Connection dropped by sip gateway

Asterisk Asterisk Support
1

Hi List!

I have a problem with a sip provider, sending sip MESSAGE messages without a good reason. I don’t know how I could handle them in a way that the provider is happy with it. The problem is that when asterisk sends him the 415 error he just sends me a BYE next.

I am calling an ISDN number through the provider and the messages in question seems to be charge information. I tried to turn it off on the providers side, but there is no way to suppress it.

Here is the way it goes. First I get an OPTIONS request from the other side, asking for the options for the extension the provider issued to me, but at my host. I added myself to my extensions.conf this way:

[options]
exten => PROVIDER_NUMBER,1,NoOp()

Now the proxy get’s some OPTIONS for this pair, but definitly not indicating MESSAGE capabilties.

<-- SIP read from 212.144.24.38:5060:
OPTIONS sip:PROVIDER_NUMBER@MY_HOST_IP SIP/2.0
Via: SIP/2.0/UDP 212.144.24.38:5060;branch=z9hG4bKlct5k81080u1uocqe 341sh00000l1.1
To: “MY_NAME” sip:PROVIDER_NUMBER@VORWAHL.sip.arcor.de;tag=as58e5 b2dd
From: sip:CALLED_NUMBER@VORWAHL.sip.arcor.de;ta g=SD6trua99-0d80bc15
Call-ID: 0b551c9a39a78edd7c9efc0953a9b143@VORWAHL.sip.arcor .de
CSeq: 106 OPTIONS
Max-Forwards: 69
Content-Length: 0

Looking for PROVIDER_NUMBER in default (domain MY_HOST_IP)
Transmitting (no NAT) to 212.144.24.38:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.144.24.38:5060;branch=z9hG4bKlct5k81080u1uocqe 341sh00000l1.1;received=212.144.24.38
From: sip:CALLED_NUMBER@VORWAHL.sip.arcor.de;ta g=SD6trua99-0d80bc15
To: “MY_NAME” sip:PROVIDER_NUMBER@VORWAHL.sip.arcor.de;tag=as58e5 b2dd
Call-ID: 0b551c9a39a78edd7c9efc0953a9b143@VORWAHL.sip.arcor .de
CSeq: 106 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:PROVIDER_NUMBER@MY_HOST_IP
Accept: application/sdp
Content-Length: 0

Anyway, the other side sends us this a few moments later.

<-- SIP read from 212.144.24.38:5060:
MESSAGE sip:PROVIDER_NUMBER@MY_HOST_IP SIP/2.0
Via: SIP/2.0/UDP 212.144.24.38:5060;branch=z9hG4bKlct5k81080u1uocqe 341sn0000gl1.1
To: “MY_NAME” sip:PROVIDER_NUMBER@VORWAHL.sip.arcor.de;tag=as58e5 b2dd
From: sip:CALLED_NUMBER@VORWAHL.sip.arcor.de;ta g=SD6trua99-0d80bc15
Call-ID: 0b551c9a39a78edd7c9efc0953a9b143@VORWAHL.sip.arcor .de
CSeq: 107 MESSAGE
Max-Forwards: 69
P-AoC: Info, type=AOC-D
Content-Type: ASN1/aoc
Content-Length: 20

Transmitting (no NAT) to 212.144.24.38:5060:
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/UDP 212.144.24.38:5060;branch=z9hG4bKlct5k81080u1uocqe 341sn0000gl1.1;received=212.144.24.38
From: sip:CALLED_NUMBER@VORWAHL.sip.arcor.de;ta g=SD6trua99-0d80bc15
To: “MY_NAME” sip:PROVIDER_NUMBER@VORWAHL.sip.arcor.de;tag=as58e5 b2dd
Call-ID: 0b551c9a39a78edd7c9efc0953a9b143@VORWAHL.sip.arcor .de
CSeq: 107 MESSAGE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

Afterwards I just get a bye from the other side.

Is asterisk causing this hangup, or is it by the the proxy? Is there a way to make asterisk answer 200 OK and redirect the MESSAGE to /dev/null or somewhere?

Is it an error on the proxy side to send the MESSAGE after receiving the correct OPTIONS - or is there an error in the communication that I can’t spot and that makes it impossible for the proxy to get it right?

Any ideas?

The pbx is running on Debian Etch with version 1.2.13.

Cheers,

Hein

2

Isn’t the current SIP-implementation in Asterisk a mess? I read something about it a while ago. And asterisk.org/node/117 talks about fixing this.

I know my pstn-provider run the Sip Express Router (SER) iptel.org/ser/

Maybe you can run this localy, and use your Asterisk to connect to SER.

I havn’t done this myself fortunatly my provider could offer me IAX.

Greetings

3

Hi,

I have experienced the same problems as HeinMueck. The hangup is caused by chan_sip because it is only accepting text/plain as message content type.

channels/chan_sip.c (search for “415”, somewhere around line 7521)

        if (strcmp(content_type, "text/plain")) { /* No text/plain attachment */
                transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
                ast_set_flag(p, SIP_NEEDDESTROY);
                return;
        }

No, not good enough! :-/

Here is my hack: It sends “202 Accepted” to Arcor and ignores the message.
Patch:
http://www.icip.de/download/asterisk-arcor.diff

Warning: This is in no way standard-compliant and might break other things.

Greetings,
Rainer

4

Thanks a lot RaptorBlue :smile: This patch also works for Alice.

It seems Alice and Arcor are sending these OPTIONS to check whether the other end is still alive. They expect a “200 OK” or “202 Accepted” and otherwise hang up.

Alice is expecting their customers to use their crappy little white hardware-box and thus does not offer any support at all. So I guess this is only an issue to a rather small group of people. Still, is there any chance to get this patch into the mainstream version of asterisk?