Asterisk Wiki

marcmac · Post by **marcmac** » Fri Sep 01, 2006 10:51 pm

I think I set it up that way thinking that the source would be a local extension, and the dest would be remote - I may even have had logic to enforce that - but now that you mention it, I guess the person initiating the call would know to wait, when they picked up the phone.
I'm at home, now, but I'll fix that.

adobrin · Post by **adobrin** » Sat Sep 02, 2006 11:16 pm

marc, getting similar problems here. the initial call goes through, to the destination number, then i get this in the debug:

Sending REFER to sip:SOURCENUM@ASTERISKIP:5060

Response received with client transaction id null: 200

Stray response -- dropping

Response received with client transaction id null: 200

Stray response -- dropping

Response received with client transaction id null: 200

Stray response -- dropping

Response received with client transaction id null: 200

Stray response -- dropping

Response received with client transaction id null: 200

Stray response -- dropping

Response received with client transaction id null: 200

Stray response -- dropping

Transaction Time out

TimeoutEvent Transaction Timeout

after that, the second time i attempt a call i get a TooManyListenersError and it no longer functions:

Calling from 3575 to 3

createSipStack gov.nist.javax.sip.SipStackImpl@8d41f2

Binding to ZIMBRAIP:45735

null

java.util.TooManyListenersException

at gov.nist.javax.sip.SipProviderImpl.addSipListener(SipProviderImpl.java:133)

at org.apache.jsp.zimlet.com_005fzimbra_005fasterisk.asterisk_jsp$1$Invite.init(org.apache.jsp.zimlet.com_005fzimbra_005fasterisk.asterisk_jsp:571)

at org.apache.jsp.zimlet.com_005fzimbra_005fasterisk.asterisk_jsp._jspService(org.apache.jsp.zimlet.com_005fzimbra_005fasterisk.asterisk_jsp:731)

at org.apache.jasper.runtime.HttpJspBase.service(HttpJspBase.java:97)

adobrin · Post by **adobrin** » Sat Sep 02, 2006 11:19 pm

btw, I never see the REFER at the Asterisk CLI

marcmac · Post by **marcmac** » Sat Sep 02, 2006 11:28 pm

Never seen that before, adobrin, but It looks like I'm getting a response that doesn't match the request I sent.
However, it's getting a 200 status code, which means that I'm probably missing something.
Can you post the whole session?
Actually, make an edit to the .jsp - line 192, change that block to:

if (tid == null) {

if (debug) {

System.out.println("Stray response -- dropping ");

System.out.println(response.toString());

}

return;

}
so that we see the full response that it's dropping.

adobrin · Post by **adobrin** » Sat Sep 02, 2006 11:47 pm

here's the goods:

Asterisk debug mode on

Calling from 1CELLPHONE to 3573

createSipStack gov.nist.javax.sip.SipStackImpl@863941

Binding to ZIMBRAIP:63156

REQUEST

INVITE sip:3573@ASTERISKIP:5060 SIP/2.0

Call-ID: 54c4bfad2910cbc058948e19947cee32@ZIMBRAIP

CSeq: 1 INVITE

From: ;tag=Zimbra46397

To:

Via: SIP/2.0/UDP ZIMBRAIP:63156

Max-Forwards: 70

Contact:

Content-Type: application/sdp

Call-Info: <http://www.antd.nist.gov>

Content-Length: 145
v=0

c=IN IP4 0.0.0.0

m=audio 63156 RTP/AVP 0 8 4 18

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:4 G723/8000

a=rtpmap:18 G729A/8000
REQUEST Sent to sip:3573@ASTERISKIP:5060

Response received with client transaction id gov.nist.javax.sip.stack.SIPClientTransaction@d72f08cd: 100

Trying number...

Response received with client transaction id gov.nist.javax.sip.stack.SIPClientTransaction@d72f08cd: 180

Ringing number...

Response received with client transaction id gov.nist.javax.sip.stack.SIPClientTransaction@d72f08cd: 200

Sending ACK

Invite accepted:

SIP/2.0 200 OK

Via: SIP/2.0/UDP ZIMBRAIP:63156;branch=z9hG4bKf67b5d2eba9fe0a59b6aa3d2b067cc36;received=ZIMBRAIP

From: ;tag=Zimbra46397

To: ;tag=as424288bc

Call-ID: 54c4bfad2910cbc058948e19947cee32@ZIMBRAIP

CSeq: 1 INVITE

User-Agent: US.EMPLIFY.SIP

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY

Contact:

Content-Type: application/sdp

Content-Length: 211
v=0

o=root 17164 17164 IN IP4 70.149.114.118

s=session

c=IN IP4 70.149.114.118

t=0 0

m=audio 16578 RTP/AVP 18 0

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=silenceSupp:off - - - -
REFER:

REFER sip:3573@ASTERISKIP:5060 SIP/2.0

Call-ID: 54c4bfad2910cbc058948e19947cee32@ZIMBRAIP

CSeq: 2 REFER

From: ;tag=Zimbra46397

To: ;tag=as424288bc

Via: SIP/2.0/UDP ZIMBRAIP:63156

Max-Forwards: 70

Refer-To:

Content-Length: 0

Sending REFER to sip:1CELLPHONE@ASTERISKIP:5060

Response received with client transaction id null: 200

Stray response -- dropping

SIP/2.0 200 OK

Via: SIP/2.0/UDP ZIMBRAIP:63156;branch=z9hG4bKf67b5d2eba9fe0a59b6aa3d2b067cc36;received=ZIMBRAIP

From: ;tag=Zimbra46397

To: ;tag=as424288bc

Call-ID: 54c4bfad2910cbc058948e19947cee32@ZIMBRAIP

CSeq: 1 INVITE

User-Agent: US.EMPLIFY.SIP

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY

Contact:

Content-Type: application/sdp

Content-Length: 211
v=0

o=root 17164 17164 IN IP4 70.149.114.118

s=session

c=IN IP4 70.149.114.118

t=0 0

m=audio 16578 RTP/AVP 18 0

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=silenceSupp:off - - - -
Response received with client transaction id null: 200

Stray response -- dropping

SIP/2.0 200 OK

Via: SIP/2.0/UDP ZIMBRAIP:63156;branch=z9hG4bKf67b5d2eba9fe0a59b6aa3d2b067cc36;received=ZIMBRAIP

From: ;tag=Zimbra46397

To: ;tag=as424288bc

Call-ID: 54c4bfad2910cbc058948e19947cee32@ZIMBRAIP

CSeq: 1 INVITE

User-Agent: US.EMPLIFY.SIP

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY

Contact:

Content-Type: application/sdp

Content-Length: 211
v=0

o=root 17164 17164 IN IP4 70.149.114.118

s=session

c=IN IP4 70.149.114.118

t=0 0

m=audio 16578 RTP/AVP 18 0

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=silenceSupp:off - - - -
Response received with client transaction id null: 200

Stray response -- dropping

SIP/2.0 200 OK

Via: SIP/2.0/UDP ZIMBRAIP:63156;branch=z9hG4bKf67b5d2eba9fe0a59b6aa3d2b067cc36;received=ZIMBRAIP

From: ;tag=Zimbra46397

To: ;tag=as424288bc

Call-ID: 54c4bfad2910cbc058948e19947cee32@ZIMBRAIP

CSeq: 1 INVITE

User-Agent: US.EMPLIFY.SIP

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY

Contact:

Content-Type: application/sdp

Content-Length: 211
v=0

o=root 17164 17164 IN IP4 70.149.114.118

s=session

c=IN IP4 70.149.114.118

t=0 0

m=audio 16578 RTP/AVP 18 0

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=silenceSupp:off - - - -
Response received with client transaction id null: 200

Stray response -- dropping

SIP/2.0 200 OK

Via: SIP/2.0/UDP ZIMBRAIP:63156;branch=z9hG4bKf67b5d2eba9fe0a59b6aa3d2b067cc36;received=ZIMBRAIP

From: ;tag=Zimbra46397

To: ;tag=as424288bc

Call-ID: 54c4bfad2910cbc058948e19947cee32@ZIMBRAIP

CSeq: 1 INVITE

User-Agent: US.EMPLIFY.SIP

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY

Contact:

Content-Type: application/sdp

Content-Length: 211
v=0

o=root 17164 17164 IN IP4 70.149.114.118

s=session

c=IN IP4 70.149.114.118

t=0 0

m=audio 16578 RTP/AVP 18 0

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=silenceSupp:off - - - -
Response received with client transaction id null: 200

Stray response -- dropping

SIP/2.0 200 OK

Via: SIP/2.0/UDP ZIMBRAIP:63156;branch=z9hG4bKf67b5d2eba9fe0a59b6aa3d2b067cc36;received=ZIMBRAIP

From: ;tag=Zimbra46397

To: ;tag=as424288bc

Call-ID: 54c4bfad2910cbc058948e19947cee32@ZIMBRAIP

CSeq: 1 INVITE

User-Agent: US.EMPLIFY.SIP

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY

Contact:

Content-Type: application/sdp

Content-Length: 211
v=0

o=root 17164 17164 IN IP4 70.149.114.118

s=session

c=IN IP4 70.149.114.118

t=0 0

m=audio 16578 RTP/AVP 18 0

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=silenceSupp:off - - - -
Response received with client transaction id null: 200

Stray response -- dropping

SIP/2.0 200 OK

Via: SIP/2.0/UDP ZIMBRAIP:63156;branch=z9hG4bKf67b5d2eba9fe0a59b6aa3d2b067cc36;received=ZIMBRAIP

From: ;tag=Zimbra46397

To: ;tag=as424288bc

Call-ID: 54c4bfad2910cbc058948e19947cee32@ZIMBRAIP

CSeq: 1 INVITE

User-Agent: US.EMPLIFY.SIP

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY

Contact:

Content-Type: application/sdp

Content-Length: 211
v=0

o=root 17164 17164 IN IP4 70.149.114.118

s=session

c=IN IP4 70.149.114.118

t=0 0

m=audio 16578 RTP/AVP 18 0

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=silenceSupp:off - - - -
Transaction Time out

TimeoutEvent Transaction Timeout

adobrin · Post by **adobrin** » Sat Sep 02, 2006 11:59 pm

sorry to waste your time.
the externip was set incorrectly in sip.conf, which i noticed in the message being dropped. after correcting that, the zimlet functions properly. i'm surprised that hasn't caused problems with other things.

thanks.

marcmac · Post by **marcmac** » Sun Sep 03, 2006 10:47 am

So it's working for you now? Cool!

sakilaine · Post by **sakilaine** » Sun Sep 03, 2006 1:37 pm

Hello marcmac,
Thanks for you job.

This is operationnel at 90%.

The drag an drop for the contacts do not working.

I sought and I saw that request SIP is formulated in this manner :
sip:@addressipofAsterisk
I have the impression that it do not take good the fields in the contacts.
I'm sorry for my english.
Thanks

Sakilaine

http://www.asterisk-france.net

marcmac · Post by **marcmac** » Sun Sep 03, 2006 2:35 pm

Yes, drag and drop is flakey - make sure the contact you drag has a phone number set.

James Brinkerhoff · Post by **James Brinkerhoff** » Wed Sep 06, 2006 4:22 pm

Well this thread has been a great help, and I have the zimlet mostly working now.. My major question would be: isn't it easier to just use the asterisk manager API to initate these calls rather than implementing a SIP client?
http://www.voip-info.org/wiki-Asterisk+manager+API