[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 154

Thu May 19 11:14:38 MST 2005

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Today's Topics:

   1. Re: Grandstream ATA 286 and ilbc (Kevin McCauley)
   2. Re: MusicOnHold probelms (admin)
   3. RE: SIP Phone Recommendations? (Ariel Batista)
   4. Re: Deleting Monitor Files After 2 Months (Steve Totaro)
   5. Two TDM04 with Poweredge (Tom Hayden)
   6. Re: Public vs. Private Network (Eric Wieling aka ManxPower)
   7. Re: asterisk-oh323 build problems (VoIP Newbie)
   8. Re: Outbound dialing issue with FXO (Johnathan Corgan)
   9. RE: Two TDM04 with Poweredge (David Brodbeck)
  10. RE: Re: Grandstream ATA 286 and ilbc (Anton Krall)
  11. (no subject) (M O)
  12. Re: Always Ringing (VoIP Newbie)
  13. Re: SIP and FastStart (VoIP Newbie)
  14. ACD Methods (Marshall, Ed)
  15. Can't make outgoing calls (Nick Heinemans)
  16. Re: Asterisk real time extensions problem... (Gentian Bajraktari)
  17. AS5300 -> Meridian Configuration (Aaron Daniel)
  18. Re: Phone keypad input not working during "menu's" (Don)
  19. Re: Public vs. Private Network (William Suffill)
  20. 3com 3101 SIP configuration (rbauer)

----------------------------------------------------------------------

Message: 1
Date: Thu, 19 May 2005 15:15:06 +0000 (UTC)
From: Kevin McCauley <kevin at fullcirclenetworks.com>
Subject: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc
To: asterisk-users at lists.digium.com
Message-ID: <loom.20050519T171208-388 at post.gmane.org>
Content-Type: text/plain; charset=us-ascii

Anton Krall <akrall-lists <at> intruder.com.mx> writes:

>> 
>> Guys, anybody having problem with ilbc and GS ata 286? I just tried it
>for
>> fun (always using alaw) and voices sounded quite bad... crappy voice
>> prompts, not bad quality, just like weird noises.
>> 
>> Anybody had this? whats the latest FW for those units?
>> 
>> _______________________________________________
>> 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
>> 
>> 
>
>
>
>
Anton,

I use iLBC exclusively on the 286/486 and it interoperates with other
devices on
my network fine.  In fact I use iLBC because some of the people I talk to
only
have dialup and it works the best for that.  

I will mention though, that I have stayed on FW version 1.0.5.16 since I
have
had troubles with newer versions.

-Kevin

------------------------------

Message: 2
Date: Thu, 19 May 2005 09:37:33 -0600
From: "admin" <dsanders at purecom.com>
Subject: Re: [Asterisk-Users] MusicOnHold probelms
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <WorldClient-F200505190937.AA37330001 at purecom.com>
Content-Type: text/plain

Do you have mpg123 installed?
Is there a .mp3 file available to play in your /var/lib/asterisk/mohmp3 
directory?

-daryl

-----Original Message-----
From: chawki hammoud <cyhammoud at yahoo.com>
To: Asterisk-Users at lists.digium.com
Cc: 
Date: Thu, 19 May 2005 06:03:55 -0700 (PDT)
Subject: [Asterisk-Users] MusicOnHold probelms

>> This is my second attempt trying to get help and I am
>> hoping someone can. When the musiconhold extension is
>> matched, Asterisk attempts to execute musiconhold and
>> stops right away, this is what I gets:
>> 
>>  Executing MusicOnHold("OSS/dsp", "") in new stack
>>     -- Started music on hold, class 'default', on
>> OSS/dsp
>>     -- Stopped music on hold on OSS/dsp
>> 
>> Is there a file that musiconhold try to play and can't
>> find. Please help withy any suggestions.
>> 
>> 
>> 
>>       
>> Discover Yahoo! 
>> Stay in touch with email, IM, photo sharing and more. Check it out! 
>> http://discover.yahoo.com/stayintouch.html
>> _______________________________________________
>> 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: 3
Date: Thu, 19 May 2005 11:44:39 -0400
From: "Ariel Batista" <arielb27 at hotmail.com>
Subject: RE: [Asterisk-Users] SIP Phone Recommendations?
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID: <BAY104-DAV156696C5A973C8D2F9623CDB080 at phx.gbl>
Content-Type: text/plain;	charset="us-ascii"

Just want to let everyone know that even if there changing it out to the
new
501 it's still on of the best. Remember that people are still buying the
Cisco 7960G which is being phased out as well.

The IP-500 works and works very well. I know that there price will be going
down soon once there are some supplies of the IP-501.  But if you need a
phone now it is a very good one for the price.  

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michael
Graves
Sent: Thursday, May 19, 2005 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phone Recommendations?

On Wed, 18 May 2005 22:29:40 -0500, Kristian Kielhofner wrote:

>>Ariel,
>>
>>	It's probably not a good idea to reccomend the IP 500/300 anymore. 
>>They are being phased out by Polycom because they (and the IP 300) only 
>>have 2mb of flash, and Polycom is looking to standardize on 4mb for 
>>their firmware (which the IP 600 has had since day one).
>>
>>	If you are going to buy a Polycom now, get an IP 600, or, wait for
the 
>>301's or 501's.  Don't say I didn't warn you!
>
>
Good advice!. BTW, I LOVE my IP600's. 

I also kinda like the Zultys 4x4/4x5.The hardware and software is good
but their support arrangement is terrible. They provide no end user
support at all. Period. They rely upon their dealers to provide all
support, but then they're ok with signing up dealers that know nothing
about the products.

Michael

--
Michael Graves                           mgraves at pixelpower.com
Sr. Product Specialist                          www.pixelpower.com
Pixel Power Inc.                                 mgraves at mstvp.com

o713-861-4005
o800-905-6412
c713-201-1262

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------------------------------

Message: 4
Date: Thu, 19 May 2005 11:44:09 -0400
From: "Steve Totaro" <asterisk at totarotechnologies.com>
Subject: Re: [Asterisk-Users] Deleting Monitor Files After 2 Months
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID:
	<001801c55c89$a50e2610$9f0aa8c0 at cfigroup.computerfrontiers.com>
Content-Type: text/plain; format=flowed; charset="utf-8";
	reply-type=original

----- Original Message ----- 
From: "Matthew Boehm" <mboehm at cytelcom.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Thursday, May 19, 2005 10:04 AM
Subject: Re: [Asterisk-Users] Deleting Monitor Files After 2 Months

>> Gavin Hamill wrote:
>>> On Thursday 19 May 2005 13:51, Steve Totaro wrote:
>>>> Does anyone know the best way to automate the deletion of monitor
>>>> files after they age two months?
>>>
>>> How about ...
>>>
>>> $ find /path/to/files -ctime +60 -exec rm {}\;
>>>
>>> Cheers,
>>> Gavin.
>>
>> Nice Gavin. I would further turn that into a shell script and pop it
>into
>> cron to run nightly.
>>
>> -Matthew
>>
>
>
Thanks! 

------------------------------

Message: 5
Date: Thu, 19 May 2005 11:45:11 -0400
From: Tom Hayden <thayden at gmail.com>
Subject: [Asterisk-Users] Two TDM04 with Poweredge
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <e5f28f9205051908455d4c4778 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Has anyone on this list succesfully managed to get two (or more) TDM04
(with four FXO each) working on a Dell PowerEdge server? If so, which
model? Was it a hassle?  I'm doing a seven-line installation and a
callbank seems like overkill, I just don't want to get suck with a
PowerEdge that gets into an IRQ mess.

Thanks in Advance,

Tom Hayden

------------------------------

Message: 6
Date: Thu, 19 May 2005 10:54:44 -0500
From: Eric Wieling aka ManxPower <eric at fnords.org>
Subject: Re: [Asterisk-Users] Public vs. Private Network
To: Andrew Latham <lathama at gmail.com>,	Asterisk Users Mailing List -
	Non-Commercial Discussion	<asterisk-users at lists.digium.com>
Message-ID: <428CB6C4.8080207 at fnords.org>
Content-Type: text/plain; charset=windows-1252; format=flowed

>>>I am looking at connecting 7 ñ 10 locations together using Asterisk and
>>>possibly some VoIP gateway appliances.  I need to insure best voice
>quality
>>>as these trunks will be used primarily for customer calls.  I am
>considering
>>>implementing a full T1 frame relay circuit to each location which can be
>>>done for a reasonable cost.  DSL and Cable are currently at each
>location
>>>and setup for automatic failover.  Should I remove one of my public
>>>connections and replace it with a private circuit for best quality?
>
>
To run VoIP over Frame Relay you need your Port Speed to be the same 
as your CIR.  Cisco has extensive docs about this, but I'm too lazy to 
look them up right now.
-- 
Always do right. This will gratify some people and astonish the rest.
Mark Twain

------------------------------

Message: 7
Date: Thu, 19 May 2005 23:57:45 +0800
From: VoIP Newbie <voip.newbie at gmail.com>
Subject: Re: [Asterisk-Users] asterisk-oh323 build problems
To: FaberK <f.faberk at gmail.com>,	Asterisk Users Mailing List -
	Non-Commercial Discussion	<asterisk-users at lists.digium.com>
Message-ID: <62b5865d05051908576d725acc at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Read README file first. You will get a clue.

On 5/19/05, FaberK <f.faberk at gmail.com> wrote:
>> Hello Guys,
>> first of all, I'm very new with asterisk.
>> I'm trying to set it up. I've already compiled and installed
>Asterisk-1.0.7
>> Now I'm trying with asterisk-oh323
>> I've already installed pwlib, oh323 and I've already set the variables.
>> Now, when I try to "make" asterisk-oh323 I receive this error messagge:
>> for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
>> make[1]: Entering directory `/root/voip/asterisk/asterisk-oh323/wrapper'
>> g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL
>> -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3
>> -DNDEBUG -I/usr/include -I/usr/include/crypto
>> -I/usr/lib/pwlib/include/ptlib/unix -I/usr/lib/pwlib/include
>> -I/usr/lib/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o
>> wrapper.o
>> wrapper.cxx: In constructor
>>   `WrapH323Connection::WrapH323Connection(WrapH323EndPoint&, unsigned
>int,
>>   int, int, short unsigned int)':
>> wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this
>function)
>> wrapper.cxx:563: (Each undeclared identifier is reported only once for
>each
>>   function it appears in.)
>> wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)':
>> wrapper.cxx:1230: warning: unused variable
>`ClearCallThread*clearCallThread'
>> make[1]: *** [wrapper.o] Error 1
>> make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper'
>> make: *** [subdirs_all] Error 1
>> 
>> 
>> What's wrong?
>> 
>> Thanks
>> 
>> --
>> .:FaberK:.
>> _______________________________________________
>> 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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>
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>
------------------------------

Message: 8
Date: Thu, 19 May 2005 09:12:22 -0700
From: Johnathan Corgan <jcorgan at aeinet.com>
Subject: Re: [Asterisk-Users] Outbound dialing issue with FXO
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <428CBAE6.4050601 at aeinet.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Mike Clark wrote:

>> However, outbound calls are hit or miss. Sometimes they work fine and 
>> other times we get a "you must first dial a 1 or 0" message back from 
>> telco when dialing out standard POTS lines.
>
>
Did you get this working yet?

-Johnathan

------------------------------

Message: 9
Date: Thu, 19 May 2005 12:16:18 -0400
From: David Brodbeck <DavidB at mail.interclean.com>
Subject: RE: [Asterisk-Users] Two TDM04 with Poweredge
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
	<asterisk-users at lists.digium.com>
Message-ID:
	<C823AC1DB499D511BB7C00B0D0F0574CC4116F at serverdell2200.interclean.com>
Content-Type: text/plain;	charset="iso-8859-1"

>> -----Original Message-----
>> From: Tom Hayden [mailto:thayden at gmail.com]
>
>
>> Has anyone on this list succesfully managed to get two (or more) TDM04
>> (with four FXO each) working on a Dell PowerEdge server? If so, which
>> model? Was it a hassle?
>
>
I've got a PowerEdge 800 tower server with two of them.  Only five FXO
modules right now, though.

It mostly works.  When I insert the driver I get an NMI, but that appears
to
be harmless.  I have to unload and reload the drivers once a week or so,
otherwise the FXO modules tend to eventually stop responding.  I haven't
had
any audio quality or interrupt problems, though.  

The system gets the job done, but I can't wholeheartedly recommend these
cards.  If I had to do it all over again, I'd consider some other method.
I'm not sure if anything else would be practical, though.  A T1 card plus
channel bank is kind of cost prohibitive for such a small installation.
I've heard good things about the Sipura gateways, but I'm interfacing to a
PBX and need the ability to flash the line for transfers, and I think
Flash() is Zap-specific.

------------------------------

Message: 10
Date: Thu, 19 May 2005 11:33:29 -0500
From: "Anton Krall" <akrall-lists at intruder.com.mx>
Subject: RE: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID: <20050519163306.839BE5B40EF at intruder.com.mx>
Content-Type: text/plain;	charset="us-ascii"

That's what I was starting to think.. Since I've always used ulaw or
alaw...
Seems that firmware 1.0.5.23 has ilbc broken. 

------------------------------

Message: 11
Date: Thu, 19 May 2005 09:33:55 -0700 (PDT)
From: M O <martinoshield at yahoo.com>
Subject: [Asterisk-Users] (no subject)
To: asterisk-users at lists.digium.com
Message-ID: <20050519163355.9122.qmail at web30312.mail.mud.yahoo.com>
Content-Type: text/plain; charset=us-ascii

BJ,

>>BJ Weschke <bweschke at gmail.com>
>>Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom
>>SIP termination vs. DS3
>>To: Asterisk Users Mailing List - Non-Commercial
>>Discussion <asterisk-users at lists.digium.com>
>>Message-ID:
<79cf63305051908056c284cc9 at mail.gmail.com>
>>Content-Type: text/plain; charset=ISO-8859-1
>
>
>>Did I miss pricing/availability announcements from
>>Digium on that DS3 card somewhere? 
>
>
No idea.  You can contact them if you dont know what
you missed :) 

>>I wasn't aware they were going to be GA in less than
3
>>weeks from now.
>
>
>>From my standpoint, I am just so anxious and 
confident that the Digium DS3 Channelized Voice PCI
Card, whenever I get my order of DID #'s and test my
configuration of Asterisk, that I am willing to
prepay, 
or have available to Digium, whatever $$$ they want 
for the card.

I am EVENTUALLY going to need it anyways, so I dont
mind prepaying wheather or not it is available today!
My knowledge of their product offering is no different

than yours.  But I fully intend on purchasing it :)!

We are starting off with a 100Mbps burstable bandwith,
though exspensive to start, after 30 days of usage, my
bandwidth costs will look like $25K.  Going off the
top of head for a Sangoma DS3 Card @ $6000 per card,
If I got 2 of them for $12,000 total, I eliminate,
almost, that $25,000 per month bandwidth cost to me.

So if Digiums DS3 Channelized Voice PCI card costs,
around what Sangomas costs, $6,000, (JUST AS A EXAMPLE
FOR THIS POST), $12,000 for 2 Digium DS3's in 1 month,
I will save almost $10,000 AUTOMATICALLY and ever
month thereafter! :)

Come on Txlink DID #'s.

Come on Digium with the DS3 Channelized Voice PCI
card.

Then all Digium would have left to do is create a
board
or work with someone on getting Radio Waves into your
computer.  :)

Sincerely,

SoftwareRadioGuy

__________________________________ 
Yahoo! Mail Mobile 
Take Yahoo! Mail with you! Check email on your mobile phone. 
http://mobile.yahoo.com/learn/mail 

------------------------------

Message: 12
Date: Fri, 20 May 2005 00:36:58 +0800
From: VoIP Newbie <voip.newbie at gmail.com>
Subject: [Asterisk-Users] Re: Always Ringing
To: Asterisk-Users at lists.digium.com
Message-ID: <62b5865d050519093621b36e6f at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Can anyone give me a big hand here??

On 5/16/05, VoIP Newbie <voip.newbie at gmail.com> wrote:
>> Hi all,
>> 
>> I am using chan_h323 from Asterisk CVS to interconnect with GNUGK
>> v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on
>> Asterisk. However, I only heard ringing when the call was answered on
>> SIP side. Below is the debug from chan_h323. Any help is welcome.
>> Thanks.
>> 
>> *CLI>   == New H.323 Connection created.
>>    -- Setting up Call
>>    --  Call token:  [ip$22.7.20.32:30012/16050]
>>    --  Calling party name:  [6907]
>>    --  Calling party number:  [6907]
>>    --  Called party name:  [0069777]
>>    --  Called party number:  [0069777]
>>        --Received SETUP message
>>        =-= In OnAnswerCall for call 16050
>>                - Progress Indicator: 0
>>                - Inserting PI of 0 into ALERTING message
>>        -- Started logical channel: sending G.729
>>                -- channelsOpen = 1
>>                External RTP Session Starting
>>                RTP channel id 1 parameters:
>>                -- remoteIpAddress: 22.7.20.32
>>                -- remotePort: 51048
>>                -- ExternalIpAddress: 0.0.0.0
>>                -- ExternalPort: 17816
>>        -- Started logical channel: receiving G.729
>>                -- channelsOpen = 2
>>                External RTP Session Starting
>>                RTP channel id 1 parameters:
>>        ExternalRTPChannel Destroyed
>>        ExternalRTPChannel Destroyed
>>    -- Executing Dial("H323/ip$22.7.20.32:30012/16050", "SIP/69777")
>> in new stack
>>    -- Called 69777
>>    -- SIP/69777-c6ce is ringing
>>        Sending alerting
>> 
>>    -- SIP/69777-c6ce answered H323/ip$22.7.20.32:30012/16050
>>        Answering call ip$22.7.20.32:30012/16050
>>        -- Transmitting RFC2833 on payload 96
>>        -- Received Facility message...
>>        =-= In OnConnectionEstablished for call 16050
>>                -- Connection Established with "6907 [22.7.20.32]"
>>        -- Received Facility message...
>>        -- Started logical channel: receiving G.729
>>                -- channelsOpen = 3
>>                External RTP Session Starting
>>                RTP channel id 1 parameters:
>>        -- Received Facility message...
>>        -- Received RELEASE COMPLETE message...
>>        -- ClearCall: Request to clear call with token
>> ip$22.7.20.32:30012/16050, cause EndedByRemoteUser
>>        -- Sending RELEASE COMPLETE
>>                channelsOpen = 2
>>                channelsOpen = 1
>>                channelsOpen = 0
>>        ExternalRTPChannel Destroyed
>>        ExternalRTPChannel Destroyed
>>        ExternalRTPChannel Destroyed
>>        -- ClearCall: Request to clear call with token
>> ip$22.7.20.32:30012/16050, cause EndedByTransportFail
>>  == Spawn extension (default, 0069777, 1) exited non-zero on
>> 'H323/ip$22.7.20.32:30012/16050'
>> -- 6907 [22.7.20.32] has cleared the call
>>        == H.323 Connection deleted.
>>
>
>
>
>
------------------------------

Message: 13
Date: Fri, 20 May 2005 00:38:46 +0800
From: VoIP Newbie <voip.newbie at gmail.com>
Subject: [Asterisk-Users] Re: SIP and FastStart
To: Asterisk-Users at lists.digium.com
Message-ID: <62b5865d0505190938e93b184 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Can anyone give me a big here?

On 5/13/05, VoIP Newbie <voip.newbie at gmail.com> wrote:
>> I am using Asterisk-oh323 v0.7.1 with GNUGK. Please advise what must
>> be done  to make FastStart work with SIP phones. Thanks.
>> 
>> On 5/12/05, VoIP Newbie <voip.newbie at gmail.com> wrote:
>> > Hi all,
>> >
>> > When I enabled faststart in oh323.conf, calls from H323 endpoint to
>> > SIP phones could not complete. The originating phone kept ringing when
>> > calls were answered by SIP phones.
>> >
>> > fastStart=yes
>> > h245Tunnelling =yes
>> > h245inSetup=yes
>> >
>> > Please can you advise.
>> >
>> > Many Thanks.
>> >
>>
>
>
>
>
------------------------------

Message: 14
Date: Thu, 19 May 2005 17:43:52 +0100
From: "Marshall, Ed" <ed.marshall at cetelem.co.uk>
Subject: [Asterisk-Users] ACD Methods
To: "'asterisk-users at lists.digium.com'"
	<asterisk-users at lists.digium.com>
Message-ID: <8ADF3F2FBF4F2C408C4C47B05CA3D267121A0C at HCCEXC02>
Content-Type: text/plain;	charset="iso-8859-1"

Can anyone point me in the right direction of info regarding ACD methods
available in Asterisk.

As far as I can see there are time based ring strategies available but I
cannot find any info regarding skills based routing or queue priorities.

Also do the current time based ring strategies work globally.  What I mean
by this is if an agent is a member of more than one queue then would the
ACD
algorithm take this into account before deciding to allocate another call ?

Any help would be much appreciated.

Regards
Ed

------------------------------

Message: 15
Date: Thu, 19 May 2005 18:46:42 +0200
From: "Nick Heinemans" <nick at heinemans.net>
Subject: [Asterisk-Users] Can't make outgoing calls
To: <asterisk-users at lists.digium.com>
Message-ID: <20050519164640.848BC18DE8 at olive.qinip.net>
Content-Type: text/plain;	charset="us-ascii"

Hello,

When I try to make an outgoing call from my X-lite softphone connected to
Asterisk, I keep getting the following error message:
May 19 18:42:58 WARNING[3086]: Forbidden - wrong password on authentication
for INVITE to '"31307110340"
<sip:31307110340 at 84.41.149.228>;tag=as13ba1ff7'

I'm running AAH 1.0 on a server which is directly hooked up to my ADSL
line.
It's second NIC is connected to my LAN on which the PC with X-lite is also
connected. I've configured the Asterisk server as a NAT router and I opened
UDP ports 5060 and 10000-20000 from the outside.

Any idea what might be wrong?

Regards, Nick

------------------------------

Message: 16
Date: Thu, 19 May 2005 17:50:38 +0100
From: "Gentian Bajraktari" <g.bajraktari at afb.net.al>
Subject: Re: [Asterisk-Users] Asterisk real time extensions problem...
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users at lists.digium.com>
Message-ID: <00c301c55c92$e7051de0$0300a8c0 at mozart>
Content-Type: text/plain; charset="iso-8859-1"

HI,

The problem is that you are using: incoming-next,60069,1
Use:  incoming-next|60069|1 instead

RG,
Gentian

  ----- Original Message ----- 
  From: Bharat M. Sarvan 
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  Cc: asterisk-dev at lists.digium.com 
  Sent: Sunday, April 17, 2005 11:52 AM
  Subject: [Asterisk-Users] Asterisk real time extensions problem...

  Hello everybody,

                           I have setup asterisk real time extensions and
its working pretty well. But the problem is when I am jumping between the
contexts using the Goto statement in the database. I  am getting a error 

  = Parsing '/etc/asterisk/sip_notify.conf': Found

      -- SIP Seeding peers from Astdb: 'ezzibpo4' at
ezzibpo4 at 210.211.246.47:5061 for 60

      -- Executing Goto("SIP/ezzibpo4-4636", "incoming-next,6069,1")

  May 19 05:00:04 NOTICE[6420]: pbx.c:1688 pbx_extension_helper: Cannot
find extension '6069' in context 'incom'

  May 19 05:00:04 WARNING[6420]: pbx.c:6256 ast_parseable_goto: Priority
'incoming-next,

  The structure of the extensions db is as given below

+----+---------------+-------+----------+-----------------+----------------------+

  | id | context           | exten | priority | app             | appdata 
            |

+----+---------------+-------+----------+-----------------+----------------------+

  |  1 | incoming        | 6069  |        1 | Goto            |
incoming-next,6069,1 |

  |  2 | incoming        | 6069  |        2 | Hangup          |           
          |

  |  3 | incoming-next | 6069  |        1 | DigitTimeout    | 10          
        |

  |  4 | incoming-next | 6069  |        2 | ResponseTimeout | 30          
        |

  |  5 | incoming-next | 6069  |        3 | Background      | welcome     
        | 

        The context "incom" in the above error is the context defined for
placing outgoing call in the sip.conf file. I don't understand why is it
looking for extension 6069 in the "incom" context.

       The "Goto" statement in the context incoming is getting executed
without any probs, but the control is not getting transferred to the
context "incoming-next" upon execution of the Goto statement. 

        Could anybody suggest me as to where might the problem be and any
way to get rid of this problem. Please do reply..

  Regards,

  Bharat M. Sarvan

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------------------------------

Message: 17
Date: Thu, 19 May 2005 11:51:15 -0500
From: Aaron Daniel <amdtech at shsu.edu>
Subject: [Asterisk-Users] AS5300 -> Meridian Configuration
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <ED6ACD5A-E677-43A4-A1E4-1502ABACF545 at shsu.edu>
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed

We're trying to set up a connection between an AS5300 and a meridian  
CSU/DSU so our asterisk system can interconnect with our current  
legacy system, and for some reason the T1 connection will not come up  
whatsoever.  I've gone through all the configurations I can think of,  
even basically copied our current cisco settings directly to the  
AS5300 so they would be nearly identical, and nothing.  Any help  
would be appreciated.

AS5300 config:

Current configuration : 2094 bytes
!
! Last configuration change at 11:45:56 CDT Thu May 19 2005
! NVRAM config last updated at 11:46:14 CDT Thu May 19 2005
!
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname cdial2
!
boot-start-marker
boot-end-marker
!
enable secret 5 *******************
enable password *******************
!
spe 1/0 1/7
firmware location system:/ucode/mica_port_firmware
spe 2/0 2/7
firmware location system:/ucode/mica_port_firmware
!
!
resource-pool disable
clock timezone CST -6
clock summer-time CDT recurring
!
no aaa new-model
ip subnet-zero
no ip routing
ip finger
ip domain name shsu.edu
ip name-server 158.135.1.20
ip name-server 158.135.1.200
!
!
isdn switch-type primary-5ess
isdn voice-call-failure 0
!
!
!
!
!
!
!
!
!
!
!
!
controller T1 0
shutdown
framing sf
linecode ami
!
controller T1 1
shutdown
framing sf
linecode ami
!
controller T1 2
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
controller T1 3
shutdown
framing sf
linecode ami
!
!
interface Ethernet0
no ip address
no ip route-cache
shutdown
!
interface Serial2:23
no ip address
ip mroute-cache
dialer-group 1
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice modem
isdn disconnect-cause 1
fair-queue 64 16 3
no cdp enable
ip rsvp bandwidth
ip rtp reserve 10000 10000
!
interface FastEthernet0
ip address 158.135.1.61 255.255.0.0
no ip route-cache
no ip mroute-cache
duplex full
speed 100
no mop enabled
!
ip classless
no ip http server
!
!
!
!
!
!
!
dial-peer voice 6 voip
incoming called-number 6....
destination-pattern 6....
session protocol sipv2
session target sip-server
!
dial-peer voice 4 pots
application session
direct-inward-dial
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server dns:sipproxy1.shsu.edu
!
!
line con 0
line 1 96
line aux 0
line vty 0 4
password *******************
login
!
scheduler interval 1000
ntp clock-period 17180204
ntp update-calendar
ntp server 158.135.1.2
!
end

meridian config:

ADAN     DCH 13
    CTYP MSDL
    GRP  0
    DNUM 6
    PORT 3
    DES  ASVOIP
    USR  PRI
    DCHL 25
    OTBF 32
    PARM RS422  DTE
    DRAT 64KC
    CLOK EXT
    IFC ESS5
    SIDE USR
    CNEG 1
    RLS  ID   36
    RCAP ND2
    T200 3
    T203 10
    N200 3
    N201 260
    K    7

Thanks,

Aaron Daniel
Senior Voice Analyst
Sam Houston State University

------------------------------

Message: 18
Date: Thu, 19 May 2005 09:54:36 -0700
From: Don <asterisk at geeksrus.ca>
Subject: Re: [Asterisk-Users] Phone keypad input not working during
	"menu's"
To: asterisk-users at lists.digium.com
Message-ID: <428CC4CC.8080502 at geeksrus.ca>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Wilson Pickett wrote:

>>What codec are your phones using and which do you have in sip.conf in
>>general and phone entries?
>>  
>>
>
>
Hi Wilson,

Thanks for the reply.  I didn't know anything about codecs but I've 
tried to look up what I can.  The Polycom documentation (SIP admin 
guide) says the hone supports G.711u-law, G.711a-law, G.729AB, SID and 
RFC2833.  The phone configuration files say that the preference is for 
u-law, a-law and AB in that order.  My sip.conf file says:

disallow=all
allow=ulaw
allow=alaw

I would guess that means I'm ok (i.e. ulaw is good on both sides) but 
this is a new area of * for me.  What do you think?

Don

------------------------------

Message: 19
Date: Thu, 19 May 2005 12:56:22 -0400
From: William Suffill <william.suffill at gmail.com>
Subject: Re: [Asterisk-Users] Public vs. Private Network
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <6b65470d05051909562a778d83 at mail.gmail.com>
Content-Type: text/plain; charset=WINDOWS-1252

Point to Point connectivity if they are close enough. Only use
DSL/Cable if you have to since results may vary depnding on
location/route/utilization/ISP.

On 5/19/05, Andrew Latham <lathama at gmail.com> wrote:
>> yes
>> 
>> On 5/19/05, David Sampson <dsampson at innseason.com> wrote:
>> >
>> >
>> >
>> > Hello ñ
>> >
>> >
>> >
>> > I am looking at connecting 7 ñ 10 locations together using Asterisk
>and
>> > possibly some VoIP gateway appliances.  I need to insure best voice
>quality
>> > as these trunks will be used primarily for customer calls.  I am
>considering
>> > implementing a full T1 frame relay circuit to each location which can
>be
>> > done for a reasonable cost.  DSL and Cable are currently at each
>location
>> > and setup for automatic failover.  Should I remove one of my public
>> > connections and replace it with a private circuit for best quality?
>> >
>> >  Thank you,
>> >
>> >
>> >  Dave
>> >
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>> 
>> 
>> --
>> <sig>
>> Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
>> WWW: http://lathama.com
>> Email: lathama at lathama.com - lathama at yahoo.com - lathama at gmail.com
>> If any of the above are down we have bigger problems than my email!
>> </sig>
>> 
>> _______________________________________________
>> 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 20: 
Is there anyone that has been able to get a 3com IP phone to function with
asterisk. I have a couple of 3com 3101's and I'm stuck trying to figure
out what I'm missing. There really shouldn't be a big difference between
the Cisco 7960's and my 3com 3101, should there be? As long as it's SIP
compliant?
This is my first time posting out here so if this posting has "newbie"
written all over it, it's true. The important thing is that I'm very
interested and excited about Asterisk as are all of you. Thanks in advance.

_______________________________________________
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Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users

End of Asterisk-Users Digest, Vol 10, Issue 154
***********************************************