[asterisk-bugs] [JIRA] (ASTERISK-22901) Originate command doesn't work through OOH323 channels

Cyril CONSTANTIN (JIRA) noreply at issues.asterisk.org
Wed Nov 27 03:52:03 CST 2013


    [ https://issues.asterisk.org/jira/browse/ASTERISK-22901?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=212294#comment-212294 ] 

Cyril CONSTANTIN commented on ASTERISK-22901:
---------------------------------------------

Hi,

Instead of using test message handler for sending command from gtalk, I just made a test from context used by my SIP softphone and dialed the extension "test" so directly SIP to OOH323 via Originate command but it still doesn't work see below:

{noformat}
[lab-cyril]
exten => test,1,Originate(OOH323/00625081954 at Avaya,exten,bridge,s,1)

[bridge]
exten => s,1,ConfBridge(118218)
{noformat}

I got no sound for 30 seconds see below logs from console it looks it try to make a call through OOH323 but nothing happens:

{noformat}
<------------>
    -- Executing [test at from-sip:1] Originate("SIP/46000-00000cfe", "OOH323/00625081954 at Avaya,exten,bridge,s,1") in new stack
---   ooh323_request - data 00625081954 at Avaya format (slin192)
---   ooh323_alloc
+++   ooh323_alloc
---   find_peer "Avaya"
                comparing with "10.147.9.64"
                comparing with "10.147.9.64"
                found matching peer
+++   find_peer "Avaya"
---   ooh323_new - Avaya
+++   h323_new
---   onNewCallCreated b0e0e898: ooh323c_o_11770
---   find_call
+++   find_call
 Outgoing call Avaya(ooh323c_o_11770) - Codec prefs - (alaw)
        Adding capabilities to call(outgoing, ooh323c_o_11770)
        Adding g711 alaw capability to call(outgoing, ooh323c_o_11770)
---   configure_local_rtp
+++   configure_local_rtp
+++   onNewCallCreated ooh323c_o_11770
+++   ooh323_request
---   ooh323_call- 00625081954 at Avaya
+++   ooh323_call
---   onOutgoingCall b0e0e898: ooh323c_o_11770
---   find_call
+++   find_call
+++   onOutgoingCall ooh323c_o_11770
{noformat}

If I do a Dial(OOH323/00625081954 at Avaya) it works great, so I don't know what happens with Originate but it looks that something is not properly done when using OOH323.

Below an example when I'm replacing Originate by Dial:
{noformat}
    -- Executing [test at from-sip:1] Dial("SIP/46000-00000d16", "OOH323/00625081954 at Avaya") in new stack
---   ooh323_request - data 00625081954 at Avaya format (alaw)
---   ooh323_alloc
+++   ooh323_alloc
---   find_peer "Avaya"
                comparing with "10.147.9.64"
                comparing with "10.147.9.64"
                found matching peer
+++   find_peer "Avaya"
---   ooh323_new - Avaya
+++   h323_new
---   onNewCallCreated b69326e8: ooh323c_o_11830
---   find_call
+++   find_call
 Outgoing call Avaya(ooh323c_o_11830) - Codec prefs - (alaw)
        Adding capabilities to call(outgoing, ooh323c_o_11830)
        Adding g711 alaw capability to call(outgoing, ooh323c_o_11830)
---   configure_local_rtp
+++   configure_local_rtp
+++   onNewCallCreated ooh323c_o_11830
+++   ooh323_request
----- ooh323_queryoption 16 on channel OOH323/Avaya-11996
+++++ ooh323_queryoption 16 on channel OOH323/Avaya-11996
+++ ooh323  get_rtp_peer 
ooh323_get_rtp_peer  OOH323/Avaya-11996 -> (null):0, 2
--- ooh323  get_rtp_peer, res = 2
---   ooh323_call- 00625081954 at Avaya
+++   ooh323_call
    -- Called OOH323/00625081954 at Avaya
---   onOutgoingCall b69326e8: ooh323c_o_11830
---   find_call
+++   find_call
setting callid number 40075
+++   onOutgoingCall ooh323c_o_11830
Reliably Transmitting (no NAT) to 10.147.116.39:51922:
OPTIONS sip:46000 at 10.147.116.39:51922;rinstance=24e333eff42d355e;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.147.113.73:5060;branch=z9hG4bK2f5ff31b
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.147.113.73>;tag=as30791647
To: <sip:46000 at 10.147.116.39:51922;rinstance=24e333eff42d355e;transport=udp>
Contact: <sip:asterisk at 10.147.113.73:5060>
Call-ID: 4ac5b3857a35181c5efd63487905197b at 10.147.113.73:5060
CSeq: 102 OPTIONS
User-Agent: 118218
Date: Wed, 27 Nov 2013 09:40:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---

<--- SIP read from UDP:10.147.116.39:51922 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.147.113.73:5060;branch=z9hG4bK2f5ff31b
Contact: <sip:10.147.116.39:51922>
To: <sip:46000 at 10.147.116.39:51922;rinstance=24e333eff42d355e;transport=udp>;tag=103c076a
From: "asterisk"<sip:asterisk at 10.147.113.73>;tag=as30791647
Call-ID: 4ac5b3857a35181c5efd63487905197b at 10.147.113.73:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces, eventlist
User-Agent: Bria 3 release 3.5.5 stamp 71238
Allow-Events: hold, talk
Content-Length: 0
{noformat}


When comparing two traces between Originate and Dial I have differences below:

On Originate:
{noformat}
ooh323_request - data 00625081954 at Avaya format (slin192)
{noformat}

On Dial I have all those line in more compared to Originate:
{noformat}
ooh323_request - data 00625081954 at Avaya format (alaw)
----- ooh323_queryoption 16 on channel OOH323/Avaya-11996
+++++ ooh323_queryoption 16 on channel OOH323/Avaya-11996
+++ ooh323  get_rtp_peer 
ooh323_get_rtp_peer  OOH323/Avaya-11996 -> (null):0, 2
--- ooh323  get_rtp_peer, res = 2
-- Called OOH323/00625081954 at Avaya
setting callid number 40075
{noformat}


Also format used by Originate was slin192 when Dial used alaw don't know if it can cause the issue. As you can see above there is lot more thing happening when using Dial command.

I have added a screenshot of my comparison between both logs.

Best Regards
                
> Originate command doesn't work through OOH323 channels
> ------------------------------------------------------
>
>                 Key: ASTERISK-22901
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22901
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Addons/chan_ooh323
>    Affects Versions: 11.6.0
>            Reporter: Cyril CONSTANTIN
>            Assignee: Alexander Anikin
>            Severity: Minor
>         Attachments: originateooh323.rar
>
>
> Hi Guys,
> When using CMD Originate into my dialplan to dial a number through OOH323 call couldn't be establish, when using tcpdump I can't see call trying to be establish with Avaya. 
> I don't have this problem with CMD Originate when doing it with SIP peer.
> Also I'm already using OOH323 with Dial command and it works great, when looking into console log with ooh323 debug activated it looks that it try to make a call through ooh323 channel but nothing happens behind, even on Avaya I can't see anything coming from Asterisk.
> See below what I'm using for CMD Originate with OOH323 (not working):
> {noformat}
> same => n,Originate(OOH323/00625081954 at Avaya,exten,bridgetwoparty,s,1)
> {noformat}
> Below when I'm dialing through OOH323 (working):
> {noformat}
> same => n,Dial(OOH323/00625081954 at Avaya)
> {noformat}
> Below when I'm using CMD Originate with SIP peer (working):
> {noformat}
> same => n,Originate(SIP/46000,exten,bridgetwoparty,s,1)
> {noformat}
> I'll attach tcpdump, debug, full, messages, console logs.
> Best Regards

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