Re: [Asterisk-Users] DTMF with IConnectHere fix

[alex]

> I am looking for someone to confirm that Iconnecthere sends DTMF either
> in-band or out-of-band.  If it is out-of-band, then I'll need to work
> with someone to figure out the issue.  If it is in-band then I will need
> to add in-band DTMF detection to SIP.
It is in-band. My original SIP patch contained code to do inband DTMF 
detection in a very ugly way, which apparently got dropped when it was 
integrated.

-alex

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Re: [Asterisk-Users] Found: inexpensive ADSI phone

[alex]

Yes, I have bought 10 of those units from these guys, and they are
unlocked. Highly recommended.

-alex

On Sun, 2 Mar 2003, John Todd wrote:

> I know I heard people looking for inexpensive ADSI phones a while 
> back.  Can't vouch for these guys, but this looks like a reasonable 
> deal for what perhaps are new phones.  No experience with ADSI, 
> myself, but thought I'd pass it along.  These folks also carry the 
> Nortel analog units that do intercom and 2-line, as has been 
> discussed in prior threads.
> 
> JT
> 
> 
> -
> http://lktelecom.zoovy.com/product/HPT350
> 
> PowerTouch 350
> 
> Excerpt:
> 
> Analog Display Services Interface (ADSI)* Caller ID, Call Waiting 
> Display, and Call Waiting Options* Six softkeys 8x20 hybrid backlit 
> display Speakerphone with mute 25-name and number Call Log* Single 
> screen implementation Personal Directory for 50 names and numbers 
> Services key to access a wide range of ADSI services Two-position 
> tilt-up base CLASS Message Waiting indicator* Adjustable 
> volume/alerter control Print and Copy keys Last five number redial 
> Preferred Name Match* Visual ringing/extension-in-use/hold indicator 
> Hold and Flash keys Contrast control Editing capabilities via scroll 
> key Programmable call timer Area code stripping Goodbye and Options 
> keys Printer port English/Spanish prompts Hearing aid compatible ADA 
> compliant Desk or wall mountable
> 
> 
> Price: $49.99
> 
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Re: [Asterisk-Users] Fax support?

[alex]

On Mon, 3 Mar 2003, Brian Johnson wrote:

> How about data/fax handling (and detection)?
> 
> I think I remember reading (on this list) that asterisk could handle fax
> but not data but don't remember for sure
Potentially, yes. (Using late Tony Fisher's fax code), but practically no:
a) license on Tony's code is unclear, and he isn't around to clarify
b) Its incomplete

-alex

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Re: [Asterisk-Users] SIP INVITEs borked with iconnecthere

[alex]

John,

A heads-up: iconnect has apparently put up a filter against my IP address, 
for whichever reason (apparently they don't like people using asterisk?).

I've sent them an email and am pursuing this also through sales side (I'm 
about to make a resale deal with them), so hopefully tomorrow I'll find 
out just what they don't like about asterisk.

To check if they put up a filter, just do a telnet 213.137.73.141 5060

If you get connection refused, its something else. if it times out, you've 
been filtered.

-alex

On Wed, 5 Mar 2003, John Todd wrote:

> 
> Symptoms: when calling my iconnect phone number (13033913323 in my 
> bogus example below) from my cell phone, I can see that the call 
> makes it to my asterisk server, and my phones even ring once as * 
> passes the call through during the "180 Ringing" period.  However, it 
> seems that iconnecthere.com cannot see my "100 Trying" and "180 
> Ringing" messages, as they continue to send INVITES to me.  After two 
> seconds, they either give up or error out and send a CANCEL message.
> 
> To further increase my suspicions of something weird in the ability 
> to "see" my replies, they send 11 CANCEL messages over the period of 
> 30 seconds, despite my "200 OK" replies.
> 
> 
> Notes: 204.31.11.32 resolves to asterisk.something.com.  204.31.11.35 
> is my ATA-186.  Neither the domains nor the IP addresses are real, 
> except when referencing iconnect servers.
> 
> 213.137.73.176 is the real IP address of the SIP proxy at 
> iconnecthere.com (deltathree.com)
> Note that I actually do my REGISTERs against 213.137.73.178, not .176 
> - not a big deal, but who knows what clues will be helpful.
> 
> Unsuccessful Asterisk->iconnect->PSTN call:
> 
> tethereal port 5060 and host 213.137.73.176
> Capturing on fxp0
>0.00 213.137.73.176 -> asterisk.something.com SIP/SDP Request: 
> INVITE sip:[EMAIL PROTECTED]:5060, with session description
>0.001293 asterisk.something.com -> 213.137.73.176 SIP Status: 100 Trying
>0.039058 asterisk.something.com -> 213.137.73.176 SIP Status: 180 Ringing
>0.490181 213.137.73.176 -> asterisk.something.com SIP/SDP Request: 
> INVITE sip:[EMAIL PROTECTED]:5060, with session description
>0.490497 asterisk.something.com -> 213.137.73.176 SIP Status: 100 Trying
>1.530125 213.137.73.176 -> asterisk.something.com SIP/SDP Request: 
> INVITE sip:[EMAIL PROTECTED]:5060, with session description
>1.530439 asterisk.something.com -> 213.137.73.176 SIP Status: 100 Trying
>2.070160 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>2.070461 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>2.594680 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>2.595419 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>3.634908 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>3.635179 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>5.674595 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>5.674889 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>9.664659 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>9.664956 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>   13.645471 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>   13.645755 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>   17.635194 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>   17.635502 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>   21.665856 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>   21.666146 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>   25.676487 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>   25.676767 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>   29.666942 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>   29.667231 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
>   33.647019 213.137.73.176 -> asterisk.something.com SIP Request: 
> CANCEL sip:[EMAIL PROTECTED]:5060
>   33.647305 asterisk.something.com -> 213.137.73.176 SIP Status: 200 OK
> 
> 
> 
> 
> A successful ATA-186 to iconnect session, no Asterisk server involved:
> 
> 1338.126534 213.137.73.17

Re: [Asterisk-Users] ISPs with QoS for VoIP?

[alex]

The only case where QoS is useful is on tail-end circuits. Everywhere 
else, having bigger pipes is much more preferable to QoS.

-alex

On Fri, 7 Mar 2003, Eric Wieling wrote:

> I'm wondering if anyone knows of ISPs with service that has QoS
> features that would be good to use with VoIP stuff.  Granted,
> the QoS would only be supported as long as you stayed within
> their network, but it might be better than nothing.
> 
> --Eric
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Re: [Asterisk-Users] Interesting VoIP device

[alex]

> Configurable fxo/fxs with dual ethernet.
> 
> 
> http://www.tekdigitel.com/website/htmlPages/content/products/product_introductions/introduction_to_V-SERVER_iGATE_Dual_Ethernet.htm


Its easy to make a device with lots of options, question is the price. 

There are many ATA-like solutions, but people converge on ATA because of
sub-200$ price. The 'sweet spot' for VoIP gateways is 100$/port. The
'super sweet' (the point where it starts replacing regular phones) is
50$/port...

-alex

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Re: [Asterisk-Users] iconnect quality?

[alex]

> 1 - From watching the udp fly by, it seems that iconnect does not know
> when we hang up.  For example, if I call a voice mail and it starts
> giving me its speal and I hang up, iconnect stays connected until the VM
> hangs up at its end.
Because Asterisk doesn't implement RTCP. 


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Re: [Asterisk-Users] Dlink DG-104S

[alex]

Sort of. There are couple of things broken if you want to make calls from 
MGCP to SIP, and some things need to be implemented (hook flash in MGCP), 
retransmission (or at least timing out) of MGCP messages. I have 
preliminary patch which is not correct by any means but at least makes it 
usable. 

 On Fri, 28 Mar 2003, Brian Capouch 
wrote:

> Does anyone know if this unit works with Asterisk?
> 
> Thx.
> 
> B.
> 
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Re: [Asterisk-Users] Buying channel banks online?

[alex]

www.channelbanks.com

I have one unit. It works. 

-alex

On Wed, 2 Apr 2003, WipeOut . wrote:

> Hi,
> 
> Anyone know of any sites selling channel banks online (apart from ebay)..
> 
> Preferably with international shipping..
> 
> Thnaks
> 

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RE: [Asterisk-Users] I have a strange problem with ICH calls

[alex]

The root issue is that ICH today stopped accepting any format other than 
g.723.1 (which Asterisk doesn't support).

-alex

On Tue, 30 Sep 2003, Andrew Joakimsen wrote:

> Try 
> 
> exten => _71NXXNXX,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
> 
> or
> 
> exten => _7.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
> 
> Regards,
>  
>  
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of listas iPfone
> > Sent: Tuesday, September 30, 2003 4:52 PM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] I have a strange problem with ICH calls
> > 
> > Ok
> > 
> > extensions.conf:
> > 
> > exten => _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
> > 
> > sip.conf:
> > 
> > register =>31451543:[EMAIL PROTECTED]/33
> > 
> > [iconnect]
> > type=friend
> > secret=
> > username=31451543
> > host=sipauth.deltathree.com
> > dtmfmode=inband
> > context=from-sip
> > 
> > miklos
> > 
> > 
> > 
> > 
> > - Original Message -
> > From: "Andrew Joakimsen" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Tuesday, September 30, 2003 5:27 PM
> > Subject: RE: [Asterisk-Users] I have a strange problem with ICH calls
> > 
> > 
> > > Please post your extensions.conf and sip.conf sections relevant to
> > > ich/deltathree.
> > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED]
> [mailto:asterisk-users-
> > > > [EMAIL PROTECTED] On Behalf Of listas iPfone
> > > > Sent: Tuesday, September 30, 2003 3:33 PM
> > > > To: [EMAIL PROTECTED]
> > > > Subject: [Asterisk-Users] I have a strange problem with ICH calls
> > > >
> > > > Hi!
> > > >
> > > > I have a strange problem with ICH calls.
> > > >
> > > > When i try to make a call with asterisk for ICH nothing happens (
> > > register
> > > > is ok)
> > > >
> > > > But when i register my snom 200 with ich it works very well with
> the
> > > same
> > > > register data.
> > > >
> > > > Someone knows anything about?
> > > >
> > > > miklos
> > > >
> > > > ___
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> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > ___
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> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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[Asterisk-Users] Firewalls

[alex]

Well, a SIP client inside a netword with a firewall without portforwarding to this
, and a asterisk server in another network, this have a internet public ip.

The client can connect to the server?, this is possible?

The port forwarding is neccesary?, only will rule a client with portforwarding?

Is usable in this case stun or anything?

The Sip proxy is not a Solution.

Thanks.

-- 
Alejandro Escanero Blanco
Administrador Sistemas CEC.
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[Asterisk-Users] Problem with SIPPS and ilbc

[alex]

Well the warning is:
Mar 18 16:46:47 WARNING[737296]: Huh?  An ilbc frame that isn't a multiple of 50 bytes 
long from RTP (38)?

And the sound is cripple (really broken).

Any solution?

-- 
Alejandro Escanero Blanco
Administrador Sistemas CEC.
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Re: [asterisk-users] List delays

[Alex]

> Most of the users using this list do not experience the issue 
> you are having, rather than insult the admins, please trouble 
> shoot and if you cannot, at least post headers so others can.



Received: from lists.digium.com ([216.207.245.17]) by
nlpiport01.prodigy.net.mx
 with ESMTP; Thu, 12 Jul 2007 10:12:49 -0500

Received: from localhost ([127.0.0.1] helo=INXS.digium.internal)
by lists.digium.com with esmtp (Exim 4.63)
(envelope-from <[EMAIL PROTECTED]>)
id 1I893a-0004UJ-BE; Tue, 10 Jul 2007 01:18:42 -0500

Received: from exprod8mx6.postini.com ([64.18.3.106] helo=psmtp.com)
by lists.digium.com with smtp (Exim 4.63)
(envelope-from <[EMAIL PROTECTED]>)
 id 1I893R-0004Tz-9nfor asterisk-users@lists.digium.com; Tue,
 10 Jul 2007 01:18:33 -0500

Received: from source ([206.168.222.161]) by exprod8mx6.postini.com
([64.18.7.10]) with SMTP; Mon, 09 Jul 2007 23:18:32 -0700 (PDT)
Date: Tue, 10 Jul 2007 00:18:32 -0600


No mail system guru here, but it would seem lists.digium.com is indeed
holding on to email for 2 days.  It looks like those of us that are not in
the U.S. seem to have a problem with delays.


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Re: [asterisk-users] Time Limit on Call or Conference Room?

[Alex]

This might get you going:

http://www.voip-info.org/wiki/view/Asterisk+cmd+AbsoluteTimeout



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> JR Richardson
> Sent: Friday, August 03, 2007 1:49 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Time Limit on Call or Conference Room?
> 
> 
> Hi All,
> 
> I recently had an incident where a conf bridge was left open 
> due to improper disconnection.  I've read about the meetme 
> options and marked callers closing the bridge when they exit. 
>  This is OK for meetme, but I'm really interested in a call 
> timer that can be set on inbound and outbound calls within 
> the dial plan, per call.
> 
> I have another customer who wants to offer free calls, for 
> 5-10 minutes with auto disconnect.
> 
> Can anyone point me int he right direction?
> 
> Thanks.
> 
> JR
> -- 
> JR Richardson
> Engineering for the Masses
> 
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> 
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.476 / Virus Database: 269.11.4/936 - Release 
> Date: 8/4/2007 2:42 PM
> 
> 

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[asterisk-users] DID Provider

[Alex]

I have the same problem. Also, the web interface is really awkward, they
don't
have DIDs in the countries where I need them (Chile, for example), and the
quality of the sound is from bad to unusable, even from the US phone they
provide
you for free. If I would have the chance, I would have them refund me the
money
I spent on that service.

I am using other services based in US (for example, rapidvox), they work
fine
and have no hassles like signing NDAs, bad quality, etc.

If you know of any other DID wholesale provider, please tell me.

Regards,
Alex


I am using DIDx.net as my DID provider but they don't seem to get their

act

together. A lot of times the phone numbers don't work. How can provide my

own

DID, my asterisk server is being hosted at a Data center and has a

reliable

vendor that does my termination and do SIP to SIP and have no T1 channels.
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Re: [asterisk-users] Asterisk from Debian Packages

[Alex]

You can run Asterisk 1.2 in sarge using the packages in backports.

Just add:

deb http://www.backports.org/debian/ sarge-backports main contrib non-free

to /etc/apt/sources.list

then apt-get update

and then apt-get -t sarge-backports install asterisk

(you can also pin-priority asterisk's packages, look at APT documentation).

-Alex

On 12/10/06, Phil Finkler <[EMAIL PROTECTED]> wrote:


 Hi all,



I've gotten asterisk installed on Debian only to realize that the packaged
version is 1.0.7.  Is there a reason why they're not up to a 1.2.xrelease?  I'm 
building a system for production and I'm wondering if I should
remain at this old version or if there are any serious issues with 1.2.13on 
Debian?  Should I be able to do an apt-get from unstable and get
1.2.13 and be on my happy way?



Thanks for the help on a stupid question,

Phil



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RE: [asterisk-users] missing chan_zap.so

[Alex]

> Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. 
> All SIP accounts were working fine, today I tried to install 
> a fxs Sangoma A200 card and got the following error.

I believe you need to download the Sangoma drivers from their site.

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RE: [asterisk-users] ZT_CHANCONFIGfailedonchannel1:Nosuchdeviceoraddress

[Alex]

> Can anyone put me out of my misery?
> 
Do the devices exist in /dev/zap?  Check /var/log/messeges and look for
errors.  Seems the last time I battled this it ended up being a permission
problem.

Alex

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[asterisk-users] Error compiling patched pppd for zapras

[Alex]

hi everybody,

i'm tryint to install a asterisk system which acts as a dialin server
using a Digium Wildcard 205P.
acording to http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS i
need a patched version of pppd, but it does not compile on my system.

Linux box 2.6.17-gentoo-r8 #1 SMP Tue Sep 26 13:17:23 CEST 2006 x86_64
AMD Athlon(tm) 64 Processor 3200+ GNU/Linux
gcc -4.1.1, glibc-2.4
output of make is below.

any suggestions ?


Alex


cd chat; make  all
make[1]: Entering directory `/usr/src/ppp-2.4.1b2.WORKING/chat'
cc -c -O2 -g -pipe -DTERMIOS
-DSIGTYPE=void -UNO_SLEEP  
-DFNDELAY=O_NDELAY  -o chat.o chat.c
chat.c:215: warning: conflicting types for built-in function 'logf'
chat.c:1275:22: warning: trigraph ??) ignored, use -trigraphs to enable
cc -o chat chat.o
make[1]: Leaving directory `/usr/src/ppp-2.4.1b2.WORKING/chat'
cd pppd/plugins; make  all
make[1]: Entering directory `/usr/src/ppp-2.4.1b2.WORKING/pppd/plugins'
gcc -o minconn.so -shared -g -O2 -I.. -I../../include -fPIC minconn.c
gcc -o passprompt.so -shared -g -O2 -I.. -I../../include -fPIC passprompt.c
make -C pppoe -w pppoe.so
make[2]: Entering directory
`/usr/src/ppp-2.4.1b2.WORKING/pppd/plugins/pppoe'
gcc -g  -I.. -I../.. -I../../../include -D_linux_=1 -fPIC   -c -o
pppoe.o pppoe.c
In file included from pppoe.c:21:
pppoe.h:109:1: warning: "PTT_SRV_NAME" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:88:1: warning: this is the location of the
previous definition
In file included from pppoe.c:21:
pppoe.h:110:1: warning: "PTT_AC_NAME" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:89:1: warning: this is the location of the
previous definition
In file included from pppoe.c:21:
pppoe.h:111:1: warning: "PTT_HOST_UNIQ" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:90:1: warning: this is the location of the
previous definition
In file included from pppoe.c:21:
pppoe.h:112:1: warning: "PTT_AC_COOKIE" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:91:1: warning: this is the location of the
previous definition
In file included from pppoe.c:21:
pppoe.h:113:1: warning: "PTT_VENDOR" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:92:1: warning: this is the location of the
previous definition
In file included from pppoe.c:21:
pppoe.h:114:1: warning: "PTT_RELAY_SID" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:93:1: warning: this is the location of the
previous definition
In file included from pppoe.c:21:
pppoe.h:115:1: warning: "PTT_SRV_ERR" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:94:1: warning: this is the location of the
previous definition
In file included from pppoe.c:21:
pppoe.h:116:1: warning: "PTT_SYS_ERR" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:95:1: warning: this is the location of the
previous definition
In file included from pppoe.c:21:
pppoe.h:117:1: warning: "PTT_GEN_ERR" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:96:1: warning: this is the location of the
previous definition
In file included from pppoe.c:21:
pppoe.h:118:1: warning: "PTT_EOL" redefined
In file included from pppoe.h:37,
 from pppoe.c:21:
/usr/include/linux/if_pppox.h:87:1: warning: this is the location of the
previous definition
gcc -g  -I.. -I../.. -I../../../include -D_linux_=1 -fPIC   -c -o
pppoehash.o pppoehash.c
In file included from pppoehash.c:11:
pppoe.h:109:1: warning: "PTT_SRV_NAME" redefined
In file included from pppoe.h:37,
 from pppoehash.c:11:
/usr/include/linux/if_pppox.h:88:1: warning: this is the location of the
previous definition
In file included from pppoehash.c:11:
pppoe.h:110:1: warning: "PTT_AC_NAME" redefined
In file included from pppoe.h:37,
 from pppoehash.c:11:
/usr/include/linux/if_pppox.h:89:1: warning: this is the location of the
previous definition
In file included from pppoehash.c:11:
pppoe.h:111:1: warning: "PTT_HOST_UNIQ" redefined
In file included from pppoe.h:37,
 from pppoehash.c:11:
/usr/include/linux/if_pppox.h:90:1: warning: this is the location of the
previous definition
In file included from pppoehash.c:11:
pppoe.h:112:1: warning: "PTT_AC_COOKIE" redefined
In file included from pppoe.h:37,
 from pppoehash.c:11:
/usr/include/linux/if_pppox.h:91:1:

[Asterisk-Users] Polycom IP 600 not ringing

[Alex]

Hi guys,
Has anyone come across a problem when Polycom IP 600 does make an audible  
ring sound, even though the call comes in? I can see it on LCD and red  
light flashes. When I pickup the phone, everything is fine. It only  
applies to SIP calls. If the call comes in from PSTN via TDM400 card,  
everything seems to be ok.

The same scenario happens when I try to ring from IP600 other Polycom  
phones, again they don't ring.
I am using 2.6.1 bootrom and 1.4.1 SIP firmware.

Anyone can point me in the right direction?
Thanks,
Alex.
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Re: [Asterisk-Users] Polycom IP 600 not ringing

[Alex]

Sorry...
It should reed Polycom IP 600 does not make an  audible ring sound
...half a sleep :-)
On Thu, 07 Apr 2005 10:07:57 +1000, Alex <[EMAIL PROTECTED]> wrote:
Hi guys,
Has anyone come across a problem when Polycom IP 600 does not make an  
audible ring sound, even though the call comes in? I can see it on LCD  
and red light flashes. When I pickup the phone, everything is fine. It  
only applies to SIP calls. If the call comes in from PSTN via TDM400  
card, everything seems to be ok.

The same scenario happens when I try to ring from IP600 other Polycom  
phones, again they don't ring.
I am using 2.6.1 bootrom and 1.4.1 SIP firmware.

Anyone can point me in the right direction?
Thanks,
Alex.
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Re: [Asterisk-Users] Polycom IP 600 not ringing

[Alex]

Bugger :-(
On Wed, 06 Apr 2005 20:14:06 -0500, Eric Wieling <[EMAIL PROTECTED]> wrote:
Alex wrote:
Sorry...
It should reed Polycom IP 600 does not make an  audible ring sound
 ...half a sleep :-)
 On Thu, 07 Apr 2005 10:07:57 +1000, Alex <[EMAIL PROTECTED]> wrote:
Hi guys,
Has anyone come across a problem when Polycom IP 600 does not make an   
audible ring sound, even though the call comes in? I can see it on  
LCD  and red light flashes. When I pickup the phone, everything is  
fine. It  only applies to SIP calls. If the call comes in from PSTN  
via TDM400  card, everything seems to be ok.

The same scenario happens when I try to ring from IP600 other Polycom   
phones, again they don't ring.
I am using 2.6.1 bootrom and 1.4.1 SIP firmware.

Anyone can point me in the right direction?
I have seen this problem on the Polycom 500.  ONLY happens when I call  
from port 1 of my SPA-2000, works fine on port 2.  I cannot see any  
significant difference in the way the two ports are configured (on the  
device or in sip.conf).  I have no idea how to fix it.
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RE: [Asterisk-Users] Call Interception

[Alex]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Josiah Bryan
> Sent: Thursday, April 07, 2005 3:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Call Interception
> 
> There is no way to do that (that I know of) in the default Asterisk setup.
> 
> Which is I wrote a little Perl AGI script that lets users dial 200 to
> pickup a
> call. (Dial 200, then dial the extension at the prompt. The users phone
> then
> rings, with caller ID on the screen.) This works for any ringing channel
> on
> Asterisk, regardless of callgroup or pickupgroup. I suppose that could be
> added to 'limit' users, but its currently not implemented. You can pickup
> any
> channel that is ringing (SIP, Zap, etc.) with this script, since it just
> issues a Manager 'Redirect' action.
> 
> Usage:
> 
> exten => 200,1,AGI(pickup.pl)
> 
> If anyone is interested in pickup.pl, let me know and I'll see what I can
> do
> to make it available.

Hi,

I think it could be a very interesting workaround. I'd like to test your
script on my Asterisk, could you make it available for download or send it
to my e-mail address, please?

Thanks,

Alex

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[Asterisk-Users] LCDial and default provider

[Alex]

Does anybody know how I could set a default provider for LCDial? Also, how
could I use it for national calls, dialling without international prefix?

TIA,

Alex

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Re: [Asterisk-Users] Digium Card Issues

[Alex]

try in /etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-10
unused=11-15,17-31
dchan=16
defaultzone=au
loadzone=au
Cheers,
Alex.
On Fri, 22 Apr 2005 15:09:32 +1000 (EST), Sahil Gupta  
<[EMAIL PROTECTED]> wrote:

Hi,
I'm trying to configure a digium card here.  Got everything working  
sweetly apart from the last bit..

dmesg shows:
TE110P: Span configured for ESF/B8ZS
Calling startup (flags is 4099)
Registered tone zone 1 (Australia)
whilst /etc/zaptel.conf has:
span = 1,1,1,ccs,hdb3,crc4
bchan = 1-10
dchan = 16
defaultzone = au
loadzone = au
Any ideas?
Regards,
Sahil Gupta
VoiceValley
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[Asterisk-Users] PRI: Zap show channels

[Alex]

Hello,
Does anybody get any data in the 'Extension' column of the 'zap show
channels' output?  I'm at a loss as to where it would be getting
any information to populate this column. I have an E1 (10 channels) and  
channel 10 only shows the number in Extension column. Furthermore, that  
number is within indial range but it's configured as an extension.

Any clues
Thanks
Alex.
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Re: [Asterisk-Users] PRI: Zap show channels

[Alex]

Typo damn.
Should be read as "NOT configured as an extension"
Hello,
Does anybody get any data in the 'Extension' column of the 'zap show
channels' output?  I'm at a loss as to where it would be getting
any information to populate this column. I have an E1 (10 channels) and  
channel 10 only shows the number in Extension column. Furthermore, that  
number is within indial range but it's NOT configured as an extension.

Any clues
Thanks
Alex.
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Re: [Asterisk-Users] outgoing-call-logs to a text file

[Alex]

/var/log/asterisk/cdr-csv/Master.csv
On Thu, 12 May 2005 08:27:53 +0300, Kumara Jayaweera  
<[EMAIL PROTECTED]> wrote:

Greeting!,
I read somewhere that without cdr, Mysql etc it is possible to take
outgoing-call-logs to a text file. (I am not sure please). is it really
possible ? if so, how do I do it? any links to refer?
Thank you.
Kumara
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Re: [Asterisk-Users] French SIP or IAX phones

[Alex]

Polycom will do the trick..
On Fri, 13 May 2005 09:00:35 +0200, Dave Cotton  
<[EMAIL PROTECTED]> wrote:

On Thu, 2005-05-12 at 19:28 -0400, Nabeel Jafferali wrote:
> I have a customer that's located in France and he wants french phones
> if possible. So I'm wondering if there's any one out there that found
> a phone that can be change to french.
I believe snom phones have the option.
Confirmed

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RE: [Asterisk-Users] VoIP hackers gut Caller ID

[Alex]

Here is what you can possibly do:
- Steal calling cards if they are useing caller id authentication
scheme
- Get access to personal banking information (Citibank uses callerid
as part of authentication process.)
- Purchase goods and services backed up by calling verification.

I can go on and on for hours. Main point of story that [EMAIL PROTECTED] will hit the 
fan
and VOIP will be regulated badly. Especially if some known terrorist will
confess about using Vonage in Afaganistan.or some of drug dealers/weapon
traders will be cought .

Bug generraly author of that article is an idiot. He does not understand the
difference beteween VOIP and ISDN PRI. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of listas iPfone
Sent: Wednesday, July 07, 2004 6:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID

This is very interesting...

Regulations..USA...

But... what can i do faking a caller id? stolen what? what is the point? 

miklos

- Original Message - 
From: "Steve Totaro" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 07, 2004 12:56 PM
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID


> why regulate?  nobody regulates the return address on a letter sent via
> USPS.
> 
> 
> - Original Message - 
> From: "Kevin Walsh" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 07, 2004 10:00 AM
> Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID
> 
> 
> > Adam Hart [EMAIL PROTECTED] wrote:
> > > Chris Foster wrote:
> > > > The Register is carrying a article written by Kevin Poulsen of
> > > > Securtiy Focus, calling asterisk  "..the most powerful tool for
> > > > manipulating and accessing CPN data.."
> > > >
> > > > I hope NuFone doesn't drop asterisk-set-able callerid's after this
> > > > article; i've been wanting that feature from voicepluse for a long
> > > > time.
> > > >
> > > These kind of things will be reason (excuse) for Voip to be regulated
> > >
> > Perhaps service providers who allow the Caller*ID to be set should
> > insist that customers provide evidence that they own the phone numbers
> > that they want to publish, and then limit the customers' choices to
> > only the numbers in their approved list.  Calling the customer on the
> > provided number(s) would be an easy way to check, and a setup fee
> > could be levied to cover the provider's time and expenses, if required.
> >
> > Being able to discover a "blocked" Caller*ID is another matter.  Both
> > are good areas for regulation.
> >
> > -- 
> >_/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
> >   _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
> >  _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
> > _/   _/  _/_/_/_/  _/_/_/_/  _/_/
> >
> > ___
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RE: [Asterisk-Users] VoIP hackers gut Caller ID

[Alex]

Hi Guys,

This topic has become pretty much pointless. CallerID was never designed to
be any kind of authentication scheme. Also, it is very hard for telco to
restrict proper usage of CallerID in PRI or SS7 (Please consider number
protability, etc.)

We all already agreed on fact that author of this article are moron.

Let's not discuss any ideas of making CallerID secure or ajusting IAX to
carry 2 or 3 CallerID records. All of this is pointless.

If someone conducts business based on CallerId, it's up to them. If somebody
comits crime with fake CallerID, it's also fine. People, this world is not
perfect. There are thousands of telco companies where you will be able to
find somebody who does not enforce proper CallerID. There are bunch of
"telephony guys" who can do a lot of stuff, which you can't even think about
it.

But people, please do not write articles like that and do not publish it on
MSNBC, NY Times and CNN.

Thanks,



Aleksandr Palatkevich
BPVN Technologies Inc.
http://www.pipeboost.com/
Phone: (917) 723-0306
Fax: (212) 937-2170


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Bougues
Sent: Saturday, July 10, 2004 7:34 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID

On Wed, Jul 07, 2004 at 11:57:31AM -0400, Timothy R. McKee wrote:
> This has always been one of my pet peeves, even as I worked in the
industry.
> A telco switch operating a DS1 on trunk side should enforce caller-id
> numbers to be within the range of DID numbers assigned to that trunk.
There
> should be a default DID number that is used to replace any *invalid*
numbers
> sent on that trunk.  Note that blocked caller ids would still be blocked,
> but the rest of the data should be corrected.  Blocking ID is ok, lying
> about it is not.
> 
> Blind trust of a non-SS7 link is a _bad_ thing. 
> 

PRI signalling enables "Network provided" or "User provided"
caller-id. Maybe IAX could implement such a thing.

It's very common in France (at least) :
- the network will provided a guaranteed caller-id
- the user (CPE) may provide another one (usually, a DID number)

and the called party gets both. Unfortunatly, as far as I understand,
Asterisk is not really designed to handle more than one caller id
number.

-- 
Nicolas Bougues
Axialys Interactive
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RE: [Asterisk-Users] Bounty! For help with echo cancellation code.

[Alex]

Please post it at http://www.voip-info.org/wiki-Asterisk+Bounty
I'm ready to cover some of the costs.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, July 13, 2004 10:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Bounty! For help with echo cancellation code.


>From the CLI and during a call I want to be able to:

  *** Pulse the outgoing line and record at least 50 ms of the incoming
line.

  The pulse waveform must be specifiable as a series of amplitudes
  for each 1/8000 sec time slot.  It would be best of these values
  could be read from a file specified on the CLI command line.

  Timing should be synced between the pulse and the echo so that the
  delay from the pulse to the echo can be accurately determined.

  Echo cancellation should be disabled during this operation.

  This would operate similar to the echo-training code that operates
  at the initiation of a call except that this could be done at
  any time.

  The initial pulse and any echoes can be combined and saved in a
  single channel.

  Output should go to a file and should be in a simple format that
  a program such as Audacity can read, display and play. 
   

  *** Pulse the outgoing line and record at least 50 ms of the incoming
line.

  Same as above EXCEPT echo cancellation would not be disabled during
  this test and the results of the echo cancellation operations should
  be recorded and saved in a separate channel.
  

  *** Change variables used to control echo cancellation.

  Only the code in mec2.h is of interest.
   
  I will help identify the variables and modify the mec2.h code as
  needed to accomplish this goal.

  There are a lot of parameters in mec2.h that may affect the quality
  of the echo cancellation.  I want to be able to adjust them 'on the
  fly' and be able to immediately hear the results.


I am open to alternative proposals which would accomplish the same goals.

Name your price.


[EMAIL PROTECTED]
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RE: [Asterisk-Users] Current echo status?

[Alex]

Very interesting that our $11 ebay knock off has no echo issues what so ever
and original X100p produces fair amount of echo echo echo.

Also we have about 200 POTS lines connected to T410E via large channel bank,
were we trying to fight echo for about 10 days now now now.

I'm already thinking to get one of external echo cancellers to kill that
echo produced by analog side side side.

Thank you you you.

P.S. Who said echo echo echo ? :P

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Thursday, July 15, 2004 7:33 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Current echo status?

Scott Laird wrote:

> I've been following the list for months, and I have a working Asterisk 
> setup, but it'd be *really* useful to me at this point if someone 
> could summarize when Asterisk has echo problems and when it doesn't.  
> For instance, I usually hear a far-end echo when talking on my 7940, 
> but not when using a POTS phone plugged into a TDM400 FXO port.  It 
> doesn't seem to matter if the call goes out over a POTS line or via 
> NuFone; either way there's a fair bit of echo on most (but not all) 
> calls.
>
> Do different SIP phones have better echo cancellation then the Cisco 
> 7940/60s?  How about the Polycom IP500/600s?
>
> Does it matter if calls go out via POTS/T-1/PRI/VoIP?
>
> The general impression that I've received is that "fast" channels 
> (basically traditionally telephony interfaces) don't exhibit 
> noticeable echo, but the slight delay associated with VoIP 
> packetization unmasks existing echo.  Is that a reasonable summary?
>
> We're starting to plan for a new office build-out at work, and I'd 
> love to use Asterisk and SIP phones in the new office, but I'm not 
> going to try to sell management on a phone system with a horrible echo 
> problem, even if it will get fixed eventually.

I have an Asterisk system I have been testing for about a year for work 
and I have NO echo problems.

I just setup an * server at home with a generic $11 ebay X100p clone at 
home and had terrable echo when dialing out POTS. I was able to reduce 
it by changing the TXgain and echocancel set to on. But its still not 
perfect like my REAL DIGIUM card in the office PC. I can live with how 
my home phone is but it would not be good for my office system. Thats 
what I get for trying to go cheap. I would definately recommend Digium 
hardware, or at least a major brand supported by * no clones.

Kyle
www.quadrasoftware.com
Asterisk Receptionist, CallerID and applications.
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[Asterisk-Users] New Beta version of Grandstream Firmware 1.0.5.9

[Alex]

It gets definitely better every day.

List of bug fixes follows:

Release 1.0.5.9  7/26/2004  
If SIPRegister doesn't proceed due to conditions unmet, release
channel resource 
Fix the LED flashing issue when connection to the SIP proxy is lost.

Fix the issue where the device will not resume registration when it
loses connection to the outbound proxy for some time. 
Fixed the registration interval overflow issue 
Fixed the no-host-name in REGISTER message when configured using a
customer's Perl script 
Fixed the bad To header in INVITE after receiving 302 response 
Fixed the wrong URI in ACK to non-2xx response 
Fixed the issue where 486 would lose registration when outbound
proxy is configured and when NAT traversal is turned ON with STUN server
field blank. 

Release 1.0.5.8 7/16/2004  
Fix the branch ID uniqueness issue of ACK to a 2xx response
Fix the CSeq not incrementing issue associated with sending 
DTMF via SIP INFO when user is dialing fast and response to 
SIP INFO is not received fast enough
Fix the bad To header field in our new INVITE request upon 
receiving 302 response. 
Fix the issue that we do not respond to SIP INFO request. 
Do not play dial tone if registration is required and device 
is not registered. 
fix the inaccuracy of the timer unit value that causes 
registration to expire about 2% faster than normal 
fix the bug in parsing expire parameter and port when multiple 
contact items are on the same line (in a same header field) 
separated by comma. 
Fix some accidental issues that break call forwarding and 
call transfer

Release 1.0.5.7 7/8/2004

Fix the issue where we only send ACK only once which causes 
signaling failure if this ACK is not delivered (due to packet 
loss, etc) to the callee. 
Enable the high-pass filter and post-filter of G723. 
Remove the unnecessary dial tone when a user presses *xx when local 
call features are enabled 
If a symmetric NAT is detected, still use mapped IP:port instead of 
using private IP. 
Allow access to 486's Web server using the WAN side IP from LAN port
Send ACK to the server in stead of per Contact header upon receiving

3xx response to an INVITE. 



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RE: [Asterisk-Users] Re: Grandstream Early Dial

[Alex]

This problem with Grandstream was fixed long ago by Mark.
You have to change your sip entry from friend to peer and enable option
insecure=very to make early dialing working. 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Besch
Sent: Tuesday, June 08, 2004 10:09 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Grandstream Early Dial

Aaron Martin wrote:
> 
> Has anyone managed to get Early-Dial working with the grandstream phones?
>  
> On my older phones running firmware 1.0.3.X it works fine, but it doesnt 
> work on the newer versions..
>  
>  

Don't waste time trying. I'm even surprised that you could get it to 
work with 1.0.3.x. In my experience, if the dial string was longer than 
4 or 5 digits, GS pooped out. It's even worse in the later revisions, 
pooping out after 3 digits. It still is not fixed in 1.0.5.0. You could 
use it if all of your dial strings are 3 0r fewer digits - not much use 
really.  GS knows about the problem, has verified on my * server, has 
indicated that they will fix it. It just requires patience on our part. 
The fix is apparently not a very high priority - true really, since it's 
failure is merely a convenience issue of not having to wait for the last 
key timeout.

Stephen R. Besch
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[Asterisk-Users] PSTN tones with ISDN4Linux

[Alex]

Hi all,

I'm playing with Asterisk and I've already configured all needed .conf
files.
It works quite well, but now I need your help to tune the system: when I
place a call from a softphone to the PSTN, I can't hear directly Telco's
tones and I can't use its services, e.g. a mobile's answering machine.
I don't know if I have to modify the dialplan or if it depends on my
configuration: what is wrong?
Any help will be appreciated!

TIA,

Alex


modem.conf

[interfaces]
context=pstn-in
driver=i4l  ; isdn4linux - an alternative to i4l is to use chan_capi
language=it
type=autodetect
stripmsd=0
dialtype=tone
mode=immediate
group=1
msn=***REMOVED***
device => /dev/ttyI0
device => /dev/ttyI1


sip.conf

[general]
port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = sip-remote-in ; Send unknown SIP callers to this context
callerid = Unknown

[201] ; X-Lite client 201
type=friend
secret=***REMOVED***
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
callerid="***REMOVED***" <201>
disallow=all
allow=gsm
context=sip-in


extensions.conf

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Modem/g1  ; Trunk interface
TRUNKMSD=1  ; MSD digits to strip
(usually 1 or 0)

; ##
; Macros
; ##

[macro-dialout]
exten => s,1,Dial(${TRUNK}:${ARG1}) ; Ring the interface
exten => s,2,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Playtones(congestion)
exten => s-BUSY,1,Playtones(busy)
exten => s-.,1,Goto(s-NOANSWER,1)   ; Treat anything else as no
answer


; ##
; Outbound Contexts
; ##

[pstn-mobiles-out]
exten => _3.,1,Macro(dialout,${EXTEN})  ; All mobiles start with 3

; other pstn-related contexts were cut

[pstn-out]
include => pstn-locals-out
include => pstn-longdistance-out
include => pstn-mobiles-out


; ##
; Inbound Contexts
; ##

[sip-remote-in]
exten => s,1,Hungup

[sip-in]
include => pstn-out
; cut

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[Asterisk-Users] HP Proliant ML110 with Adaptec 2610SA and Debian

[Alex]

Hi guys,
I am trying to install Debian sarge (latest netinstall) on ML110 server  
with two SATA hardware mirrored drives on Adaptec 2610SA controller for  
use with Asterisk with no luck.

Debian installer does not see the array. Any workarounds?
Please help.
Regards,
Alex.
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[Asterisk-Users] Some asterisk ser problems

[Alex]

I have some simple questions and i need your help guys.
 
I have ser server which working fine, between users.
I am trying to add some more features to the ser. Most important is the IVR.
 
I installed Asterisk and i am trying to register user in asterisk with no success.
Part of ser.cfg file where i am trying to redirect the call to the asterisk.
-
if (method == "INVITE") {     if (uri =~ "sip:[EMAIL PROTECTED]"){     log(1, "Forwarding to Asterisk\n");     rewritehostport("xx.xx.xx.xx:");     t_relay();     break;     }  } 
---
 
 
inside sip.conf i have
-
register => sipphonenumber:[EMAIL PROTECTED]/
 
 
error
-
chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to ';tag=as12200854'
 
I need some help with configuring asterisk to work with ser.
Let's say i am calling from sip phone to number 12345 , i would like to enter into IVR system where i can configure which number to press, what kind of music to play etc.
 
The main goal is to create IVR system for identical phone number for SIP users.
 
Thanks for any help.
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Re: [Asterisk-Users] Some asterisk ser problems

[Alex]

In ser.cfg
--if (method == "INVITE") {     if (uri =~ "sip:[EMAIL PROTECTED]"){     log(1, "Forwarding to Asterisk\n");     rewritehostport("xxx.xxx.xxx.xxx:5061");     t_relay();     break;     }  }   
 
 
In  sip.conf
---
[ser]type=friendhost=xxx.xxx.xxx.xxxcontext=from-ser
 
In extension.conf

[from-ser]exten => _1,1,Dial(SIP/[EMAIL PROTECTED],20,r)
 
Sip Debug from Asterisk
--
 
Sip read: INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKe189.20ca26d7.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK45dda3d0From: "Alex" ;tag=00036b09607e003b16a3f758-1d78797aTo: Call-ID: 00036b09-607e003b-552c14b9-021cab1d@xxx.xxx.xxx.xxx
CSeq: 101 INVITEUser-Agent: CSCO/6Contact: Expires: 180Content-Type: application/sdpContent-Length: 248Accept: application/sdp
v=0o=Cisco-SIPUA 7329 20490 IN IP4 numbercallingfroms=SIP Callc=IN IP4 numbercallingfromt=0 0m=audio 26274 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15
13 headers, 11 linesUsing latest request as basis requestSending to xxx.xxx.xxx.xxx : 5060 (non-NAT)Found peer 'ser'Found RTP audio format 0Found RTP audio format 8Found RTP audio format 18Found RTP audio format 101Peer audio RTP is at port numbercallingfrom:26274Found description format PCMUFound description format PCMAFound description format G729Found description format telephone-eventCapabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)Looking for 1x in from-serReliably Transmitting (no NAT):SIP/2.0 484 Address IncompleteVia: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKe189.20ca26d7.0Via: SIP/2.0/UDP numbercallingfrom:5060;branch=z9hG4bK45dda3d0From: "Alexg"
 ;tag=00036b09607e003b16a3f758-1d78797aTo: ;tag=as6a19e3f4Call-ID: [EMAIL PROTECTED]CSeq: 101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: Content-Length: 0
 to xxx.xxx.xxx.xxx:5060
Sip read: ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKe189.20ca26d7.0From: "Alexg" ;tag=00036b09607e003b16a3f758-1d78797aCall-ID: [EMAIL PROTECTED]To: ;tag=as6a19e3f4CSeq: 101 ACKUser-Agent: Sip EXpress router(0.8.14 (i386/linux))Content-Length: 0
8 headers, 0 linesDestroying call '[EMAIL PROTECTED]'
 
After call i hear busyvoice on the line. I have to configure it to use some IVR system in order to be abble to choose numbers (extensions) and depend on the extension to play some kind of music .
 
The help is more than welcome.
Thanks.
 
 
Alistair Cunningham <[EMAIL PROTECTED]> wrote:
Alex,If you are forwarding calls in SER based on URI patterns rather than the location database, you don't need to register Asterisk with SER. Instead of the register line, you should have a peer for SER; something like this:[ser]type = friendhost = context = There are lots more options for the peer, but this should get you started.If you'd like more detailed support, my company, Integrics Ltd, does support for both Asterisk and SER. We can also write the IVRs for you.Alistair Cunningham,Integrics Ltd,Telephony, Database, Unix consulting worldwide+44 (0)7870 699 479http://integrics.com/Alex wrote:> I have some simple questions and i need your help guys.> > I have ser server which working fine, between users.
 >
 I am trying to add some more features to the ser. Most important is the IVR.> > I installed Asterisk and i am trying to register user in asterisk with > no success.> Part of ser.cfg file where i am trying to redirect the call to the asterisk.> -> if (method == "INVITE") {> if (uri =~ "sip:[EMAIL PROTECTED]"){> log(1, "Forwarding to Asterisk\n");> rewritehostport("xx.xx.xx.xx:");> t_relay();> break;> }> }> ---> > > inside sip.conf i have> -> register => sipphonenumber:[EMAIL PROTECTED]/> > > error>
 -> chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to >

[Asterisk-Users] IVR setup problems

[Alex]

Hi guys still have the problem to setup the IVR correctly.
 
I am forwarding call from ser :
if (method == "INVITE") {     if (uri =~ "sip:[EMAIL PROTECTED]"){     log(1, "Forwarding to Asterisk\n");     rewritehostport("xxx.xxx.xxx.xxx:5061");     t_relay();     break;     }  } 
 
inside sip.conf
-
port=5061 bindaddr=0.0.0.0   srvlookup=yes 
 
[ser]type=peerhost=xxx.xxx.xxx.xxxcontext=ser1
 
inside extensions.conf
-
[ser1]Exten => 40,1,AnswerExten => 40,2,SetMusicOnHold(default)Exten => 40,3,DigitTimeout,5Exten => 40,4,ResponseTimeout,10Exten => 40,5,Background(greeting)
Exten => 1,1,Playback(secr) ; if you press <91>1<92> playback message <93>secr<94>Exten => 1,2,Dial(SIP/Phone1/20)
Exten => 2,1,Playback(studentservice)Exten => 2,2,Dial(SIP/Phone1/20)
Exten => 3,1,Playback(it)Exten => 3,2,Dial(SIP/Phone1/20)
Exten => 4,1,Playback(operator)Exten => 4,2,Dial(SIP/Phone1/20)
 
 
Inside asterisk debug i see what the forwarding of the call working :
log of ASTERISK DEBUG

Sip read: INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0Via: SIP/2.0/UDP ipoftphone:5060;branch=z9hG4bK06ffef7dFrom: "Alexg" ;tag=00036b09607e0047524bda98-4b96b81eTo: Call-ID: [EMAIL PROTECTED]CSeq: 101 INVITEUser-Agent: CSCO/6Contact: Expires: 180Content-Type: application/sdpContent-Length: 249Accept: application/sdp
v=0o=Cisco-SIPUA 28416 11732 IN IP4 ipoftphones=SIP Callc=IN IP4 ipoftphonet=0 0m=audio 26298 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15
13 headers, 11 linesUsing latest request as basis requestSending to xxx.xxx.xxx.xxx : 5060 (non-NAT)Found peer 'ser'Found RTP audio format 0Found RTP audio format 8Found RTP audio format 18Found RTP audio format 101Peer audio RTP is at port ipoftphone:26298Found description format PCMUFound description format PCMAFound description format G729Found description format telephone-eventCapabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)Looking for 1phoneiamcalling in ser1Reliably Transmitting (no NAT):SIP/2.0 404 Not FoundVia: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0Via: SIP/2.0/UDP ipoftphone:5060;branch=z9hG4bK06ffef7dFrom: "Alexg" ;tag=00036b09607e0047524bda98-4b96b81eTo:
 ;tag=as125ae8d3Call-ID: [EMAIL PROTECTED]CSeq: 101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: Content-Length: 0
 to xxx.xxx.xxx.xxx:5060
Sip read: ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0From: "Alexg" ;tag=00036b09607e0047524bda98-4b96b81eCall-ID: [EMAIL PROTECTED]To: ;tag=as125ae8d3CSeq: 101 ACKUser-Agent: Sip EXpress router(0.8.14 (i386/linux))Content-Length: 0
8 headers, 0 linesDestroying call '[EMAIL PROTECTED]'
 
 
var/log/asterisk/messages
---
Unable to open /dev/dsp: No such device
 
 
I am calling to number 122 and ser forwarding it to the asterisk (port 5061) (see configuration of sip.conf) to the ser1 context.
in extensions.conf i have ser1 context and extensions for ivr under ser1 context.
After the call i am hearing the busy line and that's it. i tried to play with extensions.conf with no success.
I need a help to setup the IVR system.
 
Thanks.
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[Asterisk-Users] Forward Call from Asterisk to SER

[Alex]

I have some problem to redirect the call from asterisk to ser.
1 thing i am redirecting call to asterisk and then on some extension i want to return the call to ser.
 
Receiving this error:
 
WARNING[23594]: chan_sip.c:6829 handle_response: Forbidden - wrong password on authentication for INVITE to  '"Alex" ;tag=as55a3adbb'    -- SIP/212.25.75.195:5060-3bc0 is circuit-busy
 
Any help will be appreciate. 
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[Asterisk-Users] Some errors on sip debug

[Alex]

I have some problem to configure the call from asterisk to ser.[globals]SERADDRESS=xxx.xxx.xxx.xxx:5060  exten => 77,1,Dial(SIP/[EMAIL PROTECTED],20,r)  Error in Sip Debug ---NOTICE[25541]: chan_sip.c:6848 handle_response: Failed to authenticate on INVITE to '"Alexg" ;tag=as3cf27769'  Any help will be appreciated.Thanks__Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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[Asterisk-Users] Bristuff e RealTime: STABLE vs. CVS-HEAD

[Alex]

Hi all!

Was anybody able to install kapejod's zaphfc drivers together with RealTime
application? I'm in big trouble because bristuff relay on STABLE version,
while RealTime is included in the CVS-HEAD.
I found this hint, "Installing zaphfc with CVS-Head" at
http://voip-info.org/wiki-Asterisk+zaphfc+install, but it was written many
months ago: may it be still useful?

TIA,

Alex

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RE: [Asterisk-Users] Bristuff e RealTime: STABLE vs. CVS-HEAD

[Alex]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Florian Overkamp
> Sent: Friday, March 04, 2005 11:21 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Bristuff e RealTime: STABLE vs. CVS-HEAD
> 
> Hi,
> 
> Very doubtfull. In the mean time there have been a number of very radical
> changes to CVS-HEAD, which are not available in STABLE, therefore, not
> compatible with BRIstuff. You would be better off trying to backport
> RealTime into STABLE, I think...
> 
> Florian

Hi,

thanks for your replay!

Backporting RealTime into STABLE version sounds quite difficult: I'm not so
skilled in Linux, but I could try. I've no idea about where to start. Do you
have any link to suggest me, please?

To have RealTime working, what about using chan_capi instead of bristuff? I
read that chan_capi supports latest CVS-HEAD, but it is not completely clear
to me whether it supports HFC based cards or not.

Thanks again,

Alex

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Re: [Asterisk-Users] Hardphone deployment recommendation

[alex]

On Monday 07 March 2005 10:26, Paul Dugas wrote:

> A more uncomfortable issue is that the speaker phones were found to be
> working very poorly.  The speakerphone user is just about inaudable to the
> user on the other end of the call.  This is the case with all of the units
> I have.  I had them all running the lates firmware from the website.  In
> Sipura's defense, they responded within about 15 minutes to my support
> email with a link to a "test" version of the firmware which improved
> things but didn't completely fix it.

The microphone is [somewhat inexplicably] mounted in the base over a hole that 
faces downwards, between two of the rubber feet, its like a 'U' with a dot 
inside it. If you're feeling brave, you could bore a new hole and mount it 
somewhere its more likely to pick up the voice of the user. Or you could try 
enlarging the hole with the tip of a knife and see if that helps. I fitted a 
few LEDs inside pointing in at the screen because of the lack of a backlight. 
Didn't make much difference.

alexd
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[Asterisk-Users] Some audio problems

[Alex]

Hi all.
 
I have a problem to hear one side, when the second is working fine.
 
softphone  ->  ser -> asterisk (IVR) -> extension in IVR -> ser -> pstn -> regular phone.
 
The audio which coming from regular phone i can hear without problem, but the audio from softphone i can not hear in the regular phone.
 
here is the log what i am receiving:
 
9 headers, 9 linesFound RTP audio format 8Found RTP audio format 101Peer audio RTP is at port xxx.xxx.xxx.xxx:27232Found description format telephone-eventCapabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)set_destination: Parsing  for address/port to send toset_destination: set destination to serserverip, port 5060
 
 
inside sip.conf 
 
disallow=all   allow=ulawallow=alaw
 
now my soft phone using G729,G723,alaw
 
Any help will be more than appreciated. 
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[Asterisk-Users] Need some help

[Alex]

Hi all
 
I have a couple of questions maybe you guys can help me with them
 
I have sip phones ,  SER server , Asterisk.
 
what is the best way to do that (also with accounting and authentication).
 
which one of those options
1)  sipphone -> SER -> ASTERISK -> SER -> PSTN
 
2)  sipphone -> SER ->ASTERISK ->PSTN
 
on the first option i am trying to return the call to the ser after it's pass the asterisk for some routing solutions and accounting. but i have some problems to hear the other side.
 
 
Thanks for any advice 
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Re: [Asterisk-Users] Need some help

[Alex]

The reason i am using 1 scenario is because of routing, authentication and accounting. If i can use these things in asterisk i will use it.
 
1)What is the best way to do that, through extensions ?
 
2)When the call coming i can check the phone number and if the number should go to pstn i will forward this to asterisk. now i need example how i can forward the call to pstn in extension.cfg (or there are other way to forward the call to pstn inside asterisk).
 
3) Another question if all my users authenticated with ser how i can send call to the other user who included in the same database and connected to the same SER server (LOCAL CALLS) should not go to the pstn. 
Should i use softphone(user1) ->SER->ASTERISK->SER->softphone(user2)
 or i need to use softphone(user1)->SER-(user2) if i use this scenario i will loose the cdrs of the asterisk.
 
The reason i am asking the these question because till now i didn't use asterisk and i forwarded the call through ser and it's working fine. I wanted to use IVR system so i installed the asterisk and also asterisk has the CDRs. now i need to use this scenario 
long distance call:   softphone -> SER -> Asterisk -> pstn (long distance calls)
local calls: softphone->SER ->Asterisk -> SER->softphone ( I am not sure if i can do that without registering users inside sip.cfg in the asterisk.)
 
Any help will be appreciated.
 Yair Hakak <[EMAIL PROTECTED]> wrote:
Duh, i'm an idiot. I meant scenario #1.-yairOn Wed, 23 Mar 2005 18:52:28 +0200, Yair Hakak <[EMAIL PROTECTED]>wrote:> Hello,> what is the benefit of your scenario #2? I'm not understanding what> it adds for you...> > -yair> > > On Wed, 23 Mar 2005 08:49:37 -0800 (PST), Alex <[EMAIL PROTECTED]>wrote:> > Hi all> >> > I have a couple of questions maybe you guys can help me with them> >> > I have sip phones , SER server , Asterisk.> >> > what is the best way to do that (also with accounting and authentication).> >> > which one of those options> > 1) sipphone -> SER -> ASTERISK -> SER -> PSTN> >> > 2) sipphone -> SER ->ASTERISK ->PSTN> >> > on 
 the
 first option i am trying to return the call to the ser after it's> > pass the asterisk for some routing solutions and accounting. but i have some> > problems to hear the other side.> >> >> > Thanks for any advice> >> > > > Do you Yahoo!?> > Yahoo! Small Business - Try our new resources site!> >> >> > ___> > Asterisk-Users mailing list> > Asterisk-Users@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> >> >>___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users<
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[Asterisk-Users] Call-ID and Unique-ID

[Alex]

Could anyone explain to me what is the difference between Call-ID and
UniqueID of SIP calls, please?
Which one could be used as an ID to trace, for example, the status of a call
with Manager API and PHP?

Thanks,

Alex

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RE: [Asterisk-Users] Call-ID and Unique-ID

[Alex]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
> Sent: Tuesday, March 29, 2005 4:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Call-ID and Unique-ID
> 
> The Call-ID is internal to the SIP protocol, and not exposed inside
> Asterisk (or via manager/AGI). The UniqueID is assigned by Asterisk to
> the call itself and should be used for tracking the call via the
> Asterisk interfaces.

Thank you very much!

Alex

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Re: [Asterisk-Users] No prompt after installing

[Alex]

On Wed, 30 Mar 2005 21:43:49 -0600, Kristian Kielhofner <[EMAIL PROTECTED]>  
wrote:

Anton Krall wrote:
Guys.
 I just finished installing a new asterisk box and here comes the first
problem.
 The box doesnt have zaptel cards or anything, its a plain RH9 with  
asterisk.
 Every compiled perfectly and when trying to run asterisk -vg
 I get all the messages shown below, no errors except for a "
res_musiconhold.c:484 monmp3thread: Request to schedule in the past?!?!"
 Whats this?  Also, after everything loads... I get the "Asterisk  
ready." message but.. No
prompt... It just hangs there until I hit ctrl-C

Check system time
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[Asterisk-Users] mISDN + chan_misdn and DTMF

[Alex]

Asterisk CVS-HEAD 30/3/05 + mISDN + chan_misdn = everything works fine but
outgoing DTMFs are not sent to the called party when placing calls from SIP
clients to PSTN (ISDN). On the contrary, incoming calls from PSTN can send
DTMF to Asterisk.
I've already tried every combinations of dtmfmode (inband, rfc2833, info)
either in sip.conf, either in clients options, with all codecs. I've also
tried setting SIPDtmfMode() before dialing, but had no success.

Any hint would be greatly appreciated!

TIA,

Alex

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[Asterisk-Users] compilation of asterisk

[Alex]

Hi guys 
Trying to compile asterisk and i am receiving this errror.
 
gcc -g  -o asterisk -Wl,-E  io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o  editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv   -lssl
 
/usr/bin/ld: cannot find -lsslcollect2: ld returned 1 exit statusmake: *** [asterisk] Error 1
 
any help will be appreciated.
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[Asterisk-Users] SIP -> PSTN: mISDN DTMF tones generation

[Alex]

How can I send DTMF tones on outgoing calls to PSTN from SIP
clients when using chan_misdn (release beta-0.1.0, 04/1/2005) and HFC-S0 based cards
without loosing compatibility with the Asterisk voicemail system, please? Is
there any option to set into misdn.conf to have DTMF tones being generated?

 

TIA,

 

Alex






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RE: [Asterisk-Users] Re: mISDN + chan_misdn and DTMF

[Alex]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Stefan Tichy
> Sent: Tuesday, April 05, 2005 11:19 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Re: mISDN + chan_misdn and DTMF
> 
> On Mon, Apr 04, 2005 at 07:46:49PM +0200, Alex wrote:
> > Asterisk CVS-HEAD 30/3/05 + mISDN + chan_misdn = everything works fine
> but
> > outgoing DTMFs are not sent to the called party when placing calls from
> SIP
> > clients to PSTN (ISDN). On the contrary, incoming calls from PSTN can
> send
> > DTMF to Asterisk.
> > I've already tried every combinations of dtmfmode (inband, rfc2833,
> info)
> > either in sip.conf, either in clients options, with all codecs. I've
> also
> > tried setting SIPDtmfMode() before dialing, but had no success.
> 
> "Turn on inband DTMF on your SIP device, this works."
> (copied from http://www.beronet.com/bugs/  bug_id 10 )
> 
> Is the problem specific to CVS-HEAD 30/3/05 ? If you think it is a
> chan_misdn problem, please use beronets bug tracker.

I didn't know if it was specific to beronet's chan_misdn or it was a general
issue, and that's why I asked here on the list before saying that's was a
bug for sure.

However, I've just had a look to your link (I had already looked at
beronet's buglist, but because of a filter applied on search results I
didn't noticed that bug :( ) and you're right, that's a chan_misdn
"feature".

Thanks,

Alex

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Re: [Asterisk-Users] MAX TNT SIP / Asterisk

[alex]

On Tue, 2 Nov 2004, James Taylor wrote:

> I can't get my MAX TNT to register with Asterisk.
> TAOS 11.0.
> 
> SIP phone registeration show up in Asterisk like this:
>   and works.
> 
> The TNT shows up as:
>  .
> 
> Does anyone have this working?
> Am I missing something here?
> Where does the TNT get it's user name?  Or, can it work without one?
It works without one.

Why do you need to register TNT to asterisk anyway?

--alex

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Re: [Asterisk-Users] MAX TNT

[alex]

On Sun, 7 Nov 2004, voip wrote:

> Any body using Asterisk with a MAX TNT?
> 
> SIP or H.323?
asterisk + ser + TNT work fine.

ser is proxy server, asterisk is feature server.

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[Asterisk-Users] register => 2345:[EMAIL PROTECTED] doesn't care about port

[Alex]

Hi,


 example:

> register => 531:[EMAIL PROTECTED]:5061/1234

Unfortunately this doesn't really fit my needs.
"/1234" means [EMAIL PROTECTED], where default is the context specified in 
the general section of sip.conf:


[general]
context=default ; Default context for incoming calls

Since I have several sip_proxy, I need different incoming contexts.
I mean I want to send calls from sip_proxy _ONE to context 
"from-proxy_ONE" and calls from sip_proxy_TWO to context "from-proxy_TWO".

So I need to use:

register => 2345:[EMAIL PROTECTED]
register => 6789:[EMAIL PROTECTED]

where

[sip_proxy_ONE]
type=peer
context=from-proxy_ONE <-- this is important for me!!
host=sip.proxy_ONE.com
port=5061

[sip_proxy_TWO]
type=peer
context=from-proxy_TWO <-- this is important for me!!
host=sip.proxy_TWO.com
port=9000

This way, port is never used.

Thanks,
Alex
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[Asterisk-Users] register => 2345:[EMAIL PROTECTED] doesn't care about port

[Alex]

Sorry for bumping this up ( 
http://lists.digium.com/pipermail/asterisk-users/2006-February/148059.html )

Any ideas, please?

TIA,
Alex
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[Asterisk-Users] Have some latency problems.

[Alex]

Hi guys i have some problems with asterisk latency.
I am trying to play online radio streaming on musiconhold and i am receiving bad quality of the sound and the latency, then i am calling from ip phone there is no problem but when i am calling from regular phone to my Asterisk server i receving this stuff:
 
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
RFC3389: 1 bytes, level 4...
 
maybe the problem in the codec or if there any way to reduce the size of the packets.
 
Any help will be appreciated.
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Re: [Asterisk-Users] Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone

[Alex]

what is the upload speed on B?

Looks to me as you have bandwidth problem!

Martin Kronstad wrote:

Hi!

 


Problem:

 

I can’t hear what the people at Location B i saying, they hear me but I 
do not hear them. They can call, I can call. Just no sound.


 


My current setup is:

 

Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> 
Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B)


 


I am having problems with sound, I have opened the following ports:

 


Location A:

10 000 -> 20 000 (TCP and UDP)

5060  (TCP and UDP)

8000  (TCP and UDP)

 


Location B:

8000  (TCP and UDP)

5060  (TCP and UDP)

 


I am using [EMAIL PROTECTED] 1.3 , and xlite as softphone.

 


I have tried to set the softphone

 


I have set the extention parameters(in sip.conf) to:

 


;; Location A

[200]

username=200

type=friend

secret=1234

record_out=On-Demand

record_in=On-Demand

qualify=no

port=5060

nat=never

[EMAIL PROTECTED]

host=dynamic

dtmfmode=rfc2833

context=from-internal

canreinvite=no

callerid="Location A" <200>

 


;; Location B

[201]

username=201

type=friend

secret=1234

record_out=On-Demand

record_in=On-Demand

qualify=no

port=5060

nat=yes

[EMAIL PROTECTED]

host=dynamic

dtmfmode=rfc2833

context=from-internal

canreinvite=no

callerid="Location B" <201>

 


My sip.conf :

 


port = 5060   ; Port to bind to (SIP is 5060)

bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)

externip=80.202.50.16

disallow=all

allow=ulaw

allow=alaw

context = from-sip-external ; Send unknown SIP callers to this context

callerid = Unknown

language=no

 

 


Best Regard Martin Kronstad




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[Asterisk-Users] Extension problems

[Alex]

Hi allI have a question:i am trying to make a dial plan with IVR with option to call some phone.exten => 3,1,Dial(SIP/"phonenumber"@xxx.xxx.xxx.xxx,,r)and i have the next problem :
INVITE sip:"phonenumber"@xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7f9a1d08..From: "Unknown" Unk
  nown@xxx.xxx.xxx.xxx>;tag=as75542381..To: ..Contact: Unknown@xxx.xxx.xxx.xxx>..Call-ID: 50e2f6c317
  [EMAIL PROTECTED]: 102 INVITE..User-Agent: Asterisk PBX..Date: Tue, 09 Aug 2005 05:47:10 GMT..Allow: I  NVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type: application/sdp..Content-Length: 218v=0..o=root 18529 18529 IN IP4 
xxx.xxx.xxx.xxx ..s=session..c=IN IP4 xxx.xxx.xxx.xxx..t=0 0..m=audio 12712 RTP/AVP 8 101..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telepho  ne-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..
 
"Unknown" in the "From" making the problem i am trying to change this value with i puting these 2 lines before the Dial with correct order.
 
exten => 212,5,SetCallerID(221222)
exten => 212,6,SetCIDNum(221222)
 
and no success.
 
any help will be appreciated
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Re: [Asterisk-Users] call "load balancing"

[Alex]

I am doing traffic shaping with a open source linux firewall 
http://www.ipcop.org/


and since i have traffic shaping configured my 3 VoIP lines work great.

I am not using Asterix yet but I will go to as soon as I have the time 
to work myself into it.


If anybody can tell me where the best information is to get a start on 
it, I would greatly appreciate it.


alex



Darren Wright wrote:

---
An ever better way is get some kind of SLA with guaranteed uptime and 
bandwith, a symetrical link, and do some traffic shaping to ensure that 
VoIP has priority. Part of the point of VoIP is to save money by 
collapsing voice and data networks onto one (presumably robust) network,


so having 2 shabby separate DSL connections kinds of defeats the
purpose.
--


How do you traffic shape incoming packets though  Without your ISP
to provide QoS for downstream voice traffic, quality can still be an
issue


-Darren


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Re: [Asterisk-Users] will a firewall slow down asterisk?

[Alex]

I recommend m0n0wall (http://m0n0.ch/wall/)  which is a NetBSD based
firewall that includes traffic shaping. Easily managed via a web
interface. Runs on any decent PC with 2 or more NICs. Also on Soekris
or WRAP embedded platforms.

I recommend IpCop www.ipcop.org
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[Asterisk-Users] "Catch all" extension

[Alex]

exten => _X.,1,AGI,catchall.agi,${EXTEN}


should do it for u



Hi,

since I also have some "applications" that starts with "*", like

[app-clir]
exten => _*67.,1,SetCallerPres(prohib)
exten => _*67.,2,Goto(${EXTEN:3},1)

I thought I could use "_." instead of "_X.", that would match only numbers.
However, what do you think about this single extension replacing the *whole* 
dialplan?
I mean that

exten => 200,1,Dial(SIP/200)
exten => 300,1,Dial(IAX/300)
exten => _0.,1,Macro(dialout,${EXTEN})

and so on would be replaced by that single extension, and then the script will 
do the rest.

Thanks,

Alex


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Re: [Asterisk-Users] IP Cop as a firewall and QOS

[Alex]

I can sign that immediately.

I am not using asterix yet, but I am having VoIP phone behind IpCop and 
never had a problem yet.


About the SOHO design you mention, the only limitation it has is that 
you can only have a single green network (internally subnet), but you 
can abuse the blue (designed for WLAN clients) and the orange(DMZ) for 
that too.


And I think for version 1.5 it is planned to have multiple green network 
possible.


But I think you should go to the IpCop users newsgroup and ask there if 
it suits your special needs and if somebody already has a config like yours.


Austin Denyer wrote:

On Wed, 2005-08-17 at 17:27 -0500, Mojo Jojo wrote:


I don't mind buying an appliance to get something solid but IP Cop just 
looks better than he appliances I see out there.


I am only concerned if it is stable for a production environment. It says 
it's designed for a SOHO environment, we are doing a bit more than that.


Will this thing hold up? Can it be trusted?



I'm not using IPCop with * (I'm very much a * newbie), but I am using it
as a general firewall, and it rocks.  


I have had no issues with it, and I have been running IPCop for several
years.

It is very stable - I have yet to have it crash on me.

It is secure - the box has yet to be successfully hacked (and the logs
show numerous attempts on a daily basis!)

It will handle your bandwidth easily as long as your hardware is not too
antiquated.  For example, I've got it running on a 133MHz Pentium, 128Mb
RAM on a 3MB/sec connection, and it hardly even notices...

Try it - you'll like it.

Regards,
Austin.

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[Asterisk-Users] Ethernet / TcpIp phones

[Alex]

Is there any VoIP phones available which can be plugged directly to the 
Ethernet network?


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[Asterisk-Users] Re: [Asterisk-biz] Modem Over IP: solutions ?

[alex]

On Sat, 22 Oct 2005, Jean-Michel Hiver wrote:

> I have a potential client who has legacy alarm systems which use modems
> to transmit encoded data to a remote location through the PSTN. They
> wish to replace the 'PSTN' bit with an IP link.
> 
> I am aware that it would be best if the data was transmitted directly
> over IP rather than modulated and then sent on the internet, but that is
> not possible because of the legacy equipment.
> 
> I was wondering if there was some specialized ATAs of some kind that
> would do TDMoIP and which could be used for this purpose?
> 
> Link latency is about 300ms with no more than 10ms jitter. If you have a
> solution please let me know!
No.

Terminate the connection on the remote side. Equipment such as Lucent MAX 
to do that is a dime a dozen now (4-port max6000 is ~200$)

-alex


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[Asterisk-Users] Call Transfer problems-am I missing something?

[alex]

Hi All,

Recently got call-transfer somewhat working on my asterisk-1.0.9 
install, and came across an interesting problem.  I have an account on a 
VOIP Provider (voipbuster using iax to be exact) and use a line like 
this in extensions.conf to have it handle all outgoing calls beginning 
with 1:
exten => _1NN,1,Dial(voipbuster/00${EXTEN},t)
When I call someone and press # on the phone ( I've tried this with 
various softphones and a regular phone connected to a linksys pap2) 
Nothing happens.However, if the called party presses # they get the 
extension prompt, and can then transfer me to an other extension.  Does 
anyone know why the calling party can't initiate the transfer? am I 
missing something?

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Re: [Asterisk-Users] Call Transfer problems-am I missing something?

[alex]

Hi,

Thanks for the clarification.  I had seen that the two options 
existed, but the docs for the dial() command didn't state the 
difference.
On Sun, Oct 30, 2005 at 08:23:32PM -0500, David Bandel wrote:
> On 10/30/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > Hi All,
> >
> > Recently got call-transfer somewhat working on my asterisk-1.0.9
> > install, and came across an interesting problem.  I have an account on a
> > VOIP Provider (voipbuster using iax to be exact) and use a line like
> > this in extensions.conf to have it handle all outgoing calls beginning
> > with 1:
> > exten => _1NN,1,Dial(voipbuster/00${EXTEN},t)
> > When I call someone and press # on the phone ( I've tried this with
> > various softphones and a regular phone connected to a linksys pap2)
> > Nothing happens.However, if the called party presses # they get the
> > extension prompt, and can then transfer me to an other extension.  Does
> > anyone know why the calling party can't initiate the transfer? am I
> > missing something?
> 
> Yes.  The ,t  in the Dial() options is for callee, the T is for
> caller.  ,tT is for both.
> 
> Ciao,
> 
> David A. Bandel
> --
> Focus on the dream, not the competition.
> - Nemesis Air Racing Team motto
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[Asterisk-Users] SJphone "Awaiting ACK" after updating Asterisk to CVS-HEAD of September

[Alex]

Hi,

 

sometimes I can’t answer calls with SJphone because of
an “Awaiting ACK” error.

The problem has come after I updated Asterisk from CVS HEAD
of August to HEAD of September. I had no other changes in my configuration, so
I think it must be related to something in Asterisk. FYI, Asterisk is now
updated to the latest CVS HEAD and the problem is still there.

Did anyone already have this kind of problem, please?

 

Full debug and tcpdump are available if needed.

 

Thanks,

 

Alex






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[asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI

[Alex]

Hi all!

We've released VoiceOne 0.4.0, a web-based and open source solution 
which allows to fully manage an Asterisk service hosted on a LAMP server.


We focused on an charming and overall user-friendly interface. Thanks to 
the authentication based on roles, once configured by a super user, the 
PBX may be easily maintained even by an Asterisk unskilled users.


From a technical point of view, the application is made up of two 
modules: one for the client - i.e. the user interface - and the other 
for the server. Thanks to the web services provided by the server module 
and the use of a database, VoiceOne may be easily integrated with other 
applications (e.g. CRM software).


The project has grown and has received positive response so far. 
Nowadays there's a little but enthusiastic community of developers, 
supporters and users. Translations in several languages (e.g. English, 
Spanish, Russian, etc.) are already available.


On the project website at http://www.voiceone.it you'll find the online 
demo and the links to download the source files from Sourceforge, as 
well as a support forum.


We would be pleased if you could give it a try and let us know your 
feedback, comments, ideas, or suggestions replying here or posting a 
message on our forum.


Thanks for your kind attention.

Regards,
Alex
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Re: [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI

[Alex]

Alex ha scritto:

Hi all!

We've released VoiceOne 0.4.0, a web-based and open source solution 
which allows to fully manage an Asterisk service hosted on a LAMP server.


Thanks guys, translators and testers are welcome!

We have a dedicated forum at 
http://www.voiceone.it/forum/viewforum.php?f=5 where you'll be able to 
obtain all details by our translations responsible. However, we're 
sending you a personal e-mail with instructions attached.


Regards,
Alex


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[asterisk-users] debian/dahdi/zaphfc - Unable to receive TEI from network!

[Alex]

Hello,

The harddisk of my etch/bristuffed asterisk1.2 box finally died. I moved
the cheap (1397:2bd0) HFC-S card to a squeeze host (i686) and built
dahdi modules 2.3.0.1 using m-a. After zaptel->dahdi and asterisk
1.2->1.6 config adaptations, everything seems ok, except for the BRI
side, unable to bring layers 1/2 up.

Asterisk reports:
chan_dahdi.c:12393 dahdi_pri_error: 1 Unable to receive TEI from network!

deskpro*CLI> pri show spans
PRI span 1/0: Provisioned, Down, Active

Of course, trying to dial out using the channel results in:
app_dial.c: Unable to create channel of type 'DAHDI' (cause 34 -
Circuit/channel congestion)

I'm a little puzzled as to what's wrong, as I didn't change anything on
the ISDN side compared to my very stable etch setup: same HFC-S card
working in TE-mode, same bri_cpe_ptmp signalling, same NT1, I even
reused the same cable. I tried the force_l1_up=1 kmod parameter, without
seeing any change. I tried using fcshdlc instead of hardhdlc with no
luck. I even swapped the NT1 with a new one, also without result. Any
pointers on how to debug this TEI assignment
problem are welcome.

I have an plain ISDN phone that's also connected to the NT1's S-bus
which works fine. I can swap the cables between the phone and HFC-S card
or disconnect the phone, it doesn't change anything.

OS-level and asterisk-level stuff looks as per my expectations (which
may be misguided...). The only strange thing is the warn_slowpath_common
warning when loading zaphfc module, but I'm not sure whether this is
relevant as people seem to have a working setup despite seeing this
warning. Can anyone confirm that this is not relevant?

I include some info on my setup below. I can also provide config files
for the previously running etch/asterisk1.2 setup, if requested.

Thanks for any help

  alex


(I filtered out stuff that's IMHO non-relevant)

# dmesg
[7.795103] WARNING: at
/usr/src/modules/dahdi/drivers/dahdi/dahdi-base.c:5866
dahdi_register+0x39/0x296 [dahdi]()
[7.795110] Hardware name: Deskpro
[7.795115] Modules linked in: zaphfc(+) dahdi snd_intel8x0(+)
snd_ac97_codec ac97_bus crc_ccitt snd_pcm snd_timer parport_pc i2c_i801
snd shpchp parport soundcore processor button pcspkr evdev i2c_core
pci_hotplug snd_page_alloc rng_core ext3 jbd mbcache dm_mod raid1 md_mod
sd_mod crc_t10dif ata_generic uhci_hcd ata_piix ehci_hcd e100 libata
usbcore thermal floppy mii nls_base scsi_mod thermal_sys [last unloaded:
scsi_wait_scan]
[7.795201] Pid: 429, comm: modprobe Not tainted 2.6.32-5-686 #1
[7.795207] Call Trace:
[7.795231]  [] ? warn_slowpath_common+0x5e/0x8a
[7.795241]  [] ? warn_slowpath_null+0xa/0xc
[7.795252]  [] ? dahdi_register+0x39/0x296 [dahdi]
[7.795275]  [] ? printk+0xe/0x13
[7.795296]  [] ? hfc_probe+0x8db/0xb64 [zaphfc]
[7.795309]  [] ? hfc_probe+0x9c3/0xb64 [zaphfc]
[7.795334]  [] ? local_pci_probe+0xb/0xc
[7.795343]  [] ? pci_device_probe+0x41/0x63
[7.795359]  [] ? driver_probe_device+0x8a/0x11e
[7.795368]  [] ? __driver_attach+0x40/0x5b
[7.795378]  [] ? bus_for_each_dev+0x37/0x5f
[7.795386]  [] ? driver_attach+0x11/0x13
[7.795395]  [] ? __driver_attach+0x0/0x5b
[7.795403]  [] ? bus_add_driver+0x99/0x1c5
[7.795412]  [] ? driver_register+0x87/0xe0
[7.795422]  [] ? proc_register+0xf8/0x142
[7.795431]  [] ? __pci_register_driver+0x33/0x89
[7.795442]  [] ? hfc_init_module+0x0/0x30 [zaphfc]
[7.795451]  [] ? do_one_initcall+0x55/0x155
[7.795461]  [] ? sys_init_module+0xa7/0x1d7
[7.795475]  [] ? sysenter_do_call+0x12/0x28
[7.795482] ---[ end trace c3edff8cea26edee ]---
[7.799466] vzaphfc: card 0: resetting
[7.816194] vzaphfc: card 0 configured for TE mode at mem 0x4010
(0xe0d84000) IRQ 18
...
[   17.835509] dahdi_transcode: Loaded.
[   17.865454] dahdi_echocan_oslec: Registered echo canceler 'OSLEC'
[   18.851312] dahdi: Registered tone zone 30 (Switzerland)
[   18.851372] vzaphfc: card 0: chan B1: TX FIFO has become empty
[   18.851381] vzaphfc: card 0: chan B1 opened as ZTHFC1/0/1.
[   18.851413] vzaphfc: card 0: chan B1 closed as ZTHFC1/0/1.
[   18.851434] vzaphfc: card 0: chan B2: TX FIFO has become empty
[   18.851440] vzaphfc: card 0: chan B2 opened as ZTHFC1/0/2.
[   18.851455] vzaphfc: card 0: chan B2 closed as ZTHFC1/0/2.
[   18.851476] vzaphfc: card 0: chan D opened as ZTHFC1/0/3.
[   18.851486] vzaphfc: card 0: chan D closed as ZTHFC1/0/3.
[   25.788850] vzaphfc: card 0: chan B1 opened as ZTHFC1/0/1.
[   25.789108] vzaphfc: card 0: chan B2 opened as ZTHFC1/0/2.
[   25.789512] vzaphfc: card 0: chan D opened as ZTHFC1/0/3.

a single module handles the card (after I blacklisted hfcpci):
# lspci -vv
02:09.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 02)
Subsystem: Cologne Chip Designs GmbH ISDN Board
Flags: bus master, medium de

Re: [asterisk-users] debian/dahdi/zaphfc - Unable to receive TEI fromnetwork!

[Alex]

Tzafrir Cohen wrote:
> On Fri, Oct 01, 2010 at 01:49:48PM +0100, Andrew Thomas wrote:
>> What happens if you change to:
>>
>> signalling=bri_cpe_ptp
> 
> It's bri_cp , not bri_cpe_ptp .
> 

yes, bri_cpe, for p2p mode, that's what my last failure report was using
(the bri_cpe vs bri_cpe_ptmp inconsistency hurts a little, but lets keep
that for later). Note that I disconnected the phone that's sharing the
S0-bus with the HFC while doing this, for good measure.

Anyway, I understand this was just a test to help diagnose the problem
rather than a hint at a potential misconfiguration, as I'm pretty sure
my line is in p2mp mode; the ISDN phone happily shared the S0 with the
asterisk box for years.

To narrow down the source, I then put a "new" hdd (w/ squeeze on it) in
the "original" machine and put the HFC back in, in the slot it used to
be. Everything behaves exactly as reported in my initial mail, including
the warn_slowpath_common warning (I still don't know what to think of
it); this should discard machine/HFC incompatibility as the cause. The
interrupt is shared in this machine, but my etch/bristuff/ast1.2 was
happy about that, so that's not the point, unless this newer driver has
"enhanced" requirements.

However, the card is fine. To confirm this, I removed all dahdi stuff,
loaded debian stock hfcpci module and mISDN_dsp, built mISDNuser from
git and I can see incoming and outgoing call setups (from/to the phone
on the shared S0 bus) with misdn_log:

# tools/misdn_log
mISDN kernel version 1.01.21 found
mISDN user   version 1.01.21 found
1 controller found
id: 0
Dprotocols: 0006
Bprotocols: 006e
protocol:   0
channelmap: 0006
nrbchan:2
name:   hfc-pci.1
log bind ch(1) return -1
log bind error Invalid argument
log bind ch(0) return 0
0
[censored packets flow...]

# dmesg | grep --relevant
[7.517932] hfcpci :01:02.0: enabling device ( -> 0003)
[7.517960] hfcpci :01:02.0: PCI INT A -> Link[LNKB] -> GSI 9
(level, low) -> IRQ 9
[7.517972] mISDN_hfcpci: found adapter CCD/Billion/Asuscom 2BD0 at
:01:02.0
[7.517981] mISDN: HFC-PCI driver 2.0
[7.518131] HFC-PCI: defined at mem 0xd8d66800 fifo
0xd73d8000(0x173d8000) IRQ 9 HZ 250
[7.558468] HFC 1 cards installed
...
[   59.173173] DSP modul 2.0
[   59.173190] mISDN_dsp: DSP clocks every 64 samples. This equals 2
jiffies.
[   81.010222] base_sock_release(d748e340) sk=d6b9b600
[  106.181034] base_sock_release(d748e340) sk=d6b9b600
[  106.181093] connect_layer1: ret -22 (dev 0)
[  106.181192] init_card: entered
[  106.181222] reset_hfcpci: entered
[  106.181229] HFC_PCI: resetting HFC ChipId(30)
[  106.181241] HFC-PCI status(4) before reset
[  106.184031] HFC-PCI status(2) after reset
[  106.184031] HFC-PCI status(4) after 5us
[  106.184031] inithfcpci: entered
[  106.268053] HFC PCI: IRQ 9 count 33
[  106.268067] connect_layer1: ret 0 (dev 0)

subsequent launches of misdn_log will log this:
[ 1047.287483] base_sock_release(d7421a00) sk=d6b9ba00
[ 1047.287542] connect_layer1: ret -22 (dev 0)
[ 1047.287642] connect_layer1: ret 0 (dev 0)

I haven't yet configured misdn properly, but I can issue calls with
misdntestlayer3, so the card seems to behave well enough with misdn to
get a TEI and make a call.

I'm not that much thrilled by ISDN these days, I mostly want to get back
to a working setup. But since I've battled with this for a few days, if
a few more days are needed to help debug what appears to be a problem
with vzaphfc (?), I can spend some time. That is, if you care to provide
test scenarios and/or test/instrumented code. Tell me if this needs to
be moved off (this) list... jabber would be fine, if you say so.


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Re: [asterisk-users] Kernel panic (asterisk 1.8.0-rc3, dahdi-linux-2.4)

[Alex]

Hello,

I'm having a very similar issue with dahdi 2.3.0.1 / 2.6.32 (and others
confirmed the occurence with same software revisions, same kind of old
hardware - P3, P4, different HFC hardware). You can look at my last
report on loosely related debian bug #598886.

http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=598886
(see messages #20 and #25)

My quick fix was to disable echo cancellation, which is a bit heavy
handed but also worked for others. A number of my crashes also pointed
out the math_state_restore function (see mess #25), but I didn't knew
what to do with this hint. I'll test if things are better off with MMX
disabled in dahdi during the week-end... it looks promising.

Also, have a look at kernel/Documentation/preempt-locking.txt...

  Alex

Karsten Wemheuer wrote:
> Hi,
> 
> I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with
> dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an
> older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN
> card. The system starts without any errors.
> 
> I discovered a severe issue. The kernel panics on a very small load. The
> first call normally gets through. If I start the second or third call
> and sometimes when I terminate the first call, the system panics (Oops
> text on console).
> 
> After solving some difficulties (the relevant part of the Oops text
> scrolls out of the monitor, no serial interface), I get the text via
> netconsole. It seems to me, that the panic occurred in oslec (function
> "oslec_update"). But maybe I am wrong with this. In the oslec code there
> is a patch to enable MMX. After switching this off, the problem
> disappeared. AFAIK the cpu supports mmx.
> 
> Where should I address this issue to? Is it a known issue?
> 
> Here comes one example for the oops:
> 
> /-
> BUG: unable to handle kernel NULL pointer dereference at (null)
> IP: [] __math_state_restore+0x56/0x90
> *pde =  
> Oops:  [#1] PREEMPT SMP 
> last sysfs file: /sys/module/configfs/initstate
> Modules linked in: netconsole configfs dahdi_echocan_oslec echo capifs
> loop wcb4xxp rtc_cmos i2c_i801 rtc_core dahdi 8250_pnp 8139too floppy
> 8250 rtc_lib mii serial_core i2c_core processor pcspkr rng_core button
> ide_pci_generic ide_core sd_mod crc_t10dif thermal [last unloaded:
> netconsole]
> 
> Pid: 1268, comm: clip.agi Not tainted 2.6.35.4 #1
> P4Dual-915GL/P4Dual-915GL
> EIP: 0060:[] EFLAGS: 00010046 CPU: 0
> EIP is at __math_state_restore+0x56/0x90
> EAX:  EBX: c5b2 ECX: cd461960 EDX: 
> ESI: cd461960 EDI: c01045a0 EBP: 0080 ESP: c5b21cb0
>  DS: 007b ES: 007b FS: 00d8 GS: 00e0 SS: 0068
> Process clip.agi (pid: 1268, ti=c5b2 task=cd461960 task.ti=c5b2)
> Stack:
>  c5b21cd0 0027 c01045a0 c01045e5 0200  cfadd500 c0432273
> <0> cfadd500 cfadd200 0008 0027 0080 0080 cf33fa00
> 007b
> <0> 007b c02d00d8 00e0  d0ae2153 0060 00010002
> 005a
> Call Trace:
>  [] ? do_device_not_available+0x0/0x60
>  [] ? do_device_not_available+0x45/0x60
>  [] ? error_code+0x73/0x80
>  [] ? DAC960_V1_ProcessCompletedCommand+0x1108/0x1510
>  [] ? oslec_update+0xe3/0x5c0 [echo]
>  [] ? echo_can_process+0x28/0x40 [dahdi_echocan_oslec]
>  [] ? echo_can_process+0x0/0x40 [dahdi_echocan_oslec]
>  [] ? dahdi_ec_span+0x268/0x2a0 [dahdi]
>  [] ? b4xxp_interrupt+0x11c/0x358 [wcb4xxp]
>  [] ? handle_IRQ_event+0x2d/0xc0
>  [] ? scsi_decide_disposition+0x16d/0x180
>  [] ? handle_fasteoi_irq+0x65/0xd0
>  [] ? handle_irq+0x15/0x30
>  [] ? do_IRQ+0x47/0xc0
>  [] ? common_interrupt+0x30/0x40
>  [] ? load_balance+0x550/0x7d0
>  [] ? _raw_spin_unlock_irq+0x4/0x20
>  [] ? finish_task_switch+0x3a/0x90
>  [] ? schedule+0x1c9/0x520
>  [] ? common_interrupt+0x30/0x40
>  [] ? preempt_schedule+0x2f/0x50
>  [] ? do_wp_page+0x160/0x960
>  [] ? handle_mm_fault+0x5d2/0xaa0
>  [] ? do_page_fault+0x0/0x370
>  [] ? do_page_fault+0x140/0x370
>  [] ? copy_strings+0x17f/0x1a0
>  [] ? do_execve+0x2be/0x310
>  [] ? do_execve+0x2be/0x310
>  [] ? sys_execve+0x40/0x70
>  [] ? do_page_fault+0x0/0x370
>  [] ? error_code+0x73/0x80
> Code: 89 c2 0f ae 2f 85 c9 75 27 83 4b 0c 01 80 86 98 00 00 00 01 8b 1c
> 24 8b 74 24 04 8b 7c 24 08 83 c4 0c c3 66 90 8b 86 50 02 00 00 <0f> ae
> 08 eb d9 e8 c0 ed 01 00 90 83 c8 08 e8 c7 ed 01 00 90 b8 
> EIP: [] __math_state_restore+0x56/0x90 SS:ESP 0068:c5b21cb0
> CR2: 
> ---[ end trace 65c27cd3a6b7bd8a ]---
> \-
> 
> Thanks,
> 
> Karsten
> 
> 
> 


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[asterisk-users] Reading DTMF sent by callee during a SIP call

[Alex]

Hi everyone,

I am looking for advice about the design of a SIP-based intercom. I
count on your help, as my current attempts are not fruitful (yet).

This will be a pretty long message, so here's my fundamental question:

Is there a way to interpret DTMF tones sent by the calee
(not the caller) while a voice call is in progress?






Here's the desired scenario:

- there is a box with speakers and a mic
- Asterisk is running on a computer inside that box
- the box is embedded in a door
- There are two user accounts, UserA and userB
- UserA is a client that runs on the server*
- UserA calls UserB and they are having a voice conversation


Throughout the call, Asterisk must react to DTMF tones sent by userB;
such that an action is executed when a specific key is pressed.

The idea is to build an intercom that would enable me to open a door
remotely, by relying entirely on SIP, so there would be no need to
have some additional communication channel to send the "open door"
signal.




I have previously implemented IVRs using `Background` and jumped to
specific extensions, when a button was pressed. But in that case, the
extensions are dialed by the caller; whereas now the input must from
the person who answered the call.

If I use `Dial` and `Read` - the latter is only executed after `Dial`
terminates - so this is not suitable.


`Background` behaves like I need - but it plays back a predefined
file, so it is not suitable for an interactive conversation.



* Having a SIP client on the same machine as the Asterisk server
itself is not possible, because both won't be able to bind to port
5060. My guess is that the solution is to originate a call from the
CLI; but I haven't gotten to that part yet.




Thank you for your patience, I am looking forward to your feedback,
Alex

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Re: [asterisk-users] Reading DTMF sent by callee during a SIP

[Alex]

Don,
> Isn't it easier just to use a SIP door phone?
Not in my case. The purpose of the whole exercise is to turn it into a
DIY project and hopefully, train a bunch of kids how to tinker with
RaspberryPi, Linux and Asterisk. Therefore, my objective is not to
"give a fish", but to "teach how to fish" and make them able to see
the fun in it :-)


Ish,
> You could create your own feature in features.conf that executes a
> Macro/Gosub defined in sip.conf...
Hmmm, I have never dealt with customized features.conf in the past, so
I will look into that direction.



So far, my approach is to leverage what I already know - and that is
to simply transfer the call to a specific extension; which, in turn -
will call System() in order to send a command to the electro-magnet
that keeps the door locked.

The downside is that one would have to press #5 (assuming that 5 is
that "magic" extension for opening the door), rather than just #.



Alex

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[asterisk-users] Transfer call placed from console (with chan_alsa)

[Alex]

Hi everyone.

Having experimented a but with a prototype of a system I described in
an earlier thread (Reading DTMF sent by callee during a SIP call), I
decided to implement my requirement by transferring the call to
another extension. This way, the callee can open the door by pressing
#1, and the dial plan for extension 1 takes care of the rest.

This works when I make a typical SIP to SIP call, but it doesn't when
I call from the console, using chan_alsa. I can see that the transfer
feature is inactive:

rasterisk*CLI> core show channeltype console
-- Info about channel driver: Console --
  Device State: no
Indication: yes
 Transfer : no
  Capabilities: 0x40 (slin)
   Digit Begin: no
 Digit End: yes
Send HTML : no
 Image Support: no
  Text Support: yes



However, I am unable to find a way to activate it. How can I transfer
placed from the console? Is it possible, in principle?


Alex

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[Asterisk-Users] FAQ

[Alex Lopez]

Is there a FAQ for this group???

I have many questions. Most of them I am sure have been answered time
and time again.


examples:

How do I configure an ATA 186 to work with *
I have an t1000 hooked up to an Adtran Atlas, but no work.  
How can I write ACD and IVRs?
Is there a error log for asterisk, I try to start it but it just
exits. No errors, no prompt, nothing..
etc. etc.



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[Asterisk-Users] HELP, I am a newbie.

[Alex Lopez]

OK this is what I have done so far.

I compiled and install all packages. No errors (wow!)

I have configured /etc/zaptel.conf with one span on a t1000 card. as follows 
span=1,0,0,esf,b8zs

I have loaded the demo configs, I have no alarms on my Atlas 550, however when ever I 
place a call to the unit I get a unavailable from the ISDN side.  What am I doing 
wrong???  All I want at this point is to have ALL the channels on the PRI be two-way, 
meaning that I want to call into * and have it answer. Later if I want to place a 
call, grab the next available B-chan and place call. I will not have a channel bank 
connected to this for analog phones, I will be a VOIP and IVR only box for now.


I few pointers in the right direction would be GREATLY appreciated. Once I understand 
the thinking behind the configs I can get going.


THANK YOU!!!

alex

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[Asterisk-Users] chan_zap.c Warning : channel already in use

[Alex Zarubin]

Title: chan_zap.c Warning : channel already in use





Hi,


There are several channels on the PRI span with the periodic warning:


WARNING[9226]: File chan_zap.c, Line 5437 (pri_dchannel): Ring requested on channel 21 already in use on span 1.  Hanging up owner.

1. Any known reason for this message?
2. Is there a way to reset the channels in question (without resetting the whole span)?


Thank you
Alex

[Asterisk-Users] segmentation fault

[Alex Zarubin]

Title: segmentation fault





Configuration:
Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown
P4 2.5 GHz, 1 GB RAM
T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s).
Each call gets transferred (Dial) to the SIP platform and stays for 5 min.


Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault.
Case 2. Asterisk built out of CVS Apr. 1. Test was running for 12 hours. Segmentation fault.


No coredump found. In case 1 there was a significant memory growth:
Top at the startup:
15986 root   9   0  6440 6436  2144 S 0.0  0.6   0:00 asterisk
15987 root   8   0  6440 6436  2144 S 0.0  0.6   0:00 asterisk
Top in several hours:
15986 root   9   0  9192 9188  2148 S 0.0  0.9   0:00 asterisk
15987 root   9   0  9192 9188  2148 S 0.0  0.9   0:00 asterisk
Top after a day:
27441 root   9   0 45980  44M  2156 S 0.0  4.5   0:00 asterisk
27442 root   8   0 45980  44M  2156 S 0.0  4.5   0:16 asterisk
Actually, I saw it over 50.


There were some warning messages on the way. For example:


Apr  1 23:22:33 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: 
Read on 86 failed: Unknown error 500
Apr  1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: 
!! Got reject for frame 102, retransmitting frame 102 now, updating n_r!
Apr  1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: 
!! Got reject for frame 103, but we have nothing -- resetting!
Apr  1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: 
!! Got reject for frame 29, retransmitting frame 29 now, updating n_r!
Apr  1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: 
!! Got reject for frame 30, but we have nothing -- resetting!
Apr  1 23:30:00 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: 
Read on 87 failed: Unknown error 500
Apr  1 23:30:24 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: 
Read on 86 failed: Unknown error 500


Question:
What do I do to give you more info? Should I issue 'ulimit -c unlimited' to get a coredump?
Are there any flags/modes to set?


Thank you.
Alex Zarubin

RE: [Asterisk-Users] segmentation fault

[Alex Zarubin]

Title: RE: [Asterisk-Users] segmentation fault





OK, here it is. On a flow of shorter calls it lasted about an hour.


[EMAIL PROTECTED] asterisk]# gdb asterisk core.12348
GNU gdb Red Hat Linux 7.x (5.0rh-15) (MI_OUT)
Copyright 2001 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
Type "show copying" to see the conditions.
There is absolutely no warranty for GDB.  Type "show warranty" for details.
This GDB was configured as "i386-redhat-linux"...
Core was generated by `asterisk -dgvvvc'.
Program terminated with signal 11, Segmentation fault.
Reading symbols from /lib/libdl.so.2...done.
Loaded symbols for /lib/libdl.so.2
Reading symbols from /lib/i686/libpthread.so.0...done.


warning: Unable to set global thread event mask: generic error
[New Thread 1024 (LWP 12342)]
Error while reading shared library symbols:
Can't attach LWP 12342: No such process
Reading symbols from /usr/lib/libncurses.so.5...done.
Loaded symbols for /usr/lib/libncurses.so.5
Reading symbols from /lib/i686/libm.so.6...done.
Loaded symbols for /lib/i686/libm.so.6
Reading symbols from /lib/i686/libc.so.6...done.
Loaded symbols for /lib/i686/libc.so.6
Reading symbols from /lib/ld-linux.so.2...done.
Loaded symbols for /lib/ld-linux.so.2
Reading symbols from /usr/lib/asterisk/modules/chan_modem.so...done.
.
.
.
.
Loaded symbols for /lib/libcrypt.so.1
Reading symbols from /lib/libnsl.so.1...done.
Loaded symbols for /lib/libnsl.so.1
Reading symbols from /usr/lib/asterisk/modules/format_pcm_alaw.so...done.
Loaded symbols for /usr/lib/asterisk/modules/format_pcm_alaw.so
#0  0x08055d86 in ast_queue_frame (chan=0x81cd398, fin=0x42c1991c, lock=1)
    at channel.c:354
354 cur = chan->pvt->readq;
(gdb) bt
#0  0x08055d86 in ast_queue_frame (chan=0x81cd398, fin=0x42c1991c, lock=1)
    at channel.c:354
#1  0x0805a9a0 in ast_queue_hangup (chan=0x81cd398, lock=1) at channel.c:391
#2  0x42412855 in handle_request (p=0x82b6350, req=0x42c1b25c, sin=0x42c1b24c)
    at chan_sip.c:3762
#3  0x42412e6d in sipsock_read (id=0x80d7850, fd=10, events=1, ignore=0x0)
    at chan_sip.c:3840
#4  0x08050d9e in ast_io_wait (ioc=0x80d9018, howlong=1000) at io.c:268
#5  0x424131f5 in do_monitor (data="" at chan_sip.c:3928
#6  0x4003ec6f in pthread_start_thread (arg=0x42c1bbe0) at manager.c:284
(gdb)






-Original Message-
From: Martin Pycko [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, April 02, 2003 11:38 AM
To: '[EMAIL PROTECTED]'
Subject: Re: [Asterisk-Users] segmentation fault



asterisk -vvvcg (use g option to generate the coredump file)
than gdb asterisk core.pid
bt


Also you might send a log of "pri intense debug span "


regards
Martin


On Wed, 2 Apr 2003, Alex Zarubin wrote:


> Configuration:
> Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown
> P4 2.5 GHz, 1 GB RAM
> T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s).
> Each call gets transferred (Dial) to the SIP platform and stays for 5 min.
>
> Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days.
> Segmentation fault.
> Case 2. Asterisk built out of CVS Apr. 1. Test was running for 12 hours.
> Segmentation fault.
>
> No coredump found. In case 1 there was a significant memory growth:
> Top at the startup:
> 15986 root   9   0  6440 6436  2144 S 0.0  0.6   0:00 asterisk
> 15987 root   8   0  6440 6436  2144 S 0.0  0.6   0:00 asterisk
> Top in several hours:
> 15986 root   9   0  9192 9188  2148 S 0.0  0.9   0:00 asterisk
> 15987 root   9   0  9192 9188  2148 S 0.0  0.9   0:00 asterisk
> Top after a day:
> 27441 root   9   0 45980  44M  2156 S 0.0  4.5   0:00 asterisk
> 27442 root   8   0 45980  44M  2156 S 0.0  4.5   0:16 asterisk
> Actually, I saw it over 50.
>
> There were some warning messages on the way. For example:
>
> Apr  1 23:22:33 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error):
> PRI:
> Read on 86 failed: Unknown error 500
> Apr  1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error):
> PRI:
> !! Got reject for frame 102, retransmitting frame 102 now, updating n_r!
> Apr  1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error):
> PRI:
> !! Got reject for frame 103, but we have nothing -- resetting!
> Apr  1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error):
> PRI:
> !! Got reject for frame 29, retransmitting frame 29 now, updating n_r!
> Apr  1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error):
> PRI:
> !! Got reject for frame 30, but we have nothing -- resetting!
> Apr  1 23:30:00 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error):
> PRI:
> Read on 87 fail

[Asterisk-Users] SIP INVITE and ACK go to different ports

[Alex Zarubin]

Title: SIP INVITE and ACK go to different ports





Greetings,


CVS 05/23/03. 10.50.4.140 is an * box. I see SIP INVITE to port 5060 and ACK (after OK) to port 32824.
The log is attached.


tcpdump shows 


18:31:48.380006 10.50.4.140.5060 > wmssqa02.webley.5060:  udp 615 (DF)
18:31:48.390007 wmssqa02.webley.32824 > 10.50.4.140.5060:  udp 331 (DF)
18:31:48.500018 wmssqa02.webley.32824 > 10.50.4.140.5060:  udp 554 (DF)
18:31:48.540022 wmssqa02.webley.32824 > 10.50.4.140.5060:  udp 540 (DF)
18:31:48.540022 10.50.4.140.5060 > wmssqa02.webley.32824:  udp 369 (DF)



17.1.1.2 Formal Description


   ...The ACK MUST be sent to the same address, port, and transport
   to which the original request was sent...


I don't see configuration problems but cannot be 100% sure. I think it was working before (on a different * box).


Thank you.


Alex Zarubin


 <> 





z
Description: Binary data

[Asterisk-Users] Call Back

[Alex Lopez]

We have Nextel Cell phones where incoming calls are free!

 

I would like to call a DID number on my Asterisk server,
have it grab my caller ID, not pick up, wait a few seconds, and call me back.  

 

I have already set up qcall to do this and pass it to a
context that asks for a password via authenticate, and is limited to dialing in
my local area so I am not worried about fraud. 

 

I am at the point where it all works except I do not know
the variables in extension.conf  {$CALLERID} is the whole strings
including name!!  I want just the number.

 

I could also use this to set up a ANI announcement where you
call the * box and it would use SayDigits to read the number you are calling
from.  I searched the archives via google and found nothing.

 

Anybody got any ideas???

 

 

[Asterisk-Users] Dual T400P, SMP, performance issues

[Alex Zarubin]

Title: Dual T400P, SMP, performance issues





Hi, 


We are trying to validate Asterisk as a media gateway PRI <-> SIP with two T400P (8 T1s) per box. The first
experience with BOX1 (Compaq, 2.53 GHz, 1 Gb RAM) and just one T400P was encouraging - on the load
test with 3 T1s worth of calls we had on average 75% idle CPU.


Not so with BOX2 (Dell, single 2.6 GHz Xeon, 1 Gb RAM, 2 T400P) and BOX3 (Dell, dual 2.6 GHz Xeon,
2 Gb RAM, 2 T400P, asterisk/zaptel is built with SMP support).


On the similar load test (as with the BOX1) BOX2 was showing 0% idle CPU 70% of the time. Just 3 T1s
out of 8.


On the load test with just 2 T1s BOX3 was very close to 0% idle on CPU0, CPU1 was at 95% idle.
The process ksoftirqd_CPU0 was close to the top of the 'top', with /proc/interrupts showing tor2 related
numbers growing very fast. We had 2 T1s plugged into the first T400P board, with nothing going into the second,
but the number of interrupts for the both boards was growing at the same pace. Here are the interrupts
(after the box reboot, so they are not that big as they were) - do they look OK?



    CPU0   CPU1   CPU2   CPU3   
  0: 122556  0  0  0    IO-APIC-edge  timer
  1:  4  0  0  0    IO-APIC-edge  keyboard
  2:  0  0  0  0  XT-PIC  cascade
  5:  0  0  0  0   IO-APIC-level  usb-ohci
  8:  1  0  0  0    IO-APIC-edge  rtc
 12: 20  0  0  0    IO-APIC-edge  PS/2 Mouse
 14: 23  0  2  0    IO-APIC-edge  ide0
 20: 516930  0  0  0   IO-APIC-level  tor2
 24: 516524  0  0  0   IO-APIC-level  tor2
 28:  10600  0  0  0   IO-APIC-level  eth0
 29:   4837  0  0  0   IO-APIC-level  eth1
 30:  24831  0  0  0   IO-APIC-level  aacraid
NMI:  0  0  0  0 
LOC: 122430 122429 122429 122428 
ERR:  0
MIS:  0


Not sure what went wrong. Any suggestions on how to work with 2 T400P in a box (without hurting performance)
and how to get advantage of SMP for Asterisk would be appreciated.


Any known Linux kernel related issues (2.4.20-13.7smp #1 SMP for BOX3 )?


Thank you.


Alex Zarubin

[Asterisk-Users] Setting local IP address for the RTP port

[Alex Zarubin]

Title: Setting local IP address for the RTP port





If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP?
Anything in rtp.conf ? 


Thank you.


Alex Zarubin

RE: [Asterisk-Users] Setting local IP address for the RTP port

[Alex Zarubin]

Title: RE: [Asterisk-Users] Setting local IP address for the RTP port





Listening is not a problem. When we send RTP packets it's important
to make sure we use the specific interface. For example, one interface
is on internal subnet and the other one is on external. QoS etc.


Do you think we'll have to change code for that? My guess it's a
feature needed by many (and easy to implement).


Thank you.
Alex Zarubin


-Original Message-
From: Tilghman Lesher [mailto:[EMAIL PROTECTED]]
Sent: Monday, June 09, 2003 8:59 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Setting local IP address for the RTP port



On Monday 09 June 2003 20:09, Alex Zarubin wrote:
> If there are multiple NICs in the box, how do we specify the local IP
> address to be used for RTP?


You can't.  RTP will automatically listen on all interfaces.


-Tilghman


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[Asterisk-Users] SIP sdp o= and c= fields

[Alex Zarubin]

Title: SIP sdp o= and c= fields





Hello,


If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip
IP address. At this point RTP packets will be coming to the default asterisk IP address.
For the machine with multiple interfaces this could be not the right one (not what we want).


Could it be configured (in rtp.conf or in sip.conf per context) ?


Thank you.


Alex Zarubin

RE: [Asterisk-Users] Bandwidth measurement tool: bmtools

[Alex Zarubin]

Title: RE: [Asterisk-Users] Bandwidth measurement tool: bmtools





http://s-tech.elsat.net.pl/bmtools/


-Original Message-
From: Steve Bourg [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, June 11, 2003 11:44 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Bandwidth measurement tool: bmtools



I can't resolve this host from anywhere.  Is there a mirror somewhere?


Thanks,


Steve Bourg


On Sat, 7 Jun 2003, John Todd wrote:


>
> This is not specifically on-topic for Asterisk, but I have found on
> many occasions while working with Asterisk that it would have been
> very handy to be able to measure, with some precision, the bandwidth
> being used by a particular host, port, or combination of the two.
>
> So, I went searching for various tools, none of which were what I
> wanted.  They either were too clever, or too limited in their
> abilities.
>
> However, someone forwarded the link to this tool to me about an hour
> ago, and I've been thrilled that it does _exactly_ what I want.  I
> can use a BPF-style filter to monitor exactly what I'd like to watch,
> and it hands back results to me in "real time" down to a one-second
> interval.  Sometimes, a small program can make me very happy, and I
> suppose after a morning full of various system problems I'm overly
> happy have something that works and does just what I want it to.
>
> This is useful for checking to see how much bandwidth a codec
> _really_ uses, or seeing what your total usage is between two IAX
> hosts, or pretty much anything that requires live examination of
> ethernet segment traffic.
>
> http://s-tech.linux-pl.com/bmtools/
>
>
> [EMAIL PROTECTED] bmtools-0.71]# ./rate -r 1 -f 'host 10.0.1.3 and not port ssh'
> -> Currently 263.05 Bps/3.01 pps, Average: 263.05 Bps/3.01 pps
> -> Currently 2706.00 Bps/17.00 pps, Average: 1486.97 Bps/10.02 pps
> -> Currently 588.00 Bps/6.00 pps, Average: 1186.92 Bps/8.68 pps
> -> Currently 440.00 Bps/4.00 pps, Average: 1000.00 Bps/7.51 pps
> -> Currently 440.00 Bps/4.00 pps, Average: 887.91 Bps/6.81 pps
> -> Currently 2080.00 Bps/16.00 pps, Average: 1086.72 Bps/8.34 pps
> -> Currently 1282.00 Bps/9.00 pps, Average: 1114.64 Bps/8.43 pps
> -> Currently 10385.00 Bps/20.00 pps, Average: 2274.01 Bps/9.88 pps
> ^C
>
>
> JT
>
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RE: [Asterisk-Users] Dual T400P, SMP, performance issues

[Alex Zarubin]

Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues





Zaptel was compiled with -D__SMP__


We've installed irqbalance and the picture improved a lot
(thanks to Jared Smith). Do you still see problems in our /proc/interrupts?


The big issue for us now is that after 24+ hours of the test load PRI->SIP
our Dell PE2650, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, 2.4.20-18.7smp #1 SMP
stops responding to anything.


So the questions are:
    - are there known issues with PE2650 and ways to fix them?
    - can someone recommend the 'stable' 2.4 SMP kernel for this
      kind of load?
    - any expertise in this area will be appreciated


   CPU0   CPU1   CPU2   CPU3   
  0: 230710  30030  50050  0    IO-APIC-edge  timer
  1:  5  0  0    233    IO-APIC-edge  keyboard
  2:  0  0  0  0  XT-PIC  cascade
  5:  0  0  0  0   IO-APIC-level  usb-ohci
  8:  1  0  0  0    IO-APIC-edge  rtc
 14: 27  0  2  0    IO-APIC-edge  ide0
 20:    2085442 400221  0 230232   IO-APIC-level  tor2
 24: 293848    1841658  10010 570568   IO-APIC-level  tor2
 28:  5  25643  0  0   IO-APIC-level  eth0
 29:  5  0    5165040  0   IO-APIC-level  eth1
 30:  43720  35467   1291   3296   IO-APIC-level  aacraid
NMI:  0  0  0  0 
LOC: 310618 310616 310616 310616 
ERR:  0
MIS:      0


Thank you.
Alex Zarubin


-Original Message-
From: Martin Pycko [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, June 10, 2003 9:48 AM
To: '[EMAIL PROTECTED]'
Subject: Re: [Asterisk-Users] Dual T400P, SMP, performance issues



Are you sure that you compiled zaptel for __SMP__ ?
Edit your zaptel/Makefile.


  0:   75283844   75241320   75286285   75247088    IO-APIC-edge  timer
  1:  1  0  1  1    IO-APIC-edge  keyboard
  2:  0  0  0  0  XT-PIC  cascade
  3:  0  0  0  0   IO-APIC-level  usb-ohci
  8:  1  0  0  0    IO-APIC-edge  rtc
 15:  1  0  0  1    IO-APIC-edge  ide1
 16:   22134870   22120997   22135905   22122829   IO-APIC-level  eth0
 25:   4670   4548   4614   4518   IO-APIC-level  tor2


All the four CPU's should have IRQ's like in the example above.


Martin


On Mon, 9 Jun 2003, Alex Zarubin wrote:


> Hi,
>
> We are trying to validate Asterisk as a media gateway PRI <-> SIP with two
> T400P (8 T1s) per box. The first
> experience with BOX1 (Compaq, 2.53 GHz, 1 Gb RAM) and just one T400P was
> encouraging - on the load
> test with 3 T1s worth of calls we had on average 75% idle CPU.
>
> Not so with BOX2 (Dell, single 2.6 GHz Xeon, 1 Gb RAM, 2 T400P) and BOX3
> (Dell, dual 2.6 GHz Xeon,
> 2 Gb RAM, 2 T400P, asterisk/zaptel is built with SMP support).
>
> On the similar load test (as with the BOX1) BOX2 was showing 0% idle CPU 70%
> of the time. Just 3 T1s
> out of 8.
>
> On the load test with just 2 T1s BOX3 was very close to 0% idle on CPU0,
> CPU1 was at 95% idle.
> The process ksoftirqd_CPU0 was close to the top of the 'top', with
> /proc/interrupts showing tor2 related
> numbers growing very fast. We had 2 T1s plugged into the first T400P board,
> with nothing going into the second,
> but the number of interrupts for the both boards was growing at the same
> pace. Here are the interrupts
> (after the box reboot, so they are not that big as they were) - do they look
> OK?
>
>
> CPU0   CPU1   CPU2   CPU3
>   0: 122556  0  0  0    IO-APIC-edge  timer
>   1:  4  0  0  0    IO-APIC-edge  keyboard
>   2:  0  0  0  0  XT-PIC  cascade
>   5:  0  0  0  0   IO-APIC-level  usb-ohci
>   8:  1  0  0  0    IO-APIC-edge  rtc
>  12: 20  0  0  0    IO-APIC-edge  PS/2 Mouse
>  14: 23  0  2  0    IO-APIC-edge  ide0
>  20: 516930  0  0  0   IO-APIC-level  tor2
>  24: 516524  0  0  0   IO-APIC-level  tor2
>  28:  10600  0  0  0   IO-APIC-level  eth0
>  29:   4837  0  0  0   IO-APIC-level  eth1
>  30:  24831  0  0  0   IO-APIC-level  aacraid
> NMI:  0  0  0  0
> LOC: 122430 122429 122429 122428
> 

RE: [Asterisk-Users] Dual T400P, SMP, performance issues

[Alex Zarubin]

Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues





Mark,


As far as pings - we have cases when we could ping the box on both
interfaces and there are cases when we could not (we tried 3-4 sets of
NICs and drivers). All telnets, X, ssh etc. are definitely dead.
No coredumps (asterisk was started with -g option), no kernel panics.
Black console, Alt-SysRq combinations don't work.
Pretty much no options but rebooting the box.


As far as SMP and single T400P - we'll try and report the results
but the idea was to go with as high density as possible ...


What do you think of using hyperthreading - should we enable or disable it
for the box running asterisk?


What about -DCONFIG_ZAPTEL_WATCHDOG ? Can it help and how to use it?


Thank you.
Alex Zarubin


-Original Message-
From: Mark Spencer [mailto:[EMAIL PROTECTED]]
Sent: Saturday, June 14, 2003 10:23 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dual T400P, SMP, performance issues



When you say "stops responding" do you mean no more pings, telnet dead,
etc?  Or do you mean asterisk stops responding?  Is there a segfault or
kernel panic, or any other failure diagnostic?


Mark


On Thu, 12 Jun 2003, Alex Zarubin wrote:


> Zaptel was compiled with -D__SMP__
>
> We've installed irqbalance and the picture improved a lot
> (thanks to Jared Smith). Do you still see problems in our /proc/interrupts?
>
> The big issue for us now is that after 24+ hours of the test load PRI->SIP
> our Dell PE2650, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, 2.4.20-18.7smp #1 SMP
> stops responding to anything.
>
> So the questions are:
>   - are there known issues with PE2650 and ways to fix them?
>   - can someone recommend the 'stable' 2.4 SMP kernel for this
>     kind of load?
>   - any expertise in this area will be appreciated
>
>    CPU0   CPU1   CPU2   CPU3
>   0: 230710  30030  50050  0    IO-APIC-edge  timer
>   1:  5  0  0    233    IO-APIC-edge  keyboard
>   2:  0  0  0  0  XT-PIC  cascade
>   5:  0  0  0  0   IO-APIC-level  usb-ohci
>   8:  1  0  0  0    IO-APIC-edge  rtc
>  14: 27  0  2  0    IO-APIC-edge  ide0
>  20:    2085442 400221  0 230232   IO-APIC-level  tor2
>  24: 293848    1841658  10010 570568   IO-APIC-level  tor2
>  28:  5  25643  0  0   IO-APIC-level  eth0
>  29:  5  0    5165040  0   IO-APIC-level  eth1
>  30:  43720  35467   1291   3296   IO-APIC-level  aacraid
> NMI:  0      0  0  0
> LOC: 310618 310616 310616 310616
> ERR:  0
> MIS:  0
>
> Thank you.
> Alex Zarubin
>
> -Original Message-
> From: Martin Pycko [mailto:[EMAIL PROTECTED]]
> Sent: Tuesday, June 10, 2003 9:48 AM
> To: '[EMAIL PROTECTED]'
> Subject: Re: [Asterisk-Users] Dual T400P, SMP, performance issues
>
>
> Are you sure that you compiled zaptel for __SMP__ ?
> Edit your zaptel/Makefile.
>
>   0:   75283844   75241320   75286285   75247088    IO-APIC-edge  timer
>   1:  1  0  1  1    IO-APIC-edge  keyboard
>   2:  0  0  0  0  XT-PIC  cascade
>   3:  0  0  0  0   IO-APIC-level  usb-ohci
>   8:  1  0  0  0    IO-APIC-edge  rtc
>  15:  1  0  0  1    IO-APIC-edge  ide1
>  16:   22134870   22120997   22135905   22122829   IO-APIC-level  eth0
>  25:   4670   4548   4614   4518   IO-APIC-level  tor2
>
> All the four CPU's should have IRQ's like in the example above.
>
> Martin
>
> On Mon, 9 Jun 2003, Alex Zarubin wrote:
>
> > Hi,
> >
> > We are trying to validate Asterisk as a media gateway PRI <-> SIP with two
> > T400P (8 T1s) per box. The first
> > experience with BOX1 (Compaq, 2.53 GHz, 1 Gb RAM) and just one T400P was
> > encouraging - on the load
> > test with 3 T1s worth of calls we had on average 75% idle CPU.
> >
> > Not so with BOX2 (Dell, single 2.6 GHz Xeon, 1 Gb RAM, 2 T400P) and BOX3
> > (Dell, dual 2.6 GHz Xeon,
> > 2 Gb RAM, 2 T400P, asterisk/zaptel is built with SMP support).
> >
> > On the similar load test (as with the BOX1) BOX2 was showing 0% idle CPU
> 70%
> > of the time. Just 3 T1s
> > out of 8.
> >
> > On the load test with just 2 T1s BOX3 was very close to 0% idle on CPU0,
> > CPU1 was at 95% idle.
> > The proce

[Asterisk-Users] zttool shows OK for the T400P board which is not configured/started

[Alex Zarubin]

Title: zttool shows OK for the T400P board which is not configured/started





There are 2 T400P boards in a box. Just one board is configured in zaptel.conf:


span=1,1,0,esf,b8zs
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,2,0,esf,b8zs
...
(no configuration for spans 5-8)


ztcfg -vv reports 96 channels configured. But zttool shows OK on all 8 spans (4 lights on board 1
are green, no lights on board 2).


A minor issue, should be easy to fix.


Thank you.
Alex Zarubin

RE: [Asterisk-Users] Dual T400P, SMP, performance issues

[Alex Zarubin]

Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues





I believe this is related to the load, there are always calls in our test.
Attached is a part of /var/log/messages file with SysRq memory info - in
case you can see something in it. The box was rebooted 06-16 17:08 and
the problem occurred 06-17 11:36.


Thank you.
Alex Zarubin




-Original Message-
From: Mark Spencer [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, June 17, 2003 6:58 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dual T400P, SMP, performance issues



> As far as SMP and single T400P - we'll try and report the results
> but the idea was to go with as high density as possible ...


Right, I'm just trying to narrow down the problem.  I'm theorizing that
the problem is some sort of spinlock deadlock.  Does it only occur if
there is activity or even if the lines are up but no calls taking place?


> What do you think of using hyperthreading - should we enable or disable it
> for the box running asterisk?


We use hyperthreading but have not run tests longer than a few hours on
those machines.


> What about -DCONFIG_ZAPTEL_WATCHDOG ? Can it help and how to use it?
k
Likely will make no difference in this situation.


Mark


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mes_ast.gz
Description: Binary data

[Asterisk-Users] Integration with external ASR engines

[Alex Zarubin]

Title: Integration with external ASR engines





Hello,


Question for developers: what is the asterisk way to integrate with ASR (speech recognition)?
Question to the community: has someone done anything in this direction?


On the first glance that shouldn't be too hard. One part is delivering audio to the engine (for example,
main ASR players Nuance and Speechworks will be happy with RTP) - can be done via RTP forking.
The other part is stopping the prompt when speech is detected (barge-in implementation). One more
thing is adding the related commands to agi.


Any comments are welcome.


Thank you.
Alex Zarubin

RE: [Asterisk-Users] Dual T400P, SMP, performance issues

[Alex Zarubin]

Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues





Mark & Oliver,


It is too early to say, but the picture is different now. Our dual CPU,
dual T400P box is up for 4 days, under the load of 10 - 100 simultaneous
PRI -> SIP calls. We installed 2.4.21 #2 SMP (it was still freezing after
that) and, what I think made the difference, recompiled zaptel-libpri-asterisk
with gcc 3.3.


The problem, on the way, was that asterisk wouldn't start after that. It was
crashing while loading mp3 and lpc10 codecs. We put 'noload' for these two
into modules.conf - temporary solution, of course.


There are problems, still, with multiple connections at the same time. Windows
to the box get frozen for a sec, D-channel error messages. The following
messages are dumped into /var/log/messages. What do you think?


Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: wait_on_irq, CPU 1:
Jun 24 18:23:25 mspgate03 kernel: irq:  1 [ 0 0 1 0 ]
Jun 24 18:23:25 mspgate03 kernel: bh:   0 [ 0 0 0 0 ]
Jun 24 18:23:25 mspgate03 kernel: Stack dumps:
Jun 24 18:23:25 mspgate03 kernel: CPU 0:0200 036f 00e14603
1802 0310 6647 008e0200 4803
Jun 24 18:23:25 mspgate03 kernel:    0078 001ffa02 5b490300
0600 01c7 074e0308 1afe 01c74d03
Jun 24 18:23:25 mspgate03 kernel:    2302 d708 e101
0900 01d7 f5030001 0423 09300207
Jun 24 18:23:25 mspgate03 kernel: Call Trace:    []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[]
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: CPU 2:  
    
Jun 24 18:23:25 mspgate03 kernel:      
    
Jun 24 18:23:25 mspgate03 kernel:      
    
Jun 24 18:23:25 mspgate03 kernel: Call Trace:
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: CPU 3:0070 cce30002 0cd8
08fa 6953 656c706d 6c616e41 73697379
Jun 24 18:23:25 mspgate03 kernel:    0009a700 46534c00 65746e69
6c6f7072 32657461 6e655f61 0a810063 6953
Jun 24 18:23:25 mspgate03 kernel:    656c706d 65746e49 6c6f7072
4c657461 39004653 530b 6c706d69 66736c65
Jun 24 18:23:25 mspgate03 kernel: Call Trace:
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: CPU 1:e14d5eac c025c896 0001
0001  0001 c010a7c2 c025c8ab
Jun 24 18:23:25 mspgate03 kernel:     f2d92124 e14d5f00
c0191104 0500 1805 00bf 8a01
Jun 24 18:23:25 mspgate03 kernel:    7f1c0300 01000415 1a131100
170f1200  e14d4000  
Jun 24 18:23:25 mspgate03 kernel: Call Trace:    []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   []
Jun 24 18:23:25 mspgate03 kernel:


Thank you.
Alex Zarubin


-Original Message-
From: The Traveller [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, June 17, 2003 3:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dual T400P, SMP, performance issues



On Tue, Jun 17, 2003 at 20:54:39 +0200, The Traveller wrote:
> 
> BTW: As I reported in my previous mail to the list, I've now installed kernel
> 2.4.21-rc2 with ACPI-patch on the box with the E100P.  I've been trying
> very hard to reproduce a freeze with this kernel, but haven't succeeded yet.
[...]


Ok, it crashed again, so that wasn't it either.  What I did to trigger
it was using the auto-dialer to loop as many calls to app_datetime out
and then back over the same E-1 as it would take, queueing the calls
to "/var/spool/asterisk/outgoing/" 14 at a time.  It froze at the first
attempt.  The "good" news is that it produced a visible kernel-panic.
this time.  My guess is that you only don't see it if the console
screensaver has already come on while it happens.


It read something like "Unable to handle kernel paging request" and
happened in the swapper-task.  As usual, it dumped a lot of numbers on the
screen, which I didn't want to write down.


Mark: If you want my help in debugging this, I'll hook it up to a
serial console, trigger the crash and provide you with the exact
panic, together with the ksyms and modules-info to trace it.




    Grtz,


   Oliver
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RE: [Asterisk-Users] Dual T400P, SMP, performance issues

[Alex Zarubin]

-Original Message-
From: Mark Spencer [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, June 25, 2003 11:11 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dual T400P, SMP, performance issues



Oooh, how neat!  I wonder if there is some sort of race and that the
kernel is detecting and defeating it somehow.  Will ksymoops on your
machine handle that output?  Maybe we can track it down!


Again, does the problem occur with only one board?  i.e. is the problem
tied to having multiple boards in the machine?


Mark


On Tue, 24 Jun 2003, Alex Zarubin wrote:


> Mark & Oliver,
>
> It is too early to say, but the picture is different now. Our dual CPU,
> dual T400P box is up for 4 days, under the load of 10 - 100 simultaneous
> PRI -> SIP calls. We installed 2.4.21 #2 SMP (it was still freezing after
> that) and, what I think made the difference, recompiled
> zaptel-libpri-asterisk
> with gcc 3.3.
>
> The problem, on the way, was that asterisk wouldn't start after that. It was
> crashing while loading mp3 and lpc10 codecs. We put 'noload' for these two
> into modules.conf - temporary solution, of course.
>
> There are problems, still, with multiple connections at the same time.
> Windows
> to the box get frozen for a sec, D-channel error messages. The following
> messages are dumped into /var/log/messages. What do you think?
>
> ...
>
> Thank you.
> Alex Zarubin
>
> -Original Message-
> From: The Traveller [mailto:[EMAIL PROTECTED]]
> Sent: Tuesday, June 17, 2003 3:10 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Dual T400P, SMP, performance issues
>
>
> On Tue, Jun 17, 2003 at 20:54:39 +0200, The Traveller wrote:
> >
> > BTW: As I reported in my previous mail to the list, I've now installed
> kernel
> > 2.4.21-rc2 with ACPI-patch on the box with the E100P.  I've been trying
> > very hard to reproduce a freeze with this kernel, but haven't succeeded
> yet.
> [...]
>
> Ok, it crashed again, so that wasn't it either.  What I did to trigger
> it was using the auto-dialer to loop as many calls to app_datetime out
> and then back over the same E-1 as it would take, queueing the calls
> to "/var/spool/asterisk/outgoing/" 14 at a time.  It froze at the first
> attempt.  The "good" news is that it produced a visible kernel-panic.
> this time.  My guess is that you only don't see it if the console
> screensaver has already come on while it happens.
>
> It read something like "Unable to handle kernel paging request" and
> happened in the swapper-task.  As usual, it dumped a lot of numbers on the
> screen, which I didn't want to write down.
>
> Mark: If you want my help in debugging this, I'll hook it up to a
> serial console, trigger the crash and provide you with the exact
> panic, together with the ksyms and modules-info to trace it.
>
>
>
> Grtz,
>
>    Oliver
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>


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RE: [Asterisk-Users] Dual T400P, SMP, performance issues

[Alex Zarubin]

Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues





Here is info on the kernel panic with the high volume (110+) of calls.
Same configuration as before. Comments would be appreciated.


ksymoops 2.4.4 on i686 2.4.21.  Options used
 -V (default)
 -k /proc/ksyms (default)
 -l /proc/modules (default)
 -o /lib/modules/2.4.21 (specified)
 -m /boot/System.map-2.4.21 (default)
 -i


eax: 0100 ebx:  ecx:  edx: f71b5a14 esi: 0002 edi: f71b4000 ebp: f71b4000 esp: f71b59ec ds: 0018 es: 0018 ss: 0018

Process irqbalance (pid: 713, stackpage=f71b5000)
Stack:  6e6d6c6b 7271706f 76757473 7a797877 0001 c0115ef4 f71b4000 c02578fd f71b5a14 0001  0003 c0115ef4 f71b4000 f71b4000  f71b0018 c0110018 ffef c0114546 0010 0286 c0114470 

Call Trace: [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] [] []

Code: 89 1d b0 e0 ff ff ff 80 04 48 33 c0 eb 02 f3 90 a1 88 f3 30  
Using defaults from ksymoops -t elf32-i386 -a i386


Trace; c0115ef4 
Trace; c0115ef4 
Trace; c0110018 
Trace; c0114546 <.text.lock.smp+19/23>
Trace; c0114470 
Trace; c011bc88 
Trace; c01144c0 
Trace; c0114470 
Trace; c011b2d5 
Trace; c011eae2 
Trace; c011badb 
Trace; c011bc88 
Trace; c0116ff0 
Trace; c010960a 
Trace; c0115ef4 
Trace; c01173a8 
Trace; f89e7737 
Trace; f89fb1e0 
Trace; f89fb1e0 
Trace; c0117000 
Trace; c0109114 
Trace; c0115ef4 
Trace; c010abe3 
Trace; f897a8c0 <[usb-ohci]rh_int_timer_do+0/70>
Trace; f897a8c0 <[usb-ohci]rh_int_timer_do+0/70>
Trace; c0110018 
Trace; c0124345 
Trace; c012042b 
Trace; c01202d1 
Trace; c012005b 
Trace; c010abfe 
Trace; c015e751 
Trace; c0147513 
Trace; c01479f1 
Trace; f89e7737 
Trace; f89fb1e0 
Trace; c010e1b6 
Trace; c0123fc0 
Trace; c01482ab 
Trace; c01487c4 
Trace; c012042b 
Trace; c01202d1 
Trace; c012005b 
Trace; c010abfe 
Trace; c013c606 
Trace; c01471ae 
Trace; c013c953 
Trace; c0109023 
Code;   Before first symbol
 <_EIP>:
Code;   Before first symbol
   0:   89 1d b0 e0 ff ff mov    %ebx,0xe0b0
Code;  0006 Before first symbol
   6:   ff 80 04 48 33 c0 incl   0xc0334804(%eax)
Code;  000c Before first symbol
   c:   eb 02 jmp    10 <_EIP+0x10> 0010 Before first symbol
Code;  000e Before first symbol
   e:   f3 90 repz nop 
Code;  0010 Before first symbol
  10:   a1 88 f3 30 00    mov    0x30f388,%eax


Thank you.
Alex Zarubin

[Asterisk-Users] SIp Registration

[Alex Lopez]

I  use Windows Messenger ( I duck as to let the hurled
penguins barely miss my head J ) and I am able to
place and receive calls. So what is the problem you ask???  If I specify a
password in the password field of WM I get a Proxy Authentication Error during
SIP debug and I am not able to connect until I remove the secret=blah and do
not specify a password.  Has anyone had this problem before???