[asterisk-dev] Trouble with originating a call through Asterisk Manager Interface

Geoffrey Mina geoffreymina at gmail.com
Sun Jul 12 16:37:27 CDT 2009


I would say you are better off with the asterisk-users list or perhaps
forums.digium.com to get someone to help you with this.  You will
probably need to post a SIP debug to get anyone to assist further.

On Sun, Jul 12, 2009 at 2:52 PM, eric weaver<ecweaver at gmail.com> wrote:
>
> I am doing a little application to originate a call through Asterisk via AMI
> (Perl Asterisk::Manager).
> It logs in successfully, does an originate command with
> Exten: 0020 (which is set up to answer and wait for 60 then hang up)
> Channel: SIP/5101234567 at test-host  (which comes to my desktop machine also
> running Asterisk).
>
> At the target machine I see only a CANCEL to which it immediately responds
> with a No Transaction.
>
> It looks like AstMan is asking for a Slin connection and the channel is set
> up only for Ulaw.  Don't know if that's a red herring.
> Any advice welcome.
>
> The debug log:
> [Jul 12 11:31:16] DEBUG[11317] manager.c: Manager received command
> 'Challenge'
> [Jul 12 11:31:16] DEBUG[11317] manager.c: Manager received command 'Login'
> [Jul 12 11:31:16] DEBUG[11317] config.c: Parsing /etc/asterisk/manager.conf
> [Jul 12 11:31:16] DEBUG[11317] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to
> acl for peer
> [Jul 12 11:31:16] DEBUG[11317] acl.c:
> 127.0.0.1/255.255.255.255/255.255.255.255 appended to acl for peer
> [Jul 12 11:31:16] DEBUG[11317] acl.c: ##### Testing 127.0.0.1 with 0.0.0.0
> [Jul 12 11:31:16] DEBUG[11317] acl.c: ##### Testing 127.0.0.1 with 127.0.0.1
> [Jul 12 11:31:16] DEBUG[11317] manager.c: Manager received command
> 'Originate'
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Asked to create a SIP channel
> with formats: 0x40 (slin)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Allocating new SIP dialog for (No
> Call-ID) - INVITE (With RTP)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Setting NAT on RTP to Off
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: *** Our native formats are 0x4
> (ulaw)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: *** Joint capabilities are 0x0
> (nothing)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: *** Our capabilities are 0x44
> (ulaw|slin)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: *** AST_CODEC_CHOOSE formats are
> 0x4 (ulaw)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: *** Our preferred formats from
> the incoming channel are 0x40 (slin)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: This channel will not be able to
> handle video.
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Outgoing Call for 5101234567
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Updating call counter for
> outgoing call
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Our T38 capability (0), joint T38
> capability (0)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: ** Our capability: 0x44
> (ulaw|slin) Video flag: False
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: ** Our prefcodec: 0x40 (slin)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: -- Done with adding codecs to SDP
> [Jul 12 11:31:16] DEBUG[11317] channel.c: Internal timing is disabled
> (option_internal_timing=0 chan->timingfd=-1)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Done building SDP. Settling with
> this capability: 0x44 (ulaw|slin)
> [Jul 12 11:31:16] DEBUG[11317] channel.c: Hanging up channel
> 'SIP/sip-flat5th-081d0500'
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Hangup call
> SIP/sip-flat5th-081d0500, SIP callid
> 30d8df253ccd7e0a5725087117c688ec at 192.168.253.35)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Hanging up channel in state Down
> (not UP)
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Acked pending invite 102
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Stopping retransmission on
> '30d8df253ccd7e0a5725087117c688ec at 192.168.253.35' of Request 102: Match
> Found
> [Jul 12 11:31:16] DEBUG[11317] chan_sip.c: Updating call counter for
> outgoing call
> [Jul 12 11:31:16] DEBUG[11317] devicestate.c: Notification of state change
> to be queued on device/channel SIP/sip-flat5th
> [Jul 12 11:31:16] DEBUG[11317] devicestate.c: Notification of state change
> to be queued on device/channel
> [Jul 12 11:31:16] DEBUG[11290] devicestate.c: No provider found, checking
> channel drivers for SIP - sip-flat5th
> [Jul 12 11:31:16] DEBUG[11290] chan_sip.c: Checking device state for peer
> sip-flat5th
> [Jul 12 11:31:16] DEBUG[11290] devicestate.c: Changing state for
> SIP/sip-flat5th - state 1 (Not in use)
> [Jul 12 11:31:16] DEBUG[11290] devicestate.c: Checking if I can find
> provider for "" - number: (null)
> [Jul 12 11:31:16] DEBUG[11290] devicestate.c: Checking provider Park with
> [Jul 12 11:31:16] DEBUG[11290] devicestate.c: Changing state for  - state 4
> (Invalid)
> [Jul 12 11:31:16] DEBUG[11314] app_queue.c: Device 'SIP/sip-flat5th' changed
> to state '1' (Not in use) but we don't care because they're not a member of
> any queue.
> [Jul 12 11:31:16] DEBUG[11313] chan_sip.c: = Found Their Call ID:
> 30d8df253ccd7e0a5725087117c688ec at 192.168.253.35 Their Tag  Our tag:
> as55af6a38
> [Jul 12 11:31:16] DEBUG[11313] chan_sip.c: Stopping retransmission on
> '30d8df253ccd7e0a5725087117c688ec at 192.168.253.35' of Request 102: Match
> Found
> [Jul 12 11:31:48] DEBUG[11313] chan_sip.c: Auto destroying SIP dialog
> '30d8df253ccd7e0a5725087117c688ec at 192.168.253.35'
> [Jul 12 11:31:48] DEBUG[11313] chan_sip.c: Destroying SIP dialog
> 30d8df253ccd7e0a5725087117c688ec at 192.168.253.35
> --
>
>
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