Gervais de Montbrun
2005-Nov-11 09:09 UTC
[Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 1
Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> on November 11, 2005 at 10:10 AM -0400 wrote:>set the keepalive to 60 or moreOK. I set this to 120> >that phone should not be able to display a hint status so >speeddial = 500,500Thanks. I've made the change>The log could be more verbose than this. >Set debug = 10 in your sccp.conf >or in the console >sccp debug 10 > >You should see what is happening with your audio streamI did this in the console and the output is below. It does not seem to say much to me about audio. Cheers, Gervais --------------------------------------------------------------------------------------------------------------------------- Asterisk Ready. *CLI> sccp debug 10 -- SEP003080629796: Old session marked down -- SEP003080629796: Killing Session 192.168.1.440|20|tr") in new stack -- SCCP: Looking for line 140eate a channel type=SCCP, format=256, data=140, options -- SCCP: Asterisk asked for the state (5) of the line 140 -- SEP003080629796: found line 140 -- SEP003080629796: New channel number: 1 on line 140 -- SEP003080629796: Global Capabilities: 268 -- SEP003080629796: format request: 4/4 -- SEP003080629796: Channel SCCP/140-00000001, capabilities: DEVICE 0x4 (ulaw) NATIVE 0x4 (ulaw) BEST 4 (ulaw) -- SEP003080629796: Allocated asterisk channel 140-1 -- SEP003080629796: Asterisk request to call SCCP/140-00000001 -- SEP003080629796: Set callingParty Name TLS Group on channel 1 -- SEP003080629796: Set callingParty Number 500 on channel 1 -- SEP003080629796: Set calledParty Name "TLS Group" on channel 1 -- SEP003080629796: Set calledParty Number 140 on channel 1 -- SEP003080629796: getting the active channel on device -- SEP003080629796: Indicate SCCP state (Ringing) on call 140-1 -- SEP003080629796: Send and Set the call state Ringing(4) for 140-1 -- SEP003080629796: Send callinfo for Inbound channel 1 -- SEP003080629796: Send lamp mode LampBlink(5) on line 1 -- SEP003080629796: Send ringer mode Outside(3) on device -- SEP003080629796: Set asterisk state Ringing (5) for call 1 -- SEP003080629796: Finish to indicate state SCCP (Ringing), SKINNY (Ringing) on call 140-1 -- Called 140 -- SCCP: Looking for line 140 -- SEP003080629796: found line 140 -- SEP003080629796: Looking for a channel with state "Ringing" (4) on device -- SEP003080629796: Looking for a channel with state "Ringing" (4) on line 140 -- SEP003080629796: Found channel (1) with state "Ringing" (4) on line 140 -- SEP003080629796: Found channel (1) with state "Ringing" (4) on device -- SCCP: Asterisk asked for the state (6) of the line 140 -- SCCP/140-00000001 is ringing -- SEP003080629796: >> Got message OffHookMessage -- SEP003080629796: getting the active channel on device -- SEP003080629796: Taken Offhook -- SEP003080629796: Looking for a channel with state "Ringing" (4) on device -- SEP003080629796: Looking for a channel with state "Ringing" (4) on line 140 -- SEP003080629796: Found channel (1) with state "Ringing" (4) on line 140 -- SEP003080629796: Found channel (1) with state "Ringing" (4) on device -- SEP003080629796: getting the active channel on device -- SEP003080629796: Answer the channel 140-1 -- SEP003080629796: Set the active channel 1 on device -- SEP003080629796: Send the active line 140 -- SEP003080629796: Indicate SCCP state (Connected) on call 140-1 -- SEP003080629796: Send ringer mode RingOff(1) on device -- SEP003080629796: Send speaker mode 1 -- SEP003080629796: Stop tone on device -- SEP003080629796: Send lamp mode LampOn(2) on line 1 -- SEP003080629796: Send and Set the call state Connected(5) for 140-1 -- SEP003080629796: Send callinfo for Inbound channel 1 -- SEP003080629796: readformat 4, payload 4 -- SEP003080629796: Ask the device to open a RTP port on channel 1. Codec: G.711 u-law 64k, echocancel: ON -- SEP003080629796: Starting RTP on channel 140-1 -- SEP003080629796: Creating rtp server connection at 192.168.1.125 -- SEP003080629796: Set asterisk state Up (6) for call 1 -- SEP003080629796: Finish to indicate state SCCP (Connected), SKINNY (Connected) on call 140-1 -- SCCP: Looking for line 140 -- SEP003080629796: found line 140 -- SEP003080629796: Looking for a channel with state "Ringing" (4) on device -- SEP003080629796: Looking for a channel with state "Ringing" (4) on line 140 -- SCCP: Asterisk asked for the state (2) of the line 140 -- SCCP/140-00000001 answered SIP/500-59ab -- SCCP: Asterisk request to hangup Inbound channel SCCP/140-00000001 -- SEP003080629796: Close openreceivechannel on channel 1 -- SEP003080629796: Stopping RTP on channel 140-1 -- SEP003080629796: Stop media transmission on channel 1 -- SEP003080629796: Requesting CallStatisticsAndClear from Phone -- SEP003080629796: Current channel 140-1 state Connected(5) -- SEP003080629796: getting the active channel on device -- SEP003080629796: Sending tone Zip (50) -- SEP003080629796: Indicate SCCP state (OnHook) on call 140-1 -- SEP003080629796: Send speaker mode 2 -- SEP003080629796: Send and Set the call state OnHook(2) for 140-1 -- SEP003080629796: Send lamp mode LampOff(1) on line 1 -- SEP003080629796: Stop tone on device -- SEP003080629796: Send date/time -- SEP003080629796: Finish to indicate state SCCP (OnHook), SKINNY (OnHook) on call 140-1 -- SEP003080629796: Deleting channel 1 from line 140 -- SEP003080629796: Deleted channel 1 from line 140 == Spawn extension (default, 140, 1) exited non-zero on 'SIP/500-59ab' -- SCCP: Looking for line 140 -- SEP003080629796: found line 140 -- SCCP: Asterisk asked for the state (1) of the line 140 -- SEP003080629796: >> Got message OnHookMessage -- SEP003080629796 is Onhook -- SEP003080629796: getting the active channel on device -- SEP003080629796: Send speaker mode 2 -- SEP003080629796: Stop tone on device -------------- next part -------------- An HTML attachment was scrubbed... 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People,
I'm tring to use 2 e-1 in Brazil. In order to get R@ signaling, I compilled
libdsp, unicall and stuff following www.soft-switch.org
<http://www.soft-switch.org/> comparing with another site (Dezert of
Zazamora, in Mexico). Asterisk is running fine with Asterisk but I can't
make calls.
The guys on the telco company tells me that I have a LOMF (Loss of Multi
Frame) error in their end (the far-end) and we can exchange digits. They
expect R2-Digital signaling and they think the implementation I use is not
quite right.
When I try to make a call the result is as follows:
Nov 11 12:02:13 VERBOSE[3451]: -- Executing Dial("SIP/200-3ced",
"UNICALL/g2/55431100") in new stack Nov 11 12:02:13 DEBUG[3451]: Using
channel 1 Nov 11 12:02:13 DEBUG[3451]: unicall_call called -
'g2/55431100'
Nov 11 12:02:13 DEBUG[3451]: unicall_call caller id - '"Mabio
Coelho" <200>'
Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Call control(1) Nov 11
12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Make call Nov 11 12:02:13
WARNING[3451]: Make call failed - Blocked Nov 11 12:02:13 DEBUG[3451]: ast
call on peer returned -1 Nov 11 12:02:13 DEBUG[3451]: Hanging up channel
'UniCall/1-1'
Nov 11 12:02:13 DEBUG[3451]: unicall_hangup(UniCall/1-1) Nov 11 12:02:13
WARNING[3451]: MFC/R2 UniCall/1 Channel gains Nov 11 12:02:13 WARNING[3451]:
MFC/R2 UniCall/1 Channel switching Nov 11 12:02:13 DEBUG[3451]: Hangup:
channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Nov 11
12:02:13 DEBUG[3451]: Updated conferencing on 1, with 0 conference users Nov
11 12:02:13 VERBOSE[3451]: -- Hungup 'UniCall/1-1'
Nov 11 12:02:13 VERBOSE[3451]: == Everyone is busy/congested at this time
I'm attaching the three most relevant configuration files. Sorry if is kind
of messy (a lot of lines commented out), that is because I tried a lot of
things before posting.
Best regards,
Mabio Coelho
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Sergio Chersovani
2005-Nov-11 11:01 UTC
[Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 1
Gervais de Montbrun ha scritto:> **I did this in the console and the output is below. It does not seem > to say much to me about audio.Dunno why, but the phone is not sending an open receive channel ack. In fact it does ot open the rtp media port so the channel don't know where to send (udp port) the rtp packets What firmware are you running? Sergio
Gervais de Montbrun
2005-Nov-12 21:25 UTC
[Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 1
Matt Riddell on November 12, 2005 at 9:53 PM -0400 wrote:>PLEASE DO NOT POST IN HTML! :)Sorry Matt, this is controlled server side for me. The server should be sending in html and plain text and displaying what your email client should be able to read... Isn't this what is happening? Any ideas with my issue? I am currently at the point where I switched to the SCCP protocol for my Cisco 12 SP+ as suggested by Sergio. Things seem to work, but I can not call into my Cisco phone. It rings, but then there is no audio and the phone resets after a short while. Cheers, Gervais -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051112/491f61ce/attachment.htm