[asterisk-dev] Dial() flags

Nicholas Blasgen nicholas at blasgen.com
Thu Jul 12 14:56:08 CDT 2007


I really can't believe I'm having this issue and I'm almost sure it's an
issue either with my SIP phone or something to do with me.  But since I can
classify it as a possible bug, I might as well ask you.

I've got a very simple dialplan going to test some Dial() options.

300,1,Dial(SIP/josh,20,hH)

I can place the call on hold, but I can't seem to get Asterisk to notice the
'hH' flag and allow the call to be ended by pressing * (star).  Here's a
Verbose 3 log from Asterisk:


>     -- Executing [300 at internal:1] Dial("SIP/nick-0914fa88",
> "SIP/josh|20|hH") in new stack
>     -- Called josh
>     -- SIP/josh-0915e518 is ringing
>     -- SIP/josh-0915e518 answered SIP/nick-0914fa88
>     -- Packet2Packet bridging SIP/nick-0914fa88 and SIP/josh-0915e518
>     -- Started music on hold, class 'default', on SIP/josh-0915e518
>     -- Stopped music on hold on SIP/josh-0915e518
>   == Spawn extension (internal, 300, 1) exited non-zero on
> 'SIP/nick-0914fa88'


According to the Asterisk Dial() docs, the flags hH should enforce Asterisk
staying in line of the stream and Asterisk should be able to notice either
side pressing *.  But for the moment, pressing * does nothing.

In the off chance someone wants the Peer information on both clients:


>   Context      : internal
>   Subscr.Cont. : <Not set>
>   Language     :
>   AMA flags    : Unknown
>   Transfer mode: open
>   CallingPres  : Presentation Allowed, Not Screened
>   Callgroup    :
>   Pickupgroup  :
>   Mailbox      :
>   VM Extension : asterisk
>   LastMsgsSent : 32767/65535
>   Call limit   : 0
>   Dynamic      : Yes
>   Callerid     : "" <>
>   MaxCallBR    : 384 kbps
>   Expire       : 3139
>   Insecure     : no
>   Nat          : Always
>   ACL          : No
>   T38 pt UDPTL : No
>   CanReinvite  : No
>   PromiscRedir : No
>   User=Phone   : No
>   Video Support: No
>   Trust RPID   : No
>   Send RPID    : No
>   Subscriptions: Yes
>   Overlap dial : No
>   DTMFmode     : rfc2833
>   LastMsg      : 0
>   ToHost       :
>   Addr->IP     : 66.236.37.85 Port 1262
>   Defaddr->IP  : 0.0.0.0 Port 5060
>   Def. Username: nick
>   SIP Options  : (none)
>   Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
>   Codec Order  : (none)
>   Auto-Framing:  No
>   Status       : OK (192 ms)
>   Useragent    : Grandstream BT120 1.0.8.17
>   Reg. Contact : sip:nick at 192.168.2.102



-- 
/Nick
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