alticelabsprojects/asterisk-i

SIP-I support for Asterisk

asterisk-i's Issues

BYE is not sent after REFER for unattended transfers

When a REFER is sent but the call transfer is not completed or accepted by the destination the original session hangs.

WMS-382: 1.0.0
WMS-626: 1.0.1

amr file format not supported

amr file extension is not registered.
Asterisk-i should incorporate patch https://issues.asterisk.org/jira/secure/attachment/28355/amr-passthrough-trunk-r82546M.txt for formats/format_amrnb.c file

RFC2833 payload type on 200 OK doesn't match remote peer's INVITE/SDP

Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response, this defect was corrected in https://issues.asterisk.org/jira/browse/ASTERISK-23279 patch http://svnview.digium.com/svn/asterisk?view=revision&revision=408729

Makecall timeout returns cause error 16

When makecall ends with timeout, it should be returned cause 19 instead of cause 16.

Fax returns resource error when dialing to non-fax terminal

Fax returns resource error when dialing to non-fax terminal

ISUP IAM

Add support for outbound SIP-I calls.

The system should send an ISUP IAM message embedded in the SIP INVITE message body

WMS responds with duplicated header "require: timer"

Invite example:

INVITE sip:[email protected]:5060 SIP/2.0
CSeq: 1 INVITE
From: <sip:[email protected].180>;tag=1
To: <sip:[email protected].220>
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.112.137.180:6060
User-Agent: X-Lite release 1011s stamp 41150
Max-Forwards: 70
Date: Sat, 13 Nov 2010 23:29:00 GMT
Contact: <sip:[email protected]:6060>
Allow: REGISTER,REFER,NOTIFY,SUBSCRIBE,INVITE,ACK,OPTIONS,CANCEL,BYE
Max-Forwards: 70
Min-SE: 900
Session-Expires: 1800
Supported: timer

Response:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.112.137.180:6060;received=10.112.137.180
From: <sip:[email protected].180>;tag=1
To: <sip:[email protected].220>;tag=as727c2726
Call-ID: [email protected]
CSeq: 1 INVITE
Server: WMS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, PRACK
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Require: timer
Content-Type: application/sdp
Require: timer

After PRACK transaction the initial dialog branch is lost

After PRACK transaction the initial dialog branch is lost

Section 171.1.3 (Construction of the ACK Request) da RFC 3261:

This section specifies the construction of ACK requests sent within
the client transaction. A UAC core that generates an ACK for 2xx
MUST instead follow the rules described in Section 13.
The ACK request constructed by the client transaction MUST contain
values for the Call-ID, From, and Request-URI that are equal to the
values of those header fields in the request passed to the transport
by the client transaction (call this the "original request"). The To
header field in the ACK MUST equal the To header field in the
response being acknowledged, and therefore will usually differ from
the To header field in the original request by the addition of the
tag parameter. The ACK MUST contain a single Via header field, and
this MUST be equal to the top Via header field of the original
request. The CSeq header field in the ACK MUST contain the same
value for the sequence number as was present in the original request,
but the method parameter MUST be equal to "ACK".

Buffer overflow handling SIP session on RHEL6

When handling SIP sessions on Red Hat EL6, asterisk-i terminates with a core dump due to a buffer overflow when copying data to url_params.

WMS callid not set on SIP Call-ID header

WMS callid not set on SIP Call-ID header, asterisk-i is generating a new Call-ID instead of using the wms callid

SIP BYE with message body following RFC 5552 4.2

Add support for RFC 5552 section 4.2 (SIP Mechanism) to allow the delivery of digits results (url encoded string) in the message body of the SIP BYE message.

A re-INVITE with inband DTMF is not enabling inband detection

If the remote peer re-invites to inband after rfc2833 in place, the dsp for dtmf detection is never enabled

Wrap up 2.0 sprints

Wrap up with isues:

1. #6
2. #8
3. #9
4. #10
5. #4

After calling send_fax or receive_fax, operations have no effect

It is necessary to throw a thread for fax in order to add the possibility to process operations while fax is enabled.

SIP stack block

res_timing_pthread fails to return from write, causing timer dependent operations to block indefinitely

https://issues.asterisk.org/jira/browse/ASTERISK-21389

http://svnview.digium.com/svn/asterisk/branches/11/res/res_timing_pthread.c?r2=386159

Integration with spandsp-i-1.1.0

Integration with spandsp-i-1.1.0, this version is based in the spandsp-fs source code.

Play of invalid wav file isn't reporting error to application

The play instruction on the app_wms is not validating the return value of the ast_streamfile() function. Without error WMS application will not be able to stop the call flow.

Jira PTIN: WMS-1005

Related to: issue #5

Release endpoint does not terminate makecall in progress

no early media with Session Progress

Dear Team,

asterisk is not sending out audio until the line is answered.
I do not see any RTP flowing out of asterisk while playing back an announcement with Playback (noanswer).

here is my dialplan

[noanswer_demo]
exten => s,1,Progress()
exten => s,n,Monitor(wav,testCall-${STRFTIME(${EPOCH},GMT+3,%C%y%m%d%H%M)})
exten => s,n,Background(hello-world, n) ; ----> 1
exten => s,n,WaitExten(1)
exten => s,n(cont),Playback(hello-world, noanswer) ; ----> 2
exten => s,n,Wait(1)
exten => s,n,Playback(hello-world) ; ----> 3
exten => s,n,Playback(demo-instruct)
exten => s,n,Playback(hello-world)
exten => s,n,Playback(demo-instruct)
exten => s,n,Hangup()

I am not hearing the 1st and 2nd prompts BUT the Monitor file contains all the prompts.

:::::::: Another observation ::::::::
Below is the Mime I am receiving and we are replying with SIP and not SIP-I message.

Content-Type:application/ISUP;base=itu-t92+;version=itu-t92+
Content-Disposition:signal;handling=required
Content-Transfer-Encoding:binary

i guess we are simply ignoring the handling=required

can you please guide me how can i solve these problems

NOTE:
all 51 patches applied

Prack notification of application

When a SIP PRACK with an SDP is received, the application controlling the call should be notified so that it can do media related operations.

SDP in answer to an INVITE without SDP should list RFC2833 DTMF capability

When configuring asterisk's DTMF use to either auto or rfc2833, answers to INVITES which have no SDP should state that RFC 2833 DTMF transmission is possible.

Código do issue no Jira PTIN: WMS-680

Calls to conf make sip stack to crash

Asterisk-i is generating a segfault when receiving a conference call.

Support for Fax termination

Support for Fax termination with spandsp

Missing privacy header when confidentiality is active in a call

Confidentiality value is not being propagated to the SIP INVITE. The header privacy needs to be added when confidentiality is set to 1.

A re-INVITE with inband DTMF is not enabling inband detection if faxdetect is active

Change patch file structure

Patch files should be numbered, making easier the patch process. With this approach there is no need for the README files to indicate the patch apply order.

Play of invalid wav file isn't reporting error to application

The play instruction on the app_wms is not validating the return value of the ast_streamfile() function. Without error WMS application will not be able to stop the call flow.

Jira PTIN: WMS-759

Memory leak when using conference resources

Memory leak when using conference resources

Enhanced conference support via asterisk-i ConfBridge applications

Enhanced conference support via asterisk-i ConfBridge applications

Makecall timeout returns cause error 16

When makecall ends with timeout and there isn't ringing it returns with cause 16, it should be returned cause 19 instead of cause 16.
Relates to #28

Conference participant's audio is missing after re-INVITE

After a re-INVITE that alter the port or codec, the conference participant's audio is missing.

See
https://reviewboard.asterisk.org/r/2364/

Re-invite to audio after fax call hangup

When fax call terminates with a BYE from the remote peer, a re-invite to audio is being issued causing 481 messages.
This is happening after the t38 resource and signaling synchronization patch.

Related to issue #26

A re-INVITE that disables RFC2833 DTMF should enable inband detection

Jira issue ID: WMS-683

Support for third-party rtp media

Support for thrid-party rtp media.
The SDPs provided by asterisk should be able to specify a different rtp media provider rather than the asterisk embeded.

tel URI scheme is not supported on headers different than From or To

The tel URI scheme is only supported on the From and To headers.
It should be supported in other headers such as P-Asserted-Identity.

Conference record filename is corrupted

Conference record filename is corrupted

200 OK without SDP in response to a PRACK with an offer

The 200 OK response to a PRACK with SDP (new offer) is sent without any body (answer).

It should comply with RFC 6337:

3.2.  Offer/Answer Exchange in Early Dialog
   A UA can send a PRACK request with a new offer only when
   acknowledging the reliable provisional response carrying the answer
   to an offer in the INVITE request.

4.  Exceptional Case Handling
      At any time, either agent MAY generate a new offer that updates
      the session.  However, it MUST NOT generate a new offer if it has
      received an offer which it has not yet answered or rejected.
      Furthermore, it MUST NOT generate a new offer if it has generated
      a prior offer for which it has not yet received an answer or a
      rejection.

And RFC 3262:

5 The Offer/Answer Model and PRACK
   If the UAC receives a reliable provisional response with an offer
   (this would occur if the UAC sent an INVITE without an offer, in
   which case the first reliable provisional response will contain the
   offer), it MUST generate an answer in the PRACK.  If the UAC receives
   a reliable provisional response with an answer, it MAY generate an
   additional offer in the PRACK.  If the UAS receives a PRACK with an
   offer, it MUST place the answer in the 2xx to the PRACK.

Inconsistent Via is being generated

Inconsistent Via is being generated when a semicolon is present at the end of original header

Support for SIP PRACK

Asterisk-i should be able to handle and generate SIP PRACK messages.
To this effect, a merge of the http://svn.digium.com/svn/asterisk/team/oej/darjeeling-prack-11 svn branch should be made with the current asterisk-i code.

This issue links with PTIN Jira Issue: WMS-693

Asterisk-i is answering with a=inactive

Asterisk-i is answering with a=inactive when receiving an invite without sdp

User isn't added at conference creation

When creating a conference, if two consecutive calls are made in a short period of time, one of the users may not be added to the conference.

WMS internal issue: WMS-635

Empty DNIS on INCOMING_CALL_EVENT

It is necessary to rejected the call if the RURI exceeds 256 characters.

Fax variable overflow destination buffer

File errors after call disconnected

In case a LOOP_CURRENT_EVENT when playing a announcement asterisk-i shouldn't send a PLAY_ANNOUNCEMENT_COMPLETED with error INVALID_VOICE_FILE_PARAMETER.

Support for asterisk 11.3.0

asterisk-i should be based on asterisk 11.3.0.
All the current functionalities should be available as is.

JIRA PTIN: WMS-728

Hangup and fax phase E event fails to synchronize

Hangup and fax phase E event fails to synchronize, although the fax is transmitted with success sometimes the hangup event is processed before the phase E event, when this happens it is returned an error.

Call for an unregistered mask causes a segmentation fault

A ack sip message with content-type leads to a segmentation fault, if the call is for an unregistered mask.

Asterisk-i is sending a=sendrecv, when offer was a=recvonly

Asterisk-i is sending a=sendrecv, when offer was a=recvonly. This happens when the header a= is not the last header in the sdp.

Configuration parameters for CED and CNG

The options for sending the signals CED and CNG should be configured in res_fax.conf

Offer/answer model regarding a=recvonly/sendonly is not being followed

Asterisk-i is not following offer/answer model regarding a=recvonly/sendonly, asterisk-i is not responding with "a=recvonly" when receives an invite with "a=sendonly" or vice versa.