[Asterisk-Users] Qestion about TDM over enthernet
FCG ZHAO Zigang
Zigang.ZHAO at alcatel-sbell.com.cn
Thu Dec 23 02:01:45 MST 2004
who can tell me how to do TDM over enthernet ?
pc a connect pc b only use TDM card?
thank you
John.
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发送时间: 2004年12月23日 11:47
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主题: Asterisk-Users Digest, Vol 5, Issue 336
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Today's Topics:
1. Re: Still unable to use g729 codec... please HELP (Rodolfo Grave)
2. Re: Can't Receive/Send Calls (Norman Zhang)
3. RE: Zaptel/Zapata config from T410p to Brooktrout T1
(Jason Kawakami)
4. Re: Still unable to use g729 codec... please HELP
(Kristian Kielhofner)
5. Re: Still unable to use g729 codec... please HELP (Rodolfo Grave)
6. Re: Still unable to use g729 codec... please HELP
(Kristian Kielhofner)
7. Re: hint extension and Snom phones - CVS or stable? (Karl Brose)
8. WARNING Maximum retries exceeded on call for seqno 102 (Kevin)
9. Re: Re: 'I'nvalid extension handling problems, even with
workaround (telmo at n1.com)
10. Re: Still unable to use g729 codec... please HELP (Rodolfo Grave)
11. Re: Still unable to use g729 codec... please HELP
(Kristian Kielhofner)
12. RE: polycom and cdp (Richard)
13. Re: RE: Zaptel/Zapata config from T410p to BrooktroutT1 (jbebeau)
----------------------------------------------------------------------
Message: 1
Date: Thu, 23 Dec 2004 03:32:22 +0100
From: Rodolfo Grave <rodolfograve at yahoo.es>
Subject: Re: [Asterisk-Users] Still unable to use g729 codec... please
HELP
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CA2E36.9030802 at yahoo.es>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Yeap... :(
Kristian Kielhofner wrote:
> Rodolfo Grave wrote:
>
>> Hi.
>>
>> Has anyone accomplished to use the g729 codec? I have the license
>> installed, and I have tried with X-Pro and a Grandstream Budgetone
>> configured to use g729 only. This is what I get from Asterisk:
>>
>> Dec 23 02:38:07 WARNING[21176]: chan_sip.c:2764 process_sdp: No
>> compatible codecs!
>> Dec 23 02:38:09 NOTICE[21176]: chan_sip.c:7295 handle_request: Unable
>> to create/find channel
>>
>> My sip.conf contains:
>>
>> disallow=all
>> allow=g729
>>
>> for both devices!!
>>
>> I dont know what to do, I need to use the g729 codec. Please help.
>>
>> If I enable GSM in the device and add allow=gsm everything works, so
>> it is a codec problem.
>>
>> The license and the codec seems to be correctly installed:
>>
>> *CLI> show g729
>> 0/0 encoders/decoders of 1 licensed channels are currently in use
>> *CLI>
>>
>> Please, help me.
>>
>> RODOLFO
>
>
> Rodolfo,
>
> I assume that you tried the obvious stuff, re-reading the docs,
> running the register program, getting the proper g729 binary from
> Digium, etc?
>
> --
> Kristian Kielhofner
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 2
Date: Wed, 22 Dec 2004 18:40:03 -0800
From: Norman Zhang <norman.zhang at rd.arkonnetworks.com>
Subject: Re: [Asterisk-Users] Can't Receive/Send Calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CA3003.7090404 at rd.arkonnetworks.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
With this trimmed down versions of sip.conf and extensions.conf. I can
now receive calls from outside. But audio will not traverse out to the
internet. I can hear the caller no problem. Also I still cannot
dial-out. Any ideas?
Regards,
Norman Zhang
> I removed all the PSTN stuffs. As I'm only trying to make SIP work.
> Would someone kindly give me a few pointers?
>
> [general]
> disallow=all
> allow=ulaw
> port=5060
> bindaddr=0.0.0.0
> externip=207.34.136.26
> localnet=192.168.22.0/255.255.255.0
> context=inbound-sip
> maxexpirey=180
> defaultexpirey=160
> tos=reliability
> srvlookup=yes
> register => 533990:normanzhang at fwd.pulver.com/533990
>
> [fwd]
> type=friend
> secret=normanzhang
> username=533990
> fromuser=533990
> fromdomain=fwd.pulver.com
> host=fwd.pulver.com
> dtmfmode=inband
> nat=yes
> canreinvite=no
>
> [101]
> disallow=all
> allow=ulaw
> type=friend
> host=dynamic
> dtmfmode=inband
> username=101
> secret=testing123
> context=home
> nat=no
>
> ; extensions.conf
>
> [general]
> static=yes
> writeprotect=no
>
> [globals]
> MAINPHONE=SIP/101
> FWDUSERID=533990
> FWDUSERNAME=Norman Zhang
> FWDPREFIX=*
>
> ; Macros
>
> [macro-fastbusy]
> exten => s,1,Answer
> exten => s,2,Wait,1
> exten => s,3,Playback(ss-noservice)
> exten => s,4,Wait(30)
> exten => s,5,Hangup
>
> [macro-dialoutsip]
> exten => s,1,SetCallerID(${FWDUSERID})
> exten => s,2,SetCIDName(${FWDUSERNAME})
> exten => s,3,Dial(SIP/${FWDPREFIX}${ARG1}@fwd,70)
> exten => s,4,Macro(fastbusy)
> exten => s,5,Hangup
> exten => s,104,Macro(fastbusy)
> exten => s,105,Wait,3
> exten => s,106,Playtones(congestion)
> exten => s,107,Wait,30
> exten => s,108,Hangup
>
> ; Outbound
>
> [fwd-out]
> exten => _8.,1,SetCallerID(${FWDUSERID})
> exten => _8.,2,SetCIDName(${FWDUSERNAME})
> exten => _8.,3,Dial(SIP/${EXTEN:1}@fwd,70)
> exten => _8.,4,Macro(fastbusy)
> exten => _8.,5,Hangup
>
> [long-distance]
> exten => _1XXXXXXXXXX,1,Macro(dialoutsip,${EXTEN})
> exten => _1XXXXXXXXXX,2,Macro(fastbusy)
>
> ; SIP lines
>
> [inbound-sip]
> exten => 533990,1,Goto(local,101,1)
>
> ; Internal Extensions
>
> [local]
> exten => 101,1,Dial(${MAINPHONE},20,Tt)
> exten => 101,2,Voicemail(u101)
> exten => 101,3,Hangup
> exten => 101,102,Voicemail(b101)
> exten => 101,103,Hangup
>
> [home]
> include => fwd-out
> include => local
> include => long-distance
------------------------------
Message: 3
Date: Wed, 22 Dec 2004 19:43:35 -0700
From: "Jason Kawakami" <jkkawakami at optellabs.com>
Subject: [Asterisk-Users] RE: Zaptel/Zapata config from T410p to
Brooktrout T1
To: <asterisk-users at lists.digium.com>
Message-ID: <200412230239.iBN2dQQZ054184 at dkh.dsl.xmission.com>
Content-Type: text/plain; charset="us-ascii"
-----Original Message-----
Message: 8
zaptel.conf
span=2,0,0,esf,b8zs
e&m=25-48
zapata.conf
signaling = em_w
context = faxserver
group = 3
channel = 25-28
exten.conf
exten => 1231231234,1,Dial(Zap/2-2/${exten})
I think if you change the last line to :
exten => 1231231234,1,Dial(Zap/G3/${EXTEN})
it should work. I have had difficulty in the past doing direct selection of
individual channels on t-1 spans.
------------------------------
Message: 4
Date: Wed, 22 Dec 2004 20:49:16 -0600
From: Kristian Kielhofner <kris at krisk.org>
Subject: Re: [Asterisk-Users] Still unable to use g729 codec... please
HELP
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CA322C.7020506 at krisk.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Rodolfo Grave wrote:
> Yeap... :(
>
Has (or are they willing) Digium support looked at it? If they haven't,
I would be willing to SSH into the machine if it is available.
Let me know, either here or directly to me.
--
Kristian Kielhofner
------------------------------
Message: 5
Date: Thu, 23 Dec 2004 03:56:51 +0100
From: Rodolfo Grave <rodolfograve at yahoo.es>
Subject: Re: [Asterisk-Users] Still unable to use g729 codec... please
HELP
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CA33F3.7080705 at yahoo.es>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I've seen other threads about the topic here (just that with a
Sipura)... this is the thread subject:
[Asterisk-Users] G729 and Sipura.
Digium's answer to this person was to blame the device, so I haven't
even try to contact Digium support. I'll do it now, and I'll let you
know. However, I did contact Grandstream support and they said they had
never have that kind of problem, and they asked me an Ethereal view of
network traffic.
Thanks a lot for your support.
RODOLFO
Kristian Kielhofner wrote:
> Rodolfo Grave wrote:
>
>> Yeap... :(
>>
>
> Has (or are they willing) Digium support looked at it? If they haven't,
> I would be willing to SSH into the machine if it is available.
>
> Let me know, either here or directly to me.
>
> --
> Kristian Kielhofner
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 6
Date: Wed, 22 Dec 2004 21:02:04 -0600
From: Kristian Kielhofner <kris at krisk.org>
Subject: Re: [Asterisk-Users] Still unable to use g729 codec... please
HELP
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CA352C.7070101 at krisk.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Rodolfo Grave wrote:
> I've seen other threads about the topic here (just that with a
> Sipura)... this is the thread subject:
>
> [Asterisk-Users] G729 and Sipura.
>
> Digium's answer to this person was to blame the device, so I haven't
> even try to contact Digium support. I'll do it now, and I'll let you
> know. However, I did contact Grandstream support and they said they had
> never have that kind of problem, and they asked me an Ethereal view of
> network traffic.
>
> Thanks a lot for your support.
>
> RODOLFO
>
It truely is strange because I have G729 running on at least 4 asterisk
systems used with Cisco 7960's, Polycom IP 300's, IP 600's, Sipura
2000's and 3000's. I have never had problems.
--
Kristian Kielhofner
------------------------------
Message: 7
Date: Wed, 22 Dec 2004 22:20:42 -0500
From: Karl Brose <khb at brose.com>
Subject: Re: [Asterisk-Users] hint extension and Snom phones - CVS or
stable?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CA398A.7000402 at brose.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
There is no such thing as "subscribecontext" parameter in SIP.
I have updated the wiki with the correct current information to make
this work.
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom
--khb
Peer Oliver Schmidt wrote:
> Hi,
>
> does the hint extension work together with the Snom phones in stable?
> I don't get an error in the dialplan, but it does not work either.
>
> On SIP/26 I want to monitor SIP/22. This is what I do right now:
>
> extension.conf
> [incoming]
> exten => 955,hint,SIP/22
> exten => 955,1,Dial(SIP/22)
>
> sip.conf
> [26]
> ...
> subscribecontext=incoming
> ...
>
> Running Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b
>
> TIA and rgds
> pos
>
>
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 8
Date: Wed, 22 Dec 2004 22:21:10 -0500
From: Kevin <Asterisk at gtcus.com>
Subject: [Asterisk-Users] WARNING Maximum retries exceeded on call for
seqno 102
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<asterisk-users at lists.digium.com>
Message-ID: <019901c4e89e$72fc4a30$2c02a8c0 at gtcp4>
Content-Type: text/plain; charset=US-ASCII
I am running the Stable 1.01 version of Asterisk on a Dell SC420, RedHat
9. I have a PRI to the Telco and the phones are SIP 7960's. Both the
phones and the asterisk box are on a private subnet behind the firewall.
No NAT involved here.
Intermittently, the users complain of the conversation 'fading out' and
sometimes getting disconnected. I have noticed the following errors in
the log which may correspond to the call drop off's.
Can anyone offer any suggestions as what may be causing this problem and
how to diagnose the source of the problem?
Thanks,
Kevin
Log example
Dec 22 13:27:37 WARNING[1087241008]: Maximum retries exceeded on call
77a100794f
4a5f8457d17224514e6315 at 10.0.0.6 for seqno 102 (Non-critical Request)
------------------------------
Message: 9
Date: Wed, 22 Dec 2004 19:27:43 -0800
From: <telmo at n1.com>
Subject: Re: [Asterisk-Users] Re: 'I'nvalid extension handling
problems, even with workaround
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>, Rich Adamson <radamson at routers.com>
Message-ID: <20041223032602.837FB2FF1F5 at lists.digium.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello Rich,
First of all, thank you very much for your help and patience.
I've just arrived home from work (yes, I'm one of the midnight oil burners :-))
and implemented and tested your suggestions; unfortunatelly it didn't work, the
same behaviour remains.
More details follow below, in-line:
On Wed Dec 22 4:14 , Rich Adamson <radamson at routers.com> sent:
>Inline...
>
>> Humrmrm... 2 days, no answers... :-/
>
>Well, let me see if I can take a stab at this one.
You sure did. Your help was most appreciated, I learned a lot from thinking about
your suggestions, even if they did not work as planned (see below) they made a
lot of sense.
>After working with * for about a year now, I'd suggest the toughest
>part of the learning curve is truly understanding how to take advantage
>of the various 'context' statements to accomplish an objective. Part of
>reason for the steep learning curve appears to relate to the lack of
>any reasonable form of tracing/debugging what the system is actually
>using for a context at each step. (What I mean is that its not intutive
>for the beginner.)
I agree. I'm working with 15 (!) "-v" on the command-line here and yet I can only
perceive Asterisk has left or entered a context indirectly, by the commands that
are executing.
>It would really be nice if there was a 'debug context' type command
>that would simply display each extensions.conf line as it is executed.
That can't be too hard to implement, when I'm a little more familiarized with
Asterisk I will certainly try to code that in.
>> Either I made a stupid question (I don't think so: I have *really* tried to solve
>> that on my own before asking the list) or this one's just something nobody has
>> ever tried but me (I also find that unlikely: even the telco here plays a message
>> when I dial a wrong number; also there's the wiki page I mentioned, which
>> indicates that someone in the past has had the same issue).
>>
>> >I'm having trouble configuring Asterisk to play an "invalid extension" message to
>> >anyone dialing an undefined extension.
>>
>> >I then did the "separate context with _." trick the above wiki page suggests; at
>> >first it seemed to work: picking up an extension and dialing any invalid
>> >extension would play the message (albeit it would play twice, can't understand
>> >why) and then hang up.
>>
>> >;;;extensions.conf
>> >[internal] ;;; context used by our internal SIP-phon
>> >include => voiptalk.org ;include context below
>> >exten => 11,1,Dial(SIP/gsbt100,20,tr);calling : dial our office phone
>> >include => invalid_calls ;all ext numbers not handled above are invalid
>
>The 'separate context' approach _does_ work, but you've just confused
>that approach by dropping the Dial statement in the middle. Change this
>to something like:
> [internal]
> include => valid-extensions
> include => voiptalk
> include => any-other-context-that-you-need
> include => invalid-calls
Makes a lot of sense, and also leads to much more intelligible structure. I
implemented it almost literally as you suggest (see below for my extensions.conf
file after the modifications).
>The above _sequence_ of include statements is maintained for each call.
>In other words, if a call does _not_ match entries in 'valid-extensions',
>then it proceeds to the next include. However, if a match is found (including
>special cases such as 't', etc) then the call processing may _not_ step
>through the remaining includes.
OK. I understand what you are stating, it makes a lot of sense. Unfortunatelly it
seems Asterisk disagrees with us... :-) please see below.
>I'm not sure at all about using context names with a period in it, so
>just to ensure that isn't causing an issue, stick to context names with
>alphanumeric characters. (At least eliminate that uncertainty.)
OK. Let's play it _really_ safe: I've removed "." from context names, and
replaced "_" (underline) for "-" (dashes), so right now I'm only using characters
from the set [a-z0-9-] in context names.
>> >[voiptalk.org]
>> >;forwards any calls starting with an "8" thru voiptalk.org
>> >exten => _8.,1,Answer
>> >exten => _8.,3,SetCIDNum(55555555)
>> >exten => _8.,4,SetCIDName(My Name And Surname)
>> >exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)
>> >exten => _8.,6,HangUp
>> >[invalid_calls] ;;; default context for invalid calls
>> >exten => _.,1,Wait(1)
>> >exten => _.,2,Answer
>> >exten => _.,3,Playback(invalid)
>> >exten => _.,4,Hangup
>> >;;;end of extensions.conf
>
>In the above [invalid_calls] context, change the order to:
> exten => _.,1,Answer
> exten => _.,2,Wait(1)
> exten => _.,3,Playback(invalid)
> exten => _.,4,Hangup
Also makes a lot of sense. Done it that way.
>It is rather important from a learning perspective that you create
>the [internal] context for use by "internal phones only". Don't try
>to use that same context for incoming voiptalk calls, etc. For incoming
>external calls, create a separate context something like:
>[incoming-voiptalk]
> include => valid-extensions
> include => invalid-calls
Ah! Thanks for the tip. Also makes a lot of sense.
>After you've made the above changes, "stop" and restart asterisk to
>ensure you know exactly what asterisk believes the dialplan is supposed
>to be. Stay away from the various 'reload' commands until you understand
>exactly what is happening.
Did that: "stop now" in the cli, and then restarted asterisk with "asterisk -cp
-vvvvvvvvvvvvvv" after modifying the files.
>Then place test calls and watch the CLI paying close attention to which
>contexts are actually in use (or which context a call stops progressing
>in). If necessary, increase the debug level to see the needed call progress
>info.
Did it; unfortunatelly the same behaviour continues: when I call (for example)
8902, I hear Voiptalk's welcome message and right after it I hear the "invalid
extension" message...
The files ended up as below:
;;;extensions.conf
[internal]
include => valid-extensions ;; internal sip phones
include => voiptalk ;; dial out thru voiptalk.org
include => tests ;; miscellaneous tests
include => invalid-calls ;all ext numbers not handled above are invalid
[valid-extensions] ;;; context used by our internal SIP-phon
exten => 11,1,Dial(SIP/gsbt100,20,tr) ;calling <11>: dial our office phone
[voiptalk]
;forwards any calls starting with an "8" thru voiptalk.org
exten => _8.,1,Answer
exten => _8.,3,SetCIDNum(55555555)
exten => _8.,4,SetCIDName(My Full Name)
exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk,999,g)
exten => _8.,6,HangUp
[invalid-calls] ;;; default context for invalid calls
exten => _.,1,Answer
exten => _.,2,Wait(1)
exten => _.,3,Playback(invalid)
exten => _.,4,Hangup
;;;eof extensions.conf
;;;sip.conf
[general]
register => 55555555:666666 at voiptalk.org/1000
maxexpirey = 180
defaultexpirey = 160
[voiptalk]
type=friend
secret=666666
username=55555555
host=voiptalk.org
fromdomain=voiptalk.org
insecure=very ;needed to allow incoming calls to bypass authentication
dtmfmode=info ;Choices are inband, rfc2833, or info
[gsbt100]
type=friend
host=dynamic ;must use dynamic even if fixed ip/hostname, or else
defaultip=192.168.1.200 ;asterisk won't allow it to register
canreinvite=no ;avoid trouble with NAT: keeps Asterisk "in the middle"
username=gsbt100 ;gsbt100 sends that when the respective fields are empty
secret=mysecret
dtmfmode=rfc2833 ;Choices are inband, rfc2833, or info
mailbox=1234 ;Mailbox for message waiting indicator
context=internal ;Context for internal SIP phones
callerid="Office" <11> ;format is "name" <number>
;;;eof sip.conf
Here comes the output on the console when I dial '8902':
Setting NAT on RTP to 0
Stopping retransmission on '5e5172baa8c28adf at 192.168.1.200' of Response 45199: F
ound
Setting NAT on RTP to 0
Check for res for gsbt100
Call from user 'gsbt100' is 1 out of 0
build_route: Contact hop: <sip:gsbt100 at 192.168.1.200;user=phone>
-- Executing Answer("SIP/gsbt100-7dcf", "") in new stack
-- Executing Wait("SIP/gsbt100-7dcf", "1") in new stack
Stopping retransmission on '5e5172baa8c28adf at 192.168.1.200' of Response 45200: F
ound
-- Executing SetCIDNum("SIP/gsbt100-7dcf", "55555555") in new stack
-- Executing SetCIDName("SIP/gsbt100-7dcf", "My Full Name") in new stack
-- Executing Dial("SIP/gsbt100-7dcf", "SIP/902 at voiptalk|999|g") in new stack
SIMPLE DIAL (NO URL)
Setting NAT on RTP to 0
Outgoing Call for 902
902 is not a local user
-- Called 902 at voiptalk
(Provisional) Stopping retransmission (but retaining packet) on
'12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' Request 102: Found
Acked pending invite 102
Stopping retransmission on '12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' of
Request 102: Found
Stopping retransmission on '12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' of
Request 102: Not Found
(Provisional) Stopping retransmission (but retaining packet) on
'12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' Request 103: Found
Acked pending invite 103
Stopping retransmission on '12e2f2cc30702a0c3647cbeb4d2d2193 at voiptalk.org' of
Request 103: Found
build_route: Record-Route hop: <sip:902 at 82.145.32.73;ftag=as3688ce90;lr=on>
build_route: Contact hop: <sip:902 at 217.14.132.184>
-- SIP/voiptalk-bbd4 answered SIP/gsbt100-7dcf
-- Attempting native bridge of SIP/gsbt100-7dcf and SIP/voiptalk-bbd4
Oooh, format changed to 8
Ooh, format changed from UNKN to ULAW
Ooh, format changed from UNKN to ALAW
Registration from '<sip:gsht286 at 192.168.1.1;user=phone>' failed for '192.168.1.201'
Auto destroying call 'd761b1512a710969 at 192.168.1.201'
Scheduled a timeout # 55
Stopping retransmission on '327b23c6643c98696633487374b0dc51 at 192.168.1.1' of
Request 107: Found
Stopping retransmission on '327b23c6643c98696633487374b0dc51 at 192.168.1.1' of
Request 108: Found
Registration successful
Cancelling timeout 55
Auto destroying call '8a7ae46c9e0d1370 at 192.168.1.201'
Didn't get a frame from channel: SIP/voiptalk-bbd4
Bridge stops bridging channels SIP/gsbt100-7dcf and SIP/voiptalk-bbd4
update_user_counter(902) - decrement outUse counter
902 is not a local user
Exiting with DIALSTATUS=ANSWER.
-- Executing Hangup("SIP/gsbt100-7dcf", "") in new stack
== Spawn extension (internal, 8902, 6) exited non-zero on 'SIP/gsbt100-7dcf'
-- Executing Answer("SIP/gsbt100-7dcf", "") in new stack
-- Executing Wait("SIP/gsbt100-7dcf", "1") in new stack
-- Executing Playback("SIP/gsbt100-7dcf", "invalid") in new stack
Difference is 50272, ms is 6304
-- Playing 'invalid' (language 'en')
-- Executing Hangup("SIP/gsbt100-7dcf", "") in new stack
== Spawn extension (internal, h, 4) exited non-zero on 'SIP/gsbt100-7dcf'
update_user_counter(gsbt100) - decrement inUse counter
Stopping retransmission on '5e5172baa8c28adf at 192.168.1.200' of Request 102: Found
As I said above, what I hear on the gsbt100 phone when I pick it up and dial
'8902' remains unchanged: first I hear VoipTalk's message, and it's followed by
my "invalid" message. Examining the console output above seems to indicate that
Asterisk is simply "falling thru" the list of contexts included in the
"[internal]" context, contrarywise to what we were expecting...
As I said, I'm using Asterisk 1.0.2; Can this be a bug in that release? If so,
can you indicate me a release where you think this works?
Thanks again for your help.
Best Regards,
Telmo.
----
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------------------------------
Message: 10
Date: Thu, 23 Dec 2004 04:28:57 +0100
From: Rodolfo Grave <rodolfograve at yahoo.es>
Subject: Re: [Asterisk-Users] Still unable to use g729 codec... please
HELP
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CA3B79.3070502 at yahoo.es>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Can you please try to set up X-Pro or a Budgetone with g729 (In case you
have one of those at hand, of course)? If you're succesful I'll know
it's Asterisk having problems. I dont have much resources to test (just
the Budgetone and X-Pro).
Thanks again,
RODOLFO
Kristian Kielhofner wrote:
> Rodolfo Grave wrote:
>
>> I've seen other threads about the topic here (just that with a
>> Sipura)... this is the thread subject:
>>
>> [Asterisk-Users] G729 and Sipura.
>>
>> Digium's answer to this person was to blame the device, so I haven't
>> even try to contact Digium support. I'll do it now, and I'll let you
>> know. However, I did contact Grandstream support and they said they
>> had never have that kind of problem, and they asked me an Ethereal
>> view of network traffic.
>>
>> Thanks a lot for your support.
>>
>> RODOLFO
>>
>
> It truely is strange because I have G729 running on at least 4 asterisk
> systems used with Cisco 7960's, Polycom IP 300's, IP 600's, Sipura
> 2000's and 3000's. I have never had problems.
>
> --
> Kristian Kielhofner
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 11
Date: Wed, 22 Dec 2004 21:31:14 -0600
From: Kristian Kielhofner <kris at krisk.org>
Subject: Re: [Asterisk-Users] Still unable to use g729 codec... please
HELP
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <41CA3C02.1070206 at krisk.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Rodolfo Grave wrote:
> Can you please try to set up X-Pro or a Budgetone with g729 (In case you
> have one of those at hand, of course)? If you're succesful I'll know
> it's Asterisk having problems. I dont have much resources to test (just
> the Budgetone and X-Pro).
>
> Thanks again,
>
> RODOLFO
Rodolfo,
I would love to, but I don't have either...
--
Kristian Kielhofner
------------------------------
Message: 12
Date: Wed, 22 Dec 2004 17:41:42 -1000
From: "Richard" <richard at o-matrix.org>
Subject: RE: [Asterisk-Users] polycom and cdp
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <EINSTEINJ2TUSzbAUfK00001110 at einstein.systemmetrics.com>
Content-Type: text/plain; charset="us-ascii"
> Richard wrote:
> > Hi,
> >
> > Has anyone tried to use cdp to push the voice vlan tag to polycom
phones?
> > The document says that it is supported, but I can't make it work.
> >
> > Thanks,
> > Richard
>
> Richard,
>
> I can't either. I've tried using HP Procurve switches and even my
> Catalyst 2950T-24. Neither work. I have been setting vlan's manually.
>
> Note: My 7960 works with the Catalyst, and I haven't tested it with the
> Procurve.
>
Hi Kristian,
I have a catalyst 3500xl and can't get it work.
On the switch, I use
int f0/xx
switch access vlan 100
switch voice vlan 200
Supposedly it uses tagged vlan 200 for voice and untagged traffic for data.
I sniff the traffic and saw CDP packets. But 7960 can't get the right vlan
configured. Any suggestion?
Thanks,
Richard
------------------------------
Message: 13
Date: Wed, 22 Dec 2004 22:46:43 -0500
From: "jbebeau" <jbebeau at 1nettw.net>
Subject: Re: [Asterisk-Users] RE: Zaptel/Zapata config from T410p to
BrooktroutT1
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <003c01c4e8a2$0796dd10$6901a8c0 at SERF>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
I had tried g3 first...the individual address was at guess that g3 didn't
work.
Thanks.
----- Original Message -----
From: "Jason Kawakami" <jkkawakami at optellabs.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, December 22, 2004 9:43 PM
Subject: [Asterisk-Users] RE: Zaptel/Zapata config from T410p to
BrooktroutT1
>
>
> -----Original Message-----
>
> Message: 8
> zaptel.conf
>
> span=2,0,0,esf,b8zs
> e&m=25-48
>
> zapata.conf
>
> signaling = em_w
> context = faxserver
> group = 3
> channel = 25-28
>
> exten.conf
>
> exten => 1231231234,1,Dial(Zap/2-2/${exten})
>
> I think if you change the last line to :
>
> exten => 1231231234,1,Dial(Zap/G3/${EXTEN})
>
> it should work. I have had difficulty in the past doing direct selection
> of
> individual channels on t-1 spans.
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
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