Wrong country prefix for sip INVITE

system · May 19, 2012, 10:30am

Hi there,

i’am very new in using asterisk pbx. But there is still a big problem since my sip provider changed his platform last week.

When ever I have a sip call - after the first INVITE the SIP call will be terminated from the provider because of a “SIP/2.0 404 Not Found”.

That error is caused by a wrong country prefix in the sip protocol which asterisk sends:
[…]
From: sip:+4930XXXXXXXX@proxy.bellsip.com;tag=as1cf5e2e0
To: “+4930XXXXXXX” sip:030XXXXXXX@proxy.bellsip.com;tag=as5e47f603
[…]
(I replaced the private numbers after the are code.)

But what anybody can see is, that asterisk didn’t send SIP address with country prefix and the “+”.
How can I affect asterisk to do so?

Thank you very much in advance.

Regards,

Sascha

david55 · May 19, 2012, 12:48pm

What gets sent depends on the dial plan. Asterisk does not come with a production dial plan. You will need tell us what is in your dial plan and what you actually dial on the phone. Alternatively, if you didn’t write the dial plan, you should take this up with the support services of whoever did.

Basically, you need to get the dialplan to do …,n,Dial(SIP/peer/+4930XXXXXXX)

(One thing I don’t really understand is how you set the human friendly part of the address.)

system · May 19, 2012, 1:14pm

Thank you david,

the diaplan has only an answer in it. Because the most call are incoming calls.

I’am dialing in to my asterisk pbx from the fixed german phone network. I have a sip provider as a media gateway.
When a phone call from the fixed phone network is introduced to the asterisk the provider invites that way:
<— Reliably Transmitting (NAT) to 178.248.240.36:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.248.240.36;branch=z9hG4bKe318.e09c1cf5.0;received=178.248.240.36;rport=5060
Via: SIP/2.0/UDP 93.189.170.12:5060;received=93.189.170.12;branch=z9hG4bK49394778;rport=5060
Record-Route: sip:178.248.240.36;lr=on;ftag=as5e47f603;did=e62.90fa22f6;vsf=AAAAABsHCQUGAQEFAgcDdUYyVktXH1pcQl1EWV4fUW9t
From: “+4930XXXXXXXX” sip:030XXXXXXXX@proxy.bellsip.com;tag=as5e47f603
To: sip:+4930XXXXXXXX@proxy.bellsip.com;tag=as1cf5e2e0
Call-ID: 414ce2150480607212ecf213448ddb0c@93.189.170.12
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 160;refresher=uas
Contact: sip:s@78.47.106.233:5060
Content-Type: application/sdp
Content-Length: 308

v=0
o=root 2116324476 2116324476 IN IP4 78.47.106.233
s=Asterisk PBX 1.8.12.0
c=IN IP4 78.47.106.233
t=0 0
m=audio 16936 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99

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

<— SIP read from UDP:178.248.240.36:5060 —>
ACK sip:s@78.47.106.233:5060 SIP/2.0
Record-Route: sip:178.248.240.36;lr=on;ftag=as5e47f603
Via: SIP/2.0/UDP 178.248.240.36;branch=z9hG4bKe318.e09c1cf5.2
Via: SIP/2.0/UDP 93.189.170.12:5060;received=93.189.170.12;branch=z9hG4bK31bc162c;rport=5060
Route: sip:178.248.240.36;lr=on;ftag=as5e47f603;did=e62.90fa22f6;vsf=AAAAABsHCQUGAQEFAgcDdUYyVktXH1pcQl1EWV4fUW9t
Max-Forwards: 69
From: “+4930XXXXXXXX” sip:+4930XXXXXXXX@93.189.170.12;tag=as5e47f603
To: sip:+4930XXXXXXXX@proxy.bellsip.com;tag=as1cf5e2e0
Contact: sip:+4930XXXXXXXX@93.189.170.12;received="sip:93.189.170.12:5060"
Call-ID: 414ce2150480607212ecf213448ddb0c@93.189.170.12
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7/ss7_build_100507 Cosini SS7 Gateway
Content-Length: 0

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

You can see that the provider routes my incoming call’s for the my asterisk from 178.248.240.36 to 93.189.170.12.
178.248.240.36 is the IP of my SIP provider.

But when asterisk asked for the INVITE from my asterisk number to the caller number I get a 404 from the provider:
[…]
From: sip:+4930XXXMYASTERISKNUMBERXXX@proxy.bellsip.com;tag=as1cf5e2e0
To: “+4930XXXCALLERNUMBERXXX” sip:030XXXCALLERNUMBERXXX@proxy.bellsip.com;tag=as5e47f603
[…]

Response:
<— SIP read from UDP:178.248.240.36:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 78.47.106.233:5060;branch=z9hG4bK5a1c13d5;rport=5060
From: sip:+4930XXXMYASTERISKNUMBERXXX@proxy.bellsip.com;tag=as1cf5e2e0
To: “+4930XXXCALLERNUMBERXXX” sip:030XXXCALLERNUMBERXXX@proxy.bellsip.com;tag=as5e47f603
Call-ID: 414ce2150480607212ecf213448ddb0c@93.189.170.12
CSeq: 102 INVITE
Content-Length: 0

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

Thanks a lot for you help.

Regards,
Sascha

david55 · May 19, 2012, 1:34pm

That clearly isn’t true, as you would not get the INVITE sent if the dialplan didn’t include at least one of Dial, Queue or Page (hope that is all).

I therefore repeat, please provide a copy of your dialplan!

system · May 19, 2012, 1:50pm

Ok. Here is my dial plan. Easy meet me app request.

[default]
exten => _.,1,Answer()
exten => _.,n,Playback(conf-hello)
exten => _.,n,Meetme(1111)

Regards,
Sascha

david55 · May 19, 2012, 2:23pm

That dialplan is not capable of making outgoing calls. It can’t even attempt them.

david55 · May 19, 2012, 2:36pm

Incidentally, your provider is also using Asterisk, although an older version.

Also, _. is not a safe pattern, because it matches h and other special extension numbers.

Also, you cut out important information, e.g. I can’t tell if the failing operation is actually a re-invite.

system · May 19, 2012, 3:41pm

Hi David,

thanks for the hints. I replaced the “_.” in the extension with my number from the provider.

Here is a hole snippet of the failure:
<— SIP read from UDP:178.248.240.36:5060 —>
INVITE sip:versammlung@78.47.106.233:5060 SIP/2.0
Record-Route: sip:178.248.240.36;lr=on;ftag=as177c889f;did=603.2fca1c2;vsf=AAAAABsFDgcDDwcAAgkHAXZBMlZLVx9aXEJdRFleH1FvbQ--
Via: SIP/2.0/UDP 178.248.240.36;branch=z9hG4bK6d1.743f14c3.1
Via: SIP/2.0/UDP 93.189.170.12:5060;received=93.189.170.12;branch=z9hG4bK58265d63;rport=5060
Max-Forwards: 69
From: “+49176<>” <sip:0176<>@proxy.bellsip.com>;tag=as177c889f
To: <sip:+4903<>@proxy.bellsip.com>
Contact: <sip:+49176<>@93.189.170.12>;received="sip:93.189.170.12:5060"
Call-ID: 41c369c51bc9c9fb698265c9457c0132@93.189.170.12
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7/ss7_build_100507 Cosini SS7 Gateway
Date: Sat, 19 May 2012 15:24:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 440
P-Asserted-Identity: <sip:0176<>@proxy.bellsip.com;user=phone>
Session-Expires: 160

v=0
o=root 1608746136 1608746136 IN IP4 178.248.240.36
s=Asterisk PBX 1.6.2.7/ss7_build_100507 Cosini SS7 Gateway
c=IN IP4 178.248.240.36
b=CT:384
t=0 0
m=audio 41598 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
m=video 35274 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
a=nortpproxy:yes
<------------->
— (18 headers 19 lines) —
Sending to 178.248.240.36:5060 (NAT)
Using INVITE request as basis request - 41c369c51bc9c9fb698265c9457c0132@93.189.170.12
Found peer ‘+4903<>’ for ‘0176<>’ from 178.248.240.36:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0xe (gsm|ulaw|alaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x20000e (gsm|ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 178.248.240.36:41598
Looking for versammlung in konferenz_in_listener (domain 78.47.106.233)

<— Reliably Transmitting (no NAT) to 178.248.240.36:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 178.248.240.36;branch=z9hG4bK6d1.743f14c3.1;received=178.248.240.36
Via: SIP/2.0/UDP 93.189.170.12:5060;received=93.189.170.12;branch=z9hG4bK58265d63;rport=5060
From: “+49176<>” <sip:0176<>@proxy.bellsip.com>;tag=as177c889f
To: <sip:+4903<>@proxy.bellsip.com>;tag=as6026088c
Call-ID: 41c369c51bc9c9fb698265c9457c0132@93.189.170.12
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[May 19 17:24:25] NOTICE[25674]: chan_sip.c:22670 handle_request_invite: Call from ‘circumconcepts.versammlung’ (178.248.240.36:5060) to extension ‘versammlung’ rejected because extension not found in context ‘konferenz_in_listener’.
Scheduling destruction of SIP dialog ‘41c369c51bc9c9fb698265c9457c0132@93.189.170.12’ in 32000 ms (Method: INVITE)
– Executing [+4903<>@konferenz_in_listener:1] Answer(“SIP/+4903<>-00000000”, “”) in new stack
Audio is at 16694
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 178.248.240.36:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.248.240.36;branch=z9hG4bK6d1.743f14c3.0;received=178.248.240.36
Via: SIP/2.0/UDP 93.189.170.12:5060;received=93.189.170.12;branch=z9hG4bK58265d63;rport=5060
Record-Route: sip:178.248.240.36;lr=on;ftag=as177c889f;did=603.2fca1c2;vsf=AAAAABsFDgcDDwcAAgkHAXZBMlZLVx9aXEJdRFleH1FvbQ--
From: “+49176<>” <sip:0176<>@proxy.bellsip.com>;tag=as177c889f
To: <sip:+4903<>@proxy.bellsip.com>;tag=as72d9f4a3
Call-ID: 41c369c51bc9c9fb698265c9457c0132@93.189.170.12
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 160;refresher=uas
Contact: <sip:+4903<>@78.47.106.233:5060>
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 791093192 791093192 IN IP4 78.47.106.233
s=Asterisk PBX 1.8.12.0
c=IN IP4 78.47.106.233
t=0 0
m=audio 16694 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99

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

<— SIP read from UDP:178.248.240.36:5060 —>
ACK sip:versammlung@78.47.106.233:5060 SIP/2.0
Via: SIP/2.0/UDP 178.248.240.36;branch=z9hG4bK6d1.743f14c3.1
From: “+49176<>” <sip:0176<>@proxy.bellsip.com>;tag=as177c889f
Call-ID: 41c369c51bc9c9fb698265c9457c0132@93.189.170.12
To: <sip:+4903<>@proxy.bellsip.com>;tag=as6026088c
CSeq: 102 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘41c369c51bc9c9fb698265c9457c0132@93.189.170.12’ Method: ACK

<— SIP read from UDP:178.248.240.36:5060 —>
ACK sip:+4903<>@78.47.106.233:5060 SIP/2.0
Record-Route: sip:178.248.240.36;lr=on;ftag=as177c889f
Via: SIP/2.0/UDP 178.248.240.36;branch=z9hG4bK6d1.743f14c3.3
Via: SIP/2.0/UDP 93.189.170.12:5060;received=93.189.170.12;branch=z9hG4bK02a4e25e;rport=5060
Route: sip:178.248.240.36;lr=on;ftag=as177c889f;did=603.2fca1c2;vsf=AAAAABsFDgcDDwcAAgkHAXZBMlZLVx9aXEJdRFleH1FvbQ--
Max-Forwards: 69
From: “+49176<>” <sip:+49176<>@93.189.170.12>;tag=as177c889f
To: <sip:+4903<>@proxy.bellsip.com>;tag=as72d9f4a3
Contact: <sip:+49176<>@93.189.170.12>;received="sip:93.189.170.12:5060"
Call-ID: 41c369c51bc9c9fb698265c9457c0132@93.189.170.12
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7/ss7_build_100507 Cosini SS7 Gateway
Content-Length: 0

<------------->
— (13 headers 0 lines) —
– Executing [+4903<>@konferenz_in_listener:2] Set(“SIP/+4903<>-00000000”, “VOLUME(TX)=2”) in new stack
– Executing [+4903<>@konferenz_in_listener:3] Playback(“SIP/+4903<>-00000000”, “conf-hello”) in new stack
– <SIP/+4903<>-00000000> Playing ‘conf-hello.slin’ (language ‘de’)
[May 19 17:24:25] NOTICE[25703]: channel.c:4169 __ast_read: Dropping incompatible voice frame on SIP/+4903<>-00000000 of format ulaw since our native format has changed to 0x200008 (alaw|h264)