[Asterisk-Users] PSTN Gateway
Deepak Mittal
deepak at mittals.com
Tue Jan 20 10:12:28 MST 2004
Hello,
I am looking for information on setting up digium FXO card for use as a
PSTN Gateway (H323-PSTN) to work with GNUGk.
I am basically looking for the setup and it would be great if anyone can
share his experiences with the same. Also, if there are any limitations in
going for such a setup and problems that may arise/things that I should keep
in consideration.
Thanks & Regards,
Deepak
----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, January 20, 2004 12:08 PM
Subject: Asterisk-Users digest, Vol 1 #2557 - 10 msgs
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> Today's Topics:
>
> 1. FW: Memory problem (T. Chan)
> 2. X101P CallOut Big Problem. (Carlos Arnt)
> 3. RE: RE: Latest version of asterisk (Aram Ter-Martirosyan)
> 4. Re: SIP: Register that isn't a register? (Ing. Angel Gomez Garcia)
> 5. RE: FW: Memory problem (Adam Goryachev)
> 6. Call token is ip$localhost (Asan M.)
> 7. Re: CVS Changes (NAT-SIP) (Brian West)
> 8. Re: PLAYBACK multiple files (Marcin Kuzmicki)
> 9. Re: user password and call waiting (Brian West)
> 10. echo cancellation (dkwok)
>
> --__--__--
>
> Message: 1
> From: "T. Chan" <tommy.chan at utimail.com>
> To: <asterisk-users at lists.digium.com>
> Date: Mon, 19 Jan 2004 23:20:27 -0500
> Subject: [Asterisk-Users] FW: Memory problem
> Reply-To: asterisk-users at lists.digium.com
>
>
> Dear all,
>
> I have had an experience which I would run by all of you to see if this is
> normal.
>
> I am running a few asterisk servers with 512M RAM memory, and as I have
> mentioned in previous notes, I have experienced frequent crashes when
faced
> with more than 15-20 simultaneous calls. I have tried to find out if it
> could be due to (a) Xeon chip running HT, (b) old Kernel version 2.4.18-3,
> (c) old redhat linux version 7.3, (d) H323 library pwlib and openh323
> versions which are 1.5.2 and 1.12.2 respectively among many other
> parameters. So far, unfortunately, the matter has not been resolved.
> However, I have noticed that the memory usage on each server has built up
> with time after the server being rebooted. I have complained about using
> close to 500M even when there were very few calls on the server but nobody
> seemed to be able to let me know if they were running at high memory
usages
> except for Jesse who was telling me that his memory usages have always
been
> low. Very recently, I noticed that after I rebooted the servers, the
memory
> usage would start at about 80 M and even after started the Asterisk
threads,
> I was running at about 100 M and even when there were calls, I was running
> at about 100M-150M, but then after hours it would start to build up to
200M
> and then 250M and then....finally close to 500M even after I stopped the
> Asterisk threads, almost like there is a memory leak somewhere.
>
> I wonder if that is normal, if someone can please tell me, or if not
normal,
> what could be the cause to it and how should this be rectified.
>
> Thanks alot
>
>
> Tom
> ---
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>
> --__--__--
>
> Message: 2
> From: Carlos Arnt <carnt at intellissence.com>
> To: <asterisk-users at lists.digium.com>
> Date: Tue, 20 Jan 2004 02:39:09 -0200
> Subject: [Asterisk-Users] X101P CallOut Big Problem.
> Reply-To: asterisk-users at lists.digium.com
>
> <html><head><meta name=3D"Generator" content=3D"PocoMail 3 HTML/CSS=
> Generator"/>
> <style type=3D"text/css"><!--
> LI{display:list-item;margin:0.00in;}
> p{display:block;margin:0.00in;}
> body{}
> --></style>
> </head><BODY ><p><SPAN style=3D"font-size:10pt;">Hi all,</SPAN></p>
> <p> </p>
> <p><SPAN style=3D"font-size:10pt;">I just now receive the FXO X101P Card
but=
> can't at any way make then call out.</SPAN></p>
> <p><SPAN style=3D"font-size:10pt;">I can hear the signal, even call but
always=
> receive from my local operator error that or the number don't exist or
need=
> more numbers.</SPAN></p>
> <p> </p>
> <p><SPAN style=3D"font-size:10pt;">I play alot with txgain and rxgain,
but=
> none help me out.</SPAN></p>
> <p><SPAN style=3D"font-size:10pt;">Being honest i try alot !!!! 5 hours
and=
> none !!!</SPAN></p>
> <p> </p>
> <p><SPAN style=3D"font-size:10pt;">I'm using asterisk in his sample=
> configs.</SPAN></p>
> <p><SPAN style=3D"font-size:10pt;">I mean i call out using 1234=
> etc..</SPAN></p>
> <p><SPAN style=3D"font-size:10pt;">Zapata.conf is Ok</SPAN></p>
> <p><SPAN style=3D"font-size:10pt;">Zaptel.conf is ok</SPAN></p>
> <p><SPAN style=3D"font-size:10pt;">(I follow the Digium faqs, then for a
good=
> person that show-me this in the Asterisk IRC)</SPAN></p>
> <p><SPAN style=3D"font-size:10pt;">( Using here is an Asterisk=
> 7.1)</SPAN></p>
> <p> </p>
> <p><SPAN style=3D"font-size:10pt;">Did anyone know a txgain and rxgain
from=
> Brazilian lines ? (I'm trying with Vesper operator)</SPAN></p>
> <p><SPAN style=3D"font-size:10pt;">Did i need make something more ( i
know=
> that need) :)</SPAN></p>
> <p> </p>
> <p><SPAN style=3D"font-size:10pt;">Please could someone with lot's of
time=
> help-me out here with this simple question ?</SPAN></p>
> <p><SPAN style=3D"font-size:10pt;">I just wanna call out too !!!
</SPAN></p>
> <p> </p>
> <p><SPAN style=3D"font-size:10pt;">Thanks alot !</SPAN></p>
> <p> </p>
> </body></html>
>
>
> --__--__--
>
> Message: 3
> From: "Aram Ter-Martirosyan" <aram at hi-teck.com>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] RE: Latest version of asterisk
> Date: Mon, 19 Jan 2004 20:42:26 -0800
> Reply-To: asterisk-users at lists.digium.com
>
> Hello Matt,
> Is that the Wildcard TE410P you are using. Digium said that it had some
> problems with Redhat 9.0 is that correct?
>
> - Digium quad T1 card
> - 3 T1's (2 x B8ZS ESF Long Distance and 1 x robbed-bit SF local)
> - Redhat 9.0
>
> Aram Ter-Martirosyan
> Senior Account Manager
> Hi-Tech Gateway, Inc.
> http://www.hi-teck.com
> 1225 Grand Central Ave.
> Glendale, CA 91201
> aram at hi-teck.com
> tel 818.546.4601
> fax 818.546.4617
> Turning Technology Into Business Solutions
>
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of mattf
> Sent: Monday, January 19, 2004 6:21 PM
> To: 'asterisk-users at lists.digium.com'
> Subject: RE: [Asterisk-Users] RE: Latest version of asterisk
>
>
> Hello,
>
> Our max for a single machine is 40 concurrent SIP -> Zap conversations for
> about a 12 hour period and over 5000 total phone calls per day. We didn't
> see crashes going over that, but we wanted to be safe and now have 2
> identical machines handling upto about 30 concurrent SIP -> Zap calls(3000
> phone calls per day), and a third old machine for office use that never
gets
> over 10 concurrent calls. Here's the specs for these systems:
>
> - 120 installed hardphones:
> - 80 x grandstream 102 hardphones
> - 20 x Sipura analog adapters(2 phones each)
> - 2 x Asterisk servers
> - 2.6 GHz Pentium4 800MHz bus w/ HyperThreading enabled
> - Asus p4c800 800MHz mobo
> - 2GB DDR400 RAM (This is actually overkill you need 1GB max if you
> reboot weekly)
> - 4 x 36GB SCSI drives in RAID 10 w/megaraid card
> - 3com 905CX ethernet card
> - Digium quad T1 card
> - 3 T1's (2 x B8ZS ESF Long Distance and 1 x robbed-bit SF local)
> - Redhat 9.0
> - Asterisk with many modules turned off and no MOH
>
> With these servers you can see the load average jump from 0.00 to 6.25 in
a
> matter of a minute and then back down again, all while never dropping a
call
> or crashing.
>
> We also recently diagnosed our lock-freeze to the touchy manager
> interface(if you are logged into the manager interface and you loose
> connection, the manager outgoing buffer seems to overflow and freeze
> Asterisk). So it doesn't seem to be a problem of hardware. But we still
> haven't figured out how to fix it.
>
> One note as to Ethernet cards, we actually fried a Realtek 8139 Ethernet
> card that we had put in a server temporarily as we were doing our testing.
> It started to generate a lot of errors and dropping packets left and
right.
> When we took it out it was VERY hot. We then put in a 3com 905 card and
> haven't had an issue with it yet.
>
> Hope this helps,
>
> MATT---
>
>
>
> -----Original Message-----
> From: T. Chan [mailto:tommy.chan at utimail.com]
> Sent: Monday, January 19, 2004 4:49 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] RE: Latest version of asterisk
>
>
> Thanks, Matt !
>
> So, am I correct in assuming that there are quite a few (or alot) of us
who
> have had not so good experiences with Asterisk? That Asterisk would crash
> after it hit a certain number of calls or after a certain period of time
> with 15-20 calls? I understand that there were others who were able to
send
> a good number of calls through but can anyone tell us if they have had
> tested and confirmed that Asterisk runs better without or with HT and in
> terms of number of calls, how many would each one support, in the
ballpark?
> It would also be nice if one could tell us the computer configuration in
> order to send that many calls without crashing Asterisk. Does it make a
> difference running the LAN on a ONBOARD LAN card as compared to a PCI
Intel
> or 3COM LAN card, since there is a chance that packets are passing more
> efficiently on a PCI LAN card?
>
> Side question: Is it possible to do passthrough faxing? Like, customers
> sending me H323 or SIP fax calls and the Asterisk will pass through to
> another gateway? Anyone successful in doing that?
>
> Tommy
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of mattf
> Sent: Monday, January 19, 2004 8:32 AM
> To: 'asterisk-users at lists.digium.com'
> Subject: RE: [Asterisk-Users] RE: Latest version of asterisk
>
>
> Hello,
>
> I've had Asterisk installed on HT capable machines in both HT mode(with
SMP)
> and non HT mode (with non-SMP) and did not notice any differences
> functionally between them. The processor load was always less in HT SMP
mode
> than non HT and I have experienced Asterisk deadlocks in both modes so it
> doesn't really seem to matter if you leave HT on(at least in my
> experiences).
>
> HT basically works by splitting off commands to one of two different
virtual
> processors that both run at about 70% of processor's speed(that's why you
> may notice compiling to take longer when in HT mode) I have heard of some
> applications having memory addressing errors with HT but I have not seen
any
> evidence to support that in Asterisk thus far.
>
> I'm going to try installing a 4 x T1 card on my Athlon 2xMP server next
week
> and see if Asterisk/Digium performance/compatibility improves over the
Intel
> platform.
>
>
> MATT---
>
>
> -----Original Message-----
> From: WipeOut [mailto:wipe_out at users.sourceforge.net]
> Sent: Monday, January 19, 2004 2:54 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] RE: Latest version of asterisk
>
>
> T. Chan wrote:
>
> >Dear All
> >
> >Should one enable HT in the chip when running Asterisk or if we don't,
> would
> >that offer alot less processing power?
> >
> >T
> >
> I have read before that HT did not help Asterisk so should be dissabled,
> but as the chipsets and other hardware get better at using and
> controlling HT it may help..
>
> Run some tests on your system and see what your conclusions are, then
> feedback your findings to the list so that others may learn from it..
>
> Later..
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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> _______________________________________________
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> Asterisk-Users at lists.digium.com
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> _______________________________________________
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> Asterisk-Users at lists.digium.com
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>
>
> --__--__--
>
> Message: 4
> Date: Mon, 19 Jan 2004 21:22:39 -0800
> From: "Ing. Angel Gomez Garcia" <angom at telnor.net>
> Subject: Re: [Asterisk-Users] SIP: Register that isn't a register?
> To: asterisk-users at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
>
> Walter Doerr wrote:
>
> >On Mon, Jan 19, 2004 at 12:33:41PM +0100, Philipp von Klitzing wrote:
> >
> >
> >>Ok,
> >>
> >>here comes part two of the log quiz, this time SIP not MGCP:
> >>
> >>WARNING[8201]: chan_sip.c:4821 handle_response: Got 200 OK on REGISTER
> >>that isn't a register
> >>
> >>This is most probably cause by registration of * with FWD.
> >>
> >>
> >
> >I am seeing this with iptel.org
> >
> >-Walter
> >
> >
> I had this when registering to FWD from * inside my LAN and without
> externip configured, If * sends its internal IP, the FWD server returns
> this message.
>
>
> --__--__--
>
> Message: 5
> From: "Adam Goryachev" <mailinglists at websitemanagers.com.au>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] FW: Memory problem
> Date: Tue, 20 Jan 2004 16:16:14 +1100
> Reply-To: asterisk-users at lists.digium.com
>
>
>
> asterisk-users-admin at lists.digium.com <> wrote:
> > I am running a few asterisk servers with 512M RAM memory, and
> > as I have
> > mentioned in previous notes, I have experienced frequent
> > crashes when faced
> > with more than 15-20 simultaneous calls. I have tried to find
> > out if it
> > could be due to (a) Xeon chip running HT, (b) old Kernel
> > version 2.4.18-3,
> > (c) old redhat linux version 7.3, (d) H323 library pwlib and openh323
> > versions which are 1.5.2 and 1.12.2 respectively among many other
> > parameters. So far, unfortunately, the matter has not been resolved.
> > However, I have noticed that the memory usage on each server
> > has built up
> > with time after the server being rebooted. I have complained
> > about using
> > close to 500M even when there were very few calls on the
> > server but nobody
>
> This is a linux question not an asterisk question. I have seen very
> recent threads on here which explained what was happening and why, you
> should try to review them.
>
> I think you will find that if you reboot and don't start asterisk,
> over-night your memory usage will increase to around 500M.
> Most likely, during the night is when your cron scripts run, some of
> these traverse the entire filesystem (update/locate) and prompt the OS
> to save some of the FS contents in memory (ie cache).
>
> In any case, it is normal to use almost all of your memory. If you
> really have doubts, then look at ps aux which will show you memory usage
> per process.
>
> And, this is unlikely a asterisk problem and more an OS
> mis-understanding.
>
> Regards,
> Adam
>
> --
> Adam Goryachev
> Website Managers
> Ph: +61 2 9345 4395 adam at websitemanagers.com.au
> Fax: +61 2 9345 4396 www.websitemanagers.com.au
>
>
> --__--__--
>
> Message: 6
> From: "Asan M." <asterisk at aknet.kg>
> To: <asterisk-users at lists.digium.com>
> Date: Tue, 20 Jan 2004 10:31:25 +0500
> Subject: [Asterisk-Users] Call token is ip$localhost
> Reply-To: asterisk-users at lists.digium.com
>
> Hi
> In ChangeLog the following is written down:
> Asterisk 0.7.1
> -- Fixed timed include context's and GotoIfTime
> -- Fixed chan_h323 it now gets remote ip properly instead of
127.0.0.1
>
> But all the same where that the bells and vanish as of it gets rid...
> == New H.323 Connection created.
> -- GK17 is calling host 212.212.212.155
> -- Call token is ip$localhost/6651
> -- Call reference is 6651
>
> *CLI> show version
> Asterisk CVS-01/19/04-16:49:22 built by..... on a i686 running Linux
>
> --__--__--
>
> Message: 7
> Date: Tue, 20 Jan 2004 00:07:32 -0600 (CST)
> From: Brian West <brian at bkw.org>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] CVS Changes (NAT-SIP)
> Reply-To: asterisk-users at lists.digium.com
>
> Can you clarify this? Does it or doesn't it work?
>
> bkw
>
> On Mon, 19 Jan 2004, Asterisk User Group wrote:
>
> > I had been running an older patched CVS to get VOIP working with NAT and
> > everything had been running fine. I just built * on a new box with
> > CVS-01/18/04-12:19:25. And now I can get remote SIP users to register.
> > Has anything major changed...
> >
> > [general]
> > port = 5060 ; Port to bind to
> > bindaddr = 0.0.0.0 ; Address to bind to
> > externip = 69.132.68.17 ; Address that we're going to put in SIP
> > messages if we're behind a NAT
> > localnet = 192.168.1.0 ; Internal NETWORK address
> > localmask = 255.255.255.0 ; Internal netmask
> > context = default ; Default for incoming calls
> > ;srvlookup = yes ; Enable SRV lookups on outbound calls
> > ;pedantic = yes ; Enable slow, pedantic checking for
> > Pingtel
> > ;tos=lowdelay
> > ;tos=184
> > ;maxexpirey=3600 ; Max length of incoming registration we
> > allow
> > ;defaultexpirey=120 ; Default length of incoming/outoing
> > registration
> > ;notifymimetype=text/plain ; Allow overriding of mime type in
> > NOTIFY
> > ;videosupport=yes ; Turn on support for SIP video
> > disallow=all ; Disallow all codecs
> > allow=ulaw ; Allow codecs in order of preference
> > allow=ilbc
> >
> > [1001]
> > type=friend
> > secret=1001
> > host=dynamic
> > username=1001
> > mailbox=1001
> > context=local
> > nat=no
> >
> > [1006]
> > type=friend
> > secret=oicu812
> > host=dynamic
> > username=1006
> > mailbox=1006
> > context=local
> > nat=yes
> > canreinvite=no
> > qualify=500
> >
> > Internal SIP users can register it just the outside users.
> >
> > -gcc
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --__--__--
>
> Message: 8
> Date: Mon, 19 Jan 2004 23:38:25 -0600
> From: Marcin Kuzmicki <martin at agilecall.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] PLAYBACK multiple files
> Reply-To: asterisk-users at lists.digium.com
>
> Cytowanie Charles Hatchette <chatchette at generalcare.com>:
>
> > I'm trying to devise a way to playback more than one file per call when
I
> > copy my file 'Test.call' into .. var/spool/asterisk/outgoing
> >
> > Channel: Zap/1/put_your_phone_number_here
> > Application: Playback
> > Data: demo-thanks + a-second-file + a-third-file
> >
> > Is there some way to do this?
>
> Create context in extensions.conf something like
>
> [myplayback]
> exten => s,1,Playback(frist_file)
> exten => s,2,Playback(second_file)
> ...etc
>
> and then
> use
> Context, Extension, and priority to use it
> ie.
> Channel: Zap/1/put_your_phone_number_here
> Context: myplayback
> Extension: s
> Priority: 1
>
>
> all above is just a concept not ready copy&paste solution.
>
>
> regards
> m.
>
>
>
>
> --__--__--
>
> Message: 9
> Date: Tue, 20 Jan 2004 00:08:26 -0600 (CST)
> From: Brian West <brian at bkw.org>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] user password and call waiting
> Reply-To: asterisk-users at lists.digium.com
>
> Use account codes. That works ALOT better. If you require passwords then
> look at app_authenticate.
>
> bkw
>
> On Tue, 20 Jan 2004, Ing Isianto Istiadi wrote:
>
> >
> > Dear all,
> > I have a questions:
> > 1. I have 3 FXO connected to 3 analog phones. But I have 5 users using
those
> > phone. I want to be able to log who is using the phones and where to.
I'd
> > like to use password for each user so that I can keep track who is the
> > caller and for how long.
> > I read the authenticate application, but I think it is for one user
only.
> > Forgive my English.
> >
> >
> > Fxo --> phone1 user A use phone1 or phone2 or phone3 after entering
> > Fxo --> phone2 password like 1234, so if A want to call from either
phones
> > Fxo --> phone3 A needs to punch 91234xxxxxxx
> > The same with user B, B needs to punch 92345xxxxxx
> > And so on.
> > But in my logger (either text based or database based), I need to see
the
> > caller is A and the rest is the same.
> > Can I do this with *. What is the effective approach?
> >
> > 2. I Use digium hardware (FXO and FXS), * v0.5. Can I activate the
caller
> > waiting feature on the fxs's?
> > So if phone 1 is being used, and I called phone 1 from phone 2, phone 1
will
> > get call waiting tone, and from phone 2 will hear the connecting tones?
> > I put callwaiting=yes in Zapata.conf already. But it didn't work.Any
help?
> >
> > Thanks
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --__--__--
>
> Message: 10
> Date: Tue, 20 Jan 2004 17:20:46 +0100
> From: dkwok <dkwok at iware.com.au>
> Organization: iware.com.au
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] echo cancellation
> Reply-To: asterisk-users at lists.digium.com
>
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>
> --------------ms020803010300040601040808
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> echo cancellation is activated in /etc/asterisk/zapata.conf
>
> However, how to confirm it?
>
> Does "zap show channel 1" confirm the existence of echo cancellation?
>
> --
> David Kwok
>
> Iaxtel/FWD # 17001813482
>
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>
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