Nothing received from Asterisk after registration

Jan Kaláb

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Oct 24, 2014, 6:27:29 AM10/24/14

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Hi,

I'm trying to use SIP.js (0.6.3) with our Asterisk PBX. The initial registration works, but I don't see any incoming packets from Asterisk after that. Especially OPTIONS.

Registration:

new SIP.UA({

uri: data.device + '@' + location.hostname,

password: data.pass,

wsServers: ['wss://' + location.hostname + ':8089/ws'],

register: true,

hackIpInContact: true,

hackViaTcp: false,

log: {

builtinEnabled: false

}

});

Which result's in following packet:

REGISTER sip:xxxxx.xxxxx.xxx SIP/2.0

Via: SIP/2.0/WSS 192.0.2.24;branch=z9hG4bK8268194

Max-Forwards: 70

To: <sip:web_103_54...@xxxxx.xxxxx.xxx>

From: <sip:web_103_54...@xxxxx.xxxxx.xxx>;tag=nf6lfg11f4

Call-ID: eccflapfiouvmsvtrqip0i

CSeq: 94 REGISTER

Authorization: Digest algorithm=MD5, username="web_103_543d42cd67433", realm="pbx", nonce="7950a459", uri="sip:xxxxx.xxxxx.xxx", response="1670cea38fff046564b9eb62285c5812"

Contact: <sip:kv2q...@192.0.2.24;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:bd248ba4-1e31-45d1-8cfe-a74fbf8ca716>";expires=600

Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE

Supported: path,gruu,outbound

User-Agent: SIP.js/0.6.3

Content-Length: 0

And I get this response:

SIP/2.0 200 OK

Via: SIP/2.0/WSS 192.0.2.24;branch=z9hG4bK8268194;received=xxx.xxx.xxx.xxx;rport=55361

From: <sip:web_103_54...@xxxxx.xxxxx.xxx>;tag=nf6lfg11f4

To: <sip:web_103_54...@xxxxx.xxxxx.xxx>;tag=as1f9da8c0

Call-ID: eccflapfiouvmsvtrqip0i

CSeq: 94 REGISTER

Server: ipbx 3.0.14

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Expires: 600

Contact: <sip:kv2q...@192.0.2.24;transport=ws>;expires=600

Date: Fri, 24 Oct 2014 10:13:43 GMT

Content-Length: 0

And silence after that.

I tried all combinations of hackIpInContact and hackViaTcp, but none works. It works with SIPML5 demo phone, so It's not problem in Asterisk.

Jan Kaláb

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Oct 24, 2014, 6:45:19 AM10/24/14

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In other words, this is what we need to work: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

Jan Kaláb

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Oct 24, 2014, 7:48:30 AM10/24/14

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Ok, it's this bug in Asterisk. As workaround, find transport:"ws" in sip.min.js and change it to wss.

Dne pátek, 24. října 2014 12:27:29 UTC+2 Jan Kaláb napsal(a):

Hi,
I'm trying to use SIP.js (0.6.3) with our Asterisk PBX. The initial registration works, but I don't see any incoming packets from Asterisk after that. Especially OPTIONS.

Registration:
new SIP.UA({
uri: data.device + '@' + location.hostname,
password: data.pass,
wsServers: ['wss://' + location.hostname + ':8089/ws'],
register: true,
hackIpInContact: true,
hackViaTcp: false,
log: {
builtinEnabled: false
}
});

Which result's in following packet:

REGISTER sip:xxxxx.xxxxx.xxx SIP/2.0
Via: SIP/2.0/WSS 192.0.2.24;branch=z9hG4bK8268194
Max-Forwards: 70

To: <sip:web_103_543d42cd67433@xxxxx.xxxxx.xxx>
From: <sip:web_103_543d42cd67433@xxxxx.xxxxx.xxx>;tag=nf6lfg11f4

Call-ID: eccflapfiouvmsvtrqip0i
CSeq: 94 REGISTER
Authorization: Digest algorithm=MD5, username="web_103_543d42cd67433", realm="pbx", nonce="7950a459", uri="sip:xxxxx.xxxxx.xxx", response="1670cea38fff046564b9eb62285c5812"
Contact: <sip:kv2q...@192.0.2.24;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:bd248ba4-1e31-45d1-8cfe-a74fbf8ca716>";expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE
Supported: path,gruu,outbound
User-Agent: SIP.js/0.6.3
Content-Length: 0

And I get this response:

SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.24;branch=z9hG4bK8268194;received=xxx.xxx.xxx.xxx;rport=55361

From: <sip:web_103_543d42cd67433@xxxxx.xxxxx.xxx>;tag=nf6lfg11f4
To: <sip:web_103_543d42cd67433@xxxxx.xxxxx.xxx>;tag=as1f9da8c0

Call-ID: eccflapfiouvmsvtrqip0i
CSeq: 94 REGISTER
Server: ipbx 3.0.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:kv2q...@192.0.2.24;transport=ws>;expires=600
Date: Fri, 24 Oct 2014 10:13:43 GMT
Content-Length: 0

And silence after that.

Eric Green

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Oct 24, 2014, 3:25:42 PM10/24/14

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Hi Jan,

I pushed a fix to the SIP.js master branch that allows you to set via a UA configuration parameter. Let me know if that works for you.

-Eric Green

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