with preferred capability 0x400 (ilbc)
Answering with preferred capability 0x200 (speex)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 14 lines
Reliably Transmitting:
INVITE sip:0647506476@budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK59e3e595
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as355334fb
To: <sip:0647506476@budgetphone.nl>
Contact: <sip:0647506476@192.168.0.1>
Call-ID: 72374d1238506b38509673f4692f3730@192.168.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 14 Aug 2005 19:17:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 17096 17096 IN IP4 192.168.0.1
s=session
c=IN IP4 192.168.0.1
t=0 0
m=audio 15958 RTP/AVP 3 97 110 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
(no NAT) to 81.23.228.150:5060
-- Called 0647506476@budgetphone.nl
linuxbak*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK59e3e595;rport=1028 ;received=145.97.216.214
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as355334fb
To: <sip:0647506476@budgetphone.nl>
Call-ID: 72374d1238506b38509673f4692f3730@192.168.0.1
CSeq: 102 INVITE
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0

8 headers, 0 lines
linuxbak*CLI>

Sip read:
SIP/2.0 403 PSTN calls are forbidden
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK59e3e595;rport=1028 ;received=145.97.216.214
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as355334fb
To: <sip:0647506476@budgetphone.nl>;tag=9b5971f23d1887 2ff678d4e9dae023f8.e58d
Call-ID: 72374d1238506b38509673f4692f3730@192.168.0.1
CSeq: 102 INVITE
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0

8 headers, 0 lines
Transmitting:
ACK sip:0647506476@budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK59e3e595
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as355334fb
To: <sip:0647506476@budgetphone.nl>;tag=9b5971f23d1887 2ff678d4e9dae023f8.e58d
Contact: <sip:0647506476@192.168.0.1>
Call-ID: 72374d1238506b38509673f4692f3730@192.168.0.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 81.23.228.150:5060
Aug 14 21:17:44 WARNING[7139]: chan_sip.c:6811 handle_response: Forbidden - wrong password on authentication for INVITE to '"0647506476" <sip:0647506476@192.168.0.1>;tag=as355334fb'
-- SIP/budgetphone.nl-9f10 is circuit-busy
Reliably Transmitting:
CANCEL sip:0647506476@budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK59e3e595
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as355334fb
To: <sip:0647506476@budgetphone.nl>
Contact: <sip:0647506476@192.168.0.1>
Call-ID: 72374d1238506b38509673f4692f3730@192.168.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 81.23.228.150:5060
Scheduling destruction of call '72374d1238506b38509673f4692f3730@192.168.0.1' in 15000 ms
== Everyone is busy/congested at this time
linuxbak*CLI>

Sip read:
SIP/2.0 487 Request cancelled
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK59e3e595;rport=1028 ;received=145.97.216.214
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as355334fb
To: <sip:0647506476@budgetphone.nl>;tag=9b5971f23d1887 2ff678d4e9dae023f8.e58d
Call-ID: 72374d1238506b38509673f4692f3730@192.168.0.1
CSeq: 102 CANCEL
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0

---------------------------------------

And When i let asterisk try to call exten => 4,1,Dial(SIP/0647506476@voipwg01.budgetphone.nl,40), sip debug reports this:

----------------------------------

We're at 192.168.0.1 port 16562
Answering/Requesting with root capability 0x2 (gsm)
Answering with preferred capability 0x400 (ilbc)
Answering with preferred capability 0x200 (speex)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 14 lines
Reliably Transmitting:
INVITE sip:0647506476@voipgw01.budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1d97f3c7
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as4c777e97
To: <sip:0647506476@voipgw01.budgetphone.nl>
Contact: <sip:0647506476@192.168.0.1>
Call-ID: 248fa6e301ac36ce1bafa531274c0331@192.168.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 14 Aug 2005 19:31:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 17118 17118 IN IP4 192.168.0.1
s=session
c=IN IP4 192.168.0.1
t=0 0
m=audio 16562 RTP/AVP 3 97 110 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
(no NAT) to 212.203.28.2:5060
-- Called 0647506476@voipgw01.budgetphone.nl
Retransmitting #1 (no NAT):
INVITE sip:0647506476@voipgw01.budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1d97f3c7
From: "0647506476" <sip:0647506476@192.168.0.1>;tag=as4c777e97
To: <sip:0647506476@voipgw01.budgetphone.nl>
Contact: <sip:0647506476@192.168.0.1>
Call-ID: 248fa6e301ac36ce1bafa531274c0331@192.168.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 14 Aug 2005 19:31:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 316

and so an to #5

and then:

to 212.203.28.2:5060
Aug 14 21:32:46 WARNING[7139]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 5616dd3c76f9c3a325afb1317d62e06d@192.168.0.1 for seqno 102 (Critical Request)
== No one is available to answer at this time
Destroying call '5616dd3c76f9c3a325afb1317d62e06d@192.168.0.1'

----------------------

any suggestion??

chandave · #12 (**permalink**) August 15th, 2005, 09:27 AM

Very long reply....get out while you still can !!! ;-)

I'll address the outgoing call leg first. Then, I'll address the incoming call leg.

Outgoing:

I'll ignore the voipgw01.budgetphone.nl stuff for now and assume you only have one SIP Proxy gateway that can be used for making outgoing calls.

I was correct. Budgetphone.nl does support anonymous SIP calls to its users. At least I can call your number, 0307110319, when my fromdomain is not budgetphone.nl. This is why your calls without a definintion inside your sip.conf is succeeding. As soon as I specify my fromdomain=budgetphone.nl, the 407 Proxy Authenication Required SIP response is generated by the SER Proxy of budgetphone.nl.

My suggestion for getting outgoing calls working is this:

Add a budgetphone section to your sip.conf:

Code:

[budgetphone]
type=peer
fromdomain=budgetphone.nl
host=budgetphone.nl
username=3***7
secret=3***7
fromuser=3***7
; Next line needed if SER doesn't Authenicate when INVITEing
;insecure=very
; Next line needed to properly route incoming
context=from-budgetphone-nl

Add directed DID from budgetphone.nl into sip.conf

Code:

[general]
register => 3***7:*secret*@budgetphone.nl/3***7

Add context for internal-extensions:

Code:

[internal-extensions]
exten => s,1,Ringing
exten => s,2,Wait,3 ; Wait a second, ju
exten => s,3,Answer ; Answer the line
exten => s,4,Playback(welkom_n)
exten => s,5,Background(een_n)
exten => s,6,Background(belgie)
exten => s,7,Background(twee_n)
exten => s,8,Background(vragenbestelling_n)
exten => s,9,Background(drie_n)
exten => s,10,Background(overigevragen_n)
exten => s,11,ResponseTimeout(20)

exten => t,1,Playback(timeout)
exten => t,2,Hangup

exten => 1,1,Goto(submenu,s,1)

exten => 2,1,Background(ordernr_n)
exten => 2,2,Background(doorverbinden_n)
exten => 2,3,Dial(SIP/0307***9@budgetphone,20,rT)
exten => 2,4,Background(doorvmislukt_n)
exten => 2,5,Voicemail(1234)

exten => 3,1,Background(doorverbinden_n)
exten => 3,2,Dial(SIP/0307***19@budgetphone,20,rT)
exten => 3,3,Background(doorvmislukt_n)
exten => 3,4,Voicemail(1234)

exten => *,1,VoicemailMain(1234)

exten => 4,1,Dial(SIP/06*****@budgetphone,20,rT)

exten => 5,1,Background(nietgeldig)
exten => 5,2,Goto(s,5)

exten => 6,1,Background(nietgeldig)
exten => 6,2,Goto(s,5)

exten => 7,1,Background(nietgeldig)
exten => 7,2,Goto(s,5)

exten => 8,1,Background(nietgeldig)
exten => 8,2,Goto(s,5)

exten => 9,1,Background(nietgeldig)
exten => 9,2,Goto(s,5)

exten => 0,1,Background(nietgeldig)
exten => 0,2,Goto(s,5)

exten => #,1,Background(nietgeldig)
exten => #,2,Goto(s,5)

Add context from-budgetphone-nl to extensions.conf:

Code:

[from-budgetphone-nl]
include => internal-extensions
exten => 3***7,1,Goto(s,1)

Make sure all instances of budgetphone.nl are changed to budgetphone:
Add budgetphone.nl to your /etc/hosts
Code:
```
81.23.228.150          budgetphone.nl
```

Setting up the system this way will:

By replacing the DNS SRV resource (lookup for _sip._udp.budgetphone.nl) with a peer definition, you can control the fromdomain and authentication used when your Asterisk submits the INVITE to budgetphone.nl
Removed some delays due to DNS SRV lookups...but conversely removed the possible auto-loadbalancing that budgetphone.nl might have setup for its SIP Proxies
Ensures that the proper destination domain is used in the termination URI (budgetphone.nl instead of sip.budgetphone.nl)
Provided a context for all incoming INVITEs from budgetphone.nl

Incoming
Your original configuration did not provide a routing context and extension for incoming INVITEs originating from the budgetphone.nl VSP. As a result, when someone called your SIP URI through budgetphone.nl, Asterisk would look at the INVITE, extract the termination extension/phonenumber, and look for it in your [default] context of your extensions.conf.

Since you did not have an extension/phonenumber specified in your [default] context, Asterisk didn't know what to do with it resulting in a failure to connect or ring your phone.

With the solution above, your Asterisk will register with budgetphone.nl using the 3***7 number and the credentials specified by your "register =>" line. Also, the "/3***7" at the end of the registration instructs Asterisk that when an INVITE comes in from budgetphone.nl, it should be routed to the extension 3***7 in the context specified by the peer matching the budgetphone.nl IP address.

See ya...

d.c.