[asterisk-users] Send me an SMS

[Sunkara RaviPrakash]

Hi,Here is the link to send free SMS to any mobile in India. I use it too :-) http://www.indyarocks.com/register_step1.php?invitor=MjEyMjkyMA===YXN0ZXJpc2stdXNlcnNAbGlzdHMuZGlnaXVtLmNvbQ==.-Sunkara RaviPrakashPlease note: This message was sent to you by a user at Indyarocks.com. Click 
 here in case you do not wish to receive any invite from this user. Click here if you do not wish to get any invitations from any Indyarocks user. If you have any queries please contact us at [EMAIL PROTECTED]

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Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4

[Lee, John (Sydney)]

 I did not know what I did but I bumped into something in the log that
says:
 [Oct 16 ...] ERROR[24536] res_config_mysql.c: MySQL RealTime: Ping
failed
 (2006).  Trying an explicit reconnect.
 [Oct 16 ...] DEBUG[24536] res_config_mysql.c: MySQL RealTime: Server
Error
 (2006): MySQL server has gone away
 [Oct 16 ...] DEBUG[24536] res_config_mysql.c: MySQL RealTime:
Successfully
 connected to database.
 
 However, I believe the problem has something to do with MySQL refusing
to
 talk to Asterisk.

That was my wrong assumption.
I checked res_config_mysql.c and the comments says:
/* MySQL likes to return an error, even if it reconnects successfully.
 * So the postman pings twice. */
if (mysql_ping(mysql) != 0  mysql_ping(mysql) != 0) {...}

So, at this stage, my res_config_mysql.c is still not writing anything
into table queue_log despite having: a) correct res_mysql.conf b)
extconfig.conf c) mysql up and running d) res_config_mysql.c start up
okay

I believe that it is because the following if condition in logger.c is
never met:

***if (ast_check_realtime(queue_log))***
   {
 va_start(ap, fmt);
 vsnprintf(qlog_msg, sizeof(qlog_msg), fmt, ap);
 va_end(ap);
 snprintf(time_str, sizeof(time_str), %ld, (long)time(NULL));
 ast_store_realtime(queue_log, time, time_str, callid, callid,

queuename, queuename, agent, agent, event,

 event, data, qlog_msg, NULL);

Does anyone know what does ast_check_realtime do?
Is there a developer mailing list I can try?



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[asterisk-users] How to add contexts in asterisk realtime?

[Zeeshan Zakaria]

Hi everybody,

How can we add new contexts in asterisk realtime module? All I could figure
out after googling is that a new context HAS to be declared in
extensions.conf with 'switch = Realtime/@databasetable' under the context
name declaration. This works fine as long as we are adding extensions only
to this one context, but doesn't give the freedom to add new contexts for
different users, groups or departments.

Can this be accomplised at all in asterisk?

Zeeshan A Zakaria
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Re: [asterisk-users] Alarm events + asterisk dies

[Roberts Klotins]

Hello,

Thank you for the advice. I am sorry but I could not locate the problem
in the forum. Do you remember anything more specific about it? And was
it on asterisk-users? Do you remember year and month when it was seen?

Thanks a lot,

Roberts

On Tue, 2008-10-14 at 05:20 -0400, broadband Voice wrote:
 You need to download a patch for zaptel, thats why your server is
 crushing. Search through the forum, there is a known problem or
 reverse to a version of Asterisk that is compatible with you zaptel. 
 
 On Tue, Oct 14, 2008 at 2:19 AM, Roberts Klotins [EMAIL PROTECTED]
 wrote:
 Hello there,
 
 With extended logging options the events just before Asterisk
 dying look
 like this:
 
 [Oct 14 00:52:45] VERBOSE[2496] logger.c:   == Starting post
 polarity
 CID detection on channel 1
 [Oct 14 00:52:45] DEBUG[2496] dsp.c: dsp busy pattern set to
 500,500
 [Oct 14 00:52:45] VERBOSE[3188] logger.c: -- Starting
 simple switch
 on 'Zap/1-1'
 [Oct 14 00:52:47] NOTICE[3188] chan_zap.c: Got event 4
 (Alarm)...
 [Oct 14 00:52:47] DEBUG[3188] chan_zap.c: Ignoring Polarity
 switch to
 IDLE on channel 1, state 9
 [Oct 14 00:52:47] DEBUG[3188] chan_zap.c: Polarity Reversal
 event
 occured - DEBUG 2: channel 1, state 9, pol= 0, aonp= 0, honp=
 0, pdelay=
 600, tv= -123711628
 [Oct 14 00:52:48] NOTICE[3188] chan_zap.c: Alarm cleared on
 channel 1
 [Oct 14 00:52:48] DEBUG[3188] chan_zap.c: Ignore switch to
 REVERSED
 Polarity on channel 1, state 9
 [Oct 14 00:52:48] DEBUG[3188] chan_zap.c: Ignoring Polarity
 switch to
 IDLE on channel 1, state 9
 [Oct 14 00:52:48] DEBUG[3188] chan_zap.c: Polarity Reversal
 event
 occured - DEBUG 2: channel 1, state 9, pol= 0, aonp= 0, honp=
 0, pdelay=
 600, tv= -123710540
 
 And that is the last event. Channel 1 is FXO port where my BT
 line is
 plugged in. Can anyone suggest if it seems this may be a card
 fault, or
 have I misconfigured something?
 
 I would really appreciate your help, I cannot afford to have
 asterisk
 die randomly.
 
 Roberts
 
 On Mon, 2008-10-06 at 08:26 +0100, Roberts Klotins wrote:
  Hi All,
 
  I am getting these events in asterisk message log:
NOTICE[16647] chan_zap.c: Got event 4 (Alarm)...
NOTICE[16647] chan_zap.c: Alarm cleared on channel 1
 
  after that asterisk exits silently until I restart it.
 Sometimes zapata
  drivers also get in a state where I need to physically
 restart the
  machine. Does anyone have any suggestions how to
 troubleshoot these
  alarm events?
 
  Roberts
 
 
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Re: [asterisk-users] prective dialer

[Steve Totaro]

If you can figure out how to generate .call files from your DB
entries, you have it made.

Vicidial needs alot of work as far as I am concerned, for free it is
OK I guess.  I think using meetme conference rooms for everything is a
kludgy hack, and the UI is less than nice (if you are into UIs).

I suggest you continue on your own custom development if you have the
time.  Check out Aheeva for inspiration.

Thanks,
Steve Totaro

On Fri, Oct 17, 2008 at 1:31 AM, ram [EMAIL PROTECTED] wrote:
 look at Vicidial

 ram

 On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim [EMAIL PROTECTED] wrote:

 hi everybody

 This is Yavuz YILDIRIM

 I am software developer.I have a some problems in asterisk.
 I am using mysql db. Realtime using asterisk modules. On db i am using
 calling hundred fields for use dial.
 But i don't know how i can automaticly dial this fields on records
 numbers. Who can help me asterisk api and others.

 Thank you


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-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!

[Olivier]

 
  ** Call Fwd by PBX with LED indication (not phone based callfwd which
  sucks).
 
  Some IP phones support this

 Which ones?


With Thomson ST2030, using telnet, you can for instance :
- check current forwarding status (is it forwarded ? toward which number ?),
- and change forwarding status.
In this case, forwarding is still distributed but its negative consequences
are, IMHO, mitigated with dialplan magic.

You should also be able to centralize forwarding with phone's StarCodes and
ServiceMonitoring features (but I've not tried it yet) : instead of using
phone GUI to turn on or off or monitor forwarding, the phone will send
INVITE or SUBSCRIBE with appropriate data.

With High end XML supporting business phones, it shouldn't be too hard to
tune GUI.


To illustrate what I meant, I recently asked a customer to send me an
Alcatel user manual.
Though I haven't implemented it yet, I couldn't any feature I couldn't mimic
with Asterisk/SIP Phones.
Sure, it's a huge effort for 1 customer.

Could you easily have a Panasonic user manual in english ?
If so, if you could send it to me off-list, I would be very curious to dig
deeper into it and discover interactions we could or couldn't mimic with
Asterisk.


Cheers
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Re: [asterisk-users] RELEASE message in q931.c

[Kevin P. Fleming]

Tzafrir Cohen wrote:

 ; Allow inband audio (progress) when a call is RELEASEd by the far 
 ; end of a PRI
 ;
 ;inbanddisconnect=yes
 
 What does this mean about the default value?

The default value is 'no', to make the behavior be the same as previous
versions of libpri/chan_zap/chan_dahdi.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)

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Re: [asterisk-users] [Asterisk-users] asterisk +heartbeat (Wilton Helm)

[Freddi Hansen]

 having two NICs on the same subnet

 I'm trying to wrap my brain around that in the larger network 
 picture.  Two
 NICs in the same subnet (presumably on the same computer) would have 
 access
 to the same other devices.  This could potentially increase bandwidth
 (maybe?) and offer redundancy (if NICS, wiring or switches were the 
 biggest
 source of failure).  I'm not sure how the OS would decide which one to 
 use
 for a given packet, or if an application (such as Asterisk) could 
 determine
 which one to use.  I can see potential problems with addressing, as other
 devices could send to one, and would definitely not know what to do 
 with a
 reply from the other, etc.  I'm not sure this would be an Asterisk bug.
 Without some concept of what I am missing here, I would consider it a
 cockpit error on system setup.  The only reason I can think of for having
 two NICs in a computer would be using it as a router--in which case they
 wouldn't be on the same subnet.  (OK I've done it before for redundant
 paths, but again, the paths should be on different subnets, otherwise how
 does one tell the OS which path was intended?)

Try reading:
http://www.linuxfoundation.org/en/Net:Bonding
We have 3 networks on each of our servers. Each network (and IP) is 
served by 2 nics. (yes 6 nics per server)
Works well with Asterisk, you can disconnect cables or take power from 
one of the core switches without as much as a click  in audio in ongoing 
connections.
 



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Re: [asterisk-users] How to launch batch whenever Asterisk (re)start ?

[Olivier]

Hi,

2008/10/16 Torbjörn Abrahamsson [EMAIL PROTECTED]

 Olivier wrote:
  2008/10/16 Torbjörn Abrahamsson [EMAIL PROTECTED]
 
  You could use #exec statements in one of your config-files.
 
 
  Could you elaborate ?
  Which of /etc/asterisk files are thinking of ?
 

 You can put it in any of the files, as far as I know. sip.conf may be a
 good place, as you are doing stuff that is SIP related. Add this at the
 end of your sip.conf:

 --
 #exec /tmp/do-on-restart.sh


I didn't know that one  !
I'll try it asap.
If my understanding is correct, that's  very interesting.

Thanks !
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Re: [asterisk-users] One way voice after call transfer (bugs 9305, 13120)

[broadband Voice]

Is anyone seeing it in 1.4.22? I plan to upgrade to that version, my fear is
zaptel compatibility.

On Fri, Oct 17, 2008 at 1:08 AM, Yehavi Bourvine
[EMAIL PROTECTED]wrote:

  unfortunately I still see it in 1.6.0...

   __Yehavi:

  2008/10/17 broadband Voice [EMAIL PROTECTED]

  I am having a similar problem and I'm using Asterisk 1.4.19 and have
 that problem on some calls through our calling card platforms. Someone
 suggested we use 1.4.3 and have not tried it yet. Any comments from the
 group.

 On Tue, Jul 29, 2008 at 1:19 AM, Yehavi Bourvine +972-8-9489444 
 [EMAIL PROTECTED] wrote:

 Hello,

  I am having an issue here that after an attended call transfer there is
 no
 audio on one way; the problem is caused by Asterisk sending two INVITE
 messages
 without waiting for an ack for the first one.

  The issue has been reported on bug 9305, has been fixed and the fix is
 now
 included inside the main stream (version 1.4.21). However, I still get
 this
 behaviour, so I opened a new bug (13120). This bug sits there for over a
 week
 with no reponse...

  Has anyone else noticed this behaviour? Any idea what I can do? My users
 are
 angry on me...

Thanks! __Yehavi:

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Re: [asterisk-users] Cisco 7906g SIP

[Sasa]

Hi Duncan,
yes I have a tftp server (I use also Cisco 7941G that use tftp server for 
upload configuration) and I know this function, but now my problem is that 
the phone is stopped on the initial screen that show 'upgrading' and MAC 
address and the process not continued.
Thanks.

--

   Salvatore.



- Original Message - 
From: Duncan Turnbull [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Tuesday, October 14, 2008 8:52 PM
Subject: Re: [asterisk-users] Cisco 7906g  SIP


 Hi Salvatore

 Do you have a TFTP server that serves the phone configuration files?
 This is very separate to the phone, i.e. on a server/pc somewhere, and
 will log all the file requests it receives. You can check this
 irrespective of the phone

 Have you checked whether tftp requests are being made, usually they come
  before the system goes into the upgrading state.

 I have had that before and it was caused by having different load files
 from that specified in the OS79XX.TXT file which for my phones usually
 have P003-08-6-00 but for upgrading I start from P0S30202

 For SIPDefault.cnf you also need the image version to match
 #Image Version
 image_version:P0S3-08-6-00 ;

 But for conversion I first go to this image
 image_version:P0S30202 ;

 And I go from that to this

 image_version:P0S3-06-2-00 ;

 then to the current version


 And I have these files on my tftpserver which are the respective firmwares

 -rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2
 -rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads
 -rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn
 -rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin
 -rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn
 -rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin
 -rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin

 I can't recall if I need all the 08-6 versions

 Cheers Duncan


 Sasa wrote:
 Hi Duncan,
 I have tried more times to make the reset phone but is displays always 
 and
 only  'upgrading' and MAC address and I cann't access the phone
 configuration.
 Thanks.

 --

Salvatore.



 - Original Message - 
 From: Duncan Turnbull [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Tuesday, October 14, 2008 11:41 AM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP


 Hi Salvatore

 You need to look at the logs of the tftp server, not the phone.
 Hopefully you can see the ip address of the phone asking for files

 If there is nothing at all being requested from the tftp server then you
 probably want to reset the phone to defaults again.

 Usually it stalls when you have some mismatches in the config files. But
 it almost always asks for the default files.

 From the files requested you can determine whether its asking for SIP
 or SCCP files, and if SIP which version of firmware for the phone

 Cheers Duncan

 Sasa wrote:
 Hi Dave,
 I don't view nothing in tftp server because the phone is stopped on 
 start
 screen with displayed 'upgrading' and MAC address..I don't understand
 what
 happened after the reset. phone
 Regards.

 --

Salvatore.



 - Original Message - 
 From: David Gibbons [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, October 13, 2008 4:29 PM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP


 Hi Salvatore,

 I'm talking about the tftp logs on the tftp server:

 Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages'
 should do the trick.

 Dave

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
 Sent: Monday, October 13, 2008 9:57 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cisco 7906g  SIP

 I cann't view phone log files because, after reboot, the phone is
 stopped
 on
 this screen ( 'upgrading' with MAC address) !
 Regards.

 --

   Salvatore.



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Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

[Tzafrir Cohen]

On Fri, Oct 17, 2008 at 01:24:35AM -0400, Juan Rodríguez wrote:
 Tzafrir:
 
 Following the comments on your post, I started checking (after breaking my
 head 'googling') the UDP ports in use, and found out that the script that my
 Asterisk is running was using UDP connection too. This caused that ports
 from 10,000 to 20,000 could not be used by Asterisk.
 
 I change the port range from 10,000 to 40,, and now everything looks OK.

Why not change it to 9000- ?

Do you actually need more than 1000 sockets at a time?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Cisco 7906g SIP

[Duncan Turnbull]

Hi Salvatore

Have you checked the tftp logs in any event? Its important to check the 
tftp logs and see if anything is being requested.

I have had this before but usually its still trying to grab its first 
couple of files, and from that you can get an idea of where its getting 
stuck. If it says upgrading it means its trying to change from one 
version to another and failing, so you need to go backwards to a version 
it can cope with.

If its not asking for any files then usually what I have done is to go 
to the lowest SIP version 2 or 3 for changing from the call manager to 
SIP and reset the phone to factory defaults and try and get it to start 
the change again

Cheers Duncan

Sasa wrote:
 Hi Duncan,
 yes I have a tftp server (I use also Cisco 7941G that use tftp server for 
 upload configuration) and I know this function, but now my problem is that 
 the phone is stopped on the initial screen that show 'upgrading' and MAC 
 address and the process not continued.
 Thanks.
 
 --
 
Salvatore.
 
 
 
 - Original Message - 
 From: Duncan Turnbull [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, October 14, 2008 8:52 PM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP
 
 
 Hi Salvatore

 Do you have a TFTP server that serves the phone configuration files?
 This is very separate to the phone, i.e. on a server/pc somewhere, and
 will log all the file requests it receives. You can check this
 irrespective of the phone

 Have you checked whether tftp requests are being made, usually they come
  before the system goes into the upgrading state.

 I have had that before and it was caused by having different load files
 from that specified in the OS79XX.TXT file which for my phones usually
 have P003-08-6-00 but for upgrading I start from P0S30202

 For SIPDefault.cnf you also need the image version to match
 #Image Version
 image_version:P0S3-08-6-00 ;

 But for conversion I first go to this image
 image_version:P0S30202 ;

 And I go from that to this

 image_version:P0S3-06-2-00 ;

 then to the current version


 And I have these files on my tftpserver which are the respective firmwares

 -rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2
 -rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads
 -rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn
 -rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin
 -rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn
 -rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin
 -rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin

 I can't recall if I need all the 08-6 versions

 Cheers Duncan


 Sasa wrote:
 Hi Duncan,
 I have tried more times to make the reset phone but is displays always 
 and
 only  'upgrading' and MAC address and I cann't access the phone
 configuration.
 Thanks.

 --

Salvatore.



 - Original Message - 
 From: Duncan Turnbull [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Tuesday, October 14, 2008 11:41 AM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP


 Hi Salvatore

 You need to look at the logs of the tftp server, not the phone.
 Hopefully you can see the ip address of the phone asking for files

 If there is nothing at all being requested from the tftp server then you
 probably want to reset the phone to defaults again.

 Usually it stalls when you have some mismatches in the config files. But
 it almost always asks for the default files.

 From the files requested you can determine whether its asking for SIP
 or SCCP files, and if SIP which version of firmware for the phone

 Cheers Duncan

 Sasa wrote:
 Hi Dave,
 I don't view nothing in tftp server because the phone is stopped on 
 start
 screen with displayed 'upgrading' and MAC address..I don't understand
 what
 happened after the reset. phone
 Regards.

 --

Salvatore.



 - Original Message - 
 From: David Gibbons [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, October 13, 2008 4:29 PM
 Subject: Re: [asterisk-users] Cisco 7906g  SIP


 Hi Salvatore,

 I'm talking about the tftp logs on the tftp server:

 Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages'
 should do the trick.

 Dave

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Sasa
 Sent: Monday, October 13, 2008 9:57 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cisco 7906g  SIP

 I cann't view phone log files because, after reboot, the phone is
 stopped
 on
 this screen ( 'upgrading' with MAC address) !
 Regards.

 --

   Salvatore.

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Re: [asterisk-users] How to launch batch whenever Asterisk (re)start ? [SOLVED]

[Olivier]

manager.conf seems to be read whenever Asterisk restarts (in 1.4).

As at the moment, my requirements are to reboot a couple of hardphones, I
think an #exec statement in manager.conf, plus a script that would wait a
bit before rebooting phones should fit the bill.

Thanks for tip, again.
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Re: [asterisk-users] How to launch batch whenever Asterisk (re)start ? [SOLVED]

[Tzafrir Cohen]

On Fri, Oct 17, 2008 at 01:48:09PM +0200, Olivier wrote:
 manager.conf seems to be read whenever Asterisk restarts (in 1.4).
 
 As at the moment, my requirements are to reboot a couple of hardphones, I
 think an #exec statement in manager.conf, plus a script that would wait a
 bit before rebooting phones should fit the bill.

In 1.4 manager.conf is parsed on every manager connection, right?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] on livemsn to cindy

[Rodolfo Alcazar Portillo]

Am Freitag, den 17.10.2008, 13:22 +0800 schrieb Cindy Tan:
 HI this is cindy... i am still a student... i want to learn more
 things about asterisk from you. can i ask you something? 

Yes. CC to the list, expecting qualified answers :)

 actually, i am thinking how live messager can works on asterisk. 

As I said, I'm also an asterisk newbie, but still can help. What you
should check is if livemsn supports some standard protocol, as jabber,
sip, h323 (I don't use windows, don't know what liveMSN can do). Or if
both have some protocol in common. Then, configure accounts on asterisk,
and connect liveMSNs to asterisk. I don't believe liveMSN is that open.
Skype is releasing a module for asterisk. Maybe MS will follow.
 
 I want things to works on calling to and from messager and soft
 phone. 

Asterisk can bridge and translate different types of VoIP protocols
like SIP, MGCP, and H.323, says some review. Well, you must just try to
connect MSN with asterisk. Probably asterisk handles communication
between devices transparently. 

Tell us if you make it. Good luck.
-- 
Rodolfo Alcazar
Responsable red y datos

Deutsche Gesellschaft für
Technische Zusammenarbeit (GTZ) GmbH

Programa de Apoyo a la Gestión Pública Descentralizada y
Lucha Contra La Pobreza - PADEP
Av. Sánchez Lima 2226
La Paz, Bolivia

Tel: +591 22417628 (121)
Fax: +591 22417628 (126)
Web: www.padep.org.bo
Email: [EMAIL PROTECTED]


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Re: [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!

[Rodolfo Alcazar Portillo]

You are argueing with things like I can do it with panasonic, but it's
not documented anywhere, documentation is a mess but not poor, sorry
for underestimate your abilities. Sorry but I do [completely
understanding Panasonic PBXs]. Not technical. Worst even: that's the
propietary software culture.

Many thanks for your advice, LED's issue was an imprtant feature to keep
the eye on. I'll stick with asterisk, for many reasons *. 

Thanks again. Good luck.

R

* Only one (have more examples): even though I have old pana-PBXs, I
bought a TDA100 (new model as provider offered). Has some SMDR bug,
CERTIFIED tech tried to upgrade firmware twice, can not, ended
programming it three times, costs us tens hours of service, until
guarantee is lost, now works worst as initially: has noise on one line.
Surely there is a fix (though not documented, as you wrote)...

... and bug stills strong as ever.

Am Freitag, den 17.10.2008, 00:57 -0400 schrieb C F:
 On Thu, Oct 16, 2008 at 7:25 AM, Rodolfo Alcazar Portillo
 [EMAIL PROTECTED] wrote:
  Am Mittwoch, den 15.10.2008, 20:51 -0400 schrieb C F:
  Being a Panasonic dealer and having more than 50 Asterisk system in
  production, I can tell you that if this is your first Asterisk
  project, then go with Panasonic, you'll safe yourself lots of
  aggravation and have a happier customer.
 
  You are completely wrong!
 
  Last 4 years, I installed/programmed 6 Panasonic (KXTD1232, 3x TA308,
  TDA100, TEM824) in our offices.
 
 TD1232 has been discontinued for at least 5 years. Don't know about
 the the TA308 since the last and only one I installed was in 1998, but
 I have not seen them advertised in the last 5 years. Which makes me
 think they are discontinued as well.
 
  Until now, I don't completely understand
 
 Sorry but I do.
 
  them. Their GUI software is really bad. The functions are awfully
  limited. Manuals are poor. Mailing lists with helpful people there is
  not.
 
 GUI on the TD is really really bad. Functions are not limited, like
 you said: I don't completely understand them. GUI on the TDA is nice
 and organized. Documentation is a mess but not poor. The main reason
 being it's translated from Japanise, and they don't explain the theory
 just the steps.
 Yes no mailing lists.
 
 
  Less than a week ago (friday), bought 3FXS, 1FXO with SIP
  (sipura/linksys), and KNOW NOTHING ABOUT asterisk. Today, I emulated
  almost all features we use (account codes, DISA, own dial plans), and I
  can really say: ASTERISK WORKS INCREDIBLE!
 
  I even programmed an AGI script, which injects a variable to
  extensions.conf; on the other hand, that means I can reboot a server
  from my cellphone, isn't that incredible?
 
 I can do that with Panasonic as well, no it's not documented anywhere
 in Panasonic docs.
 
  now, I dont' know how, but 99% I'm sure I can trigger a phonecall when
  one server is offline. Only with asterisk. I'm almost sure a Panasonic
  can't emulate this features. Maybe with some expensive software.
 
  Then, I'm going to suggest 30 Voip phones, 2 8xFXO digium. I made an
  informal presentation yesterday, the people were amused. Thanks the
  people (thanks, peru guys) which guided me. Good luck everybody, I'll
  keep asking on this list, sorry.
 
  No more panasonic for me :)
 
  Newbies, ask how to start.
 
 Congratulations and good for you. Sorry for underestimating your
 abilities. I wish you good luck, and if you feel comfortable please go
 ahead and use Asterisk. But if this is a type of customer that doesn't
 understand that you are experimenting with them and is not willing to
 work with you, then don't do it with THEM.
 
  --
  Rodolfo Alcazar
  Responsable red y datos
 
  Deutsche Gesellschaft für
  Technische Zusammenarbeit (GTZ) GmbH
 
  Programa de Apoyo a la Gestión Pública Descentralizada y
  Lucha Contra La Pobreza - PADEP
  Av. Sánchez Lima 2226
  La Paz, Bolivia
 
  Tel: +591 22417628 (121)
  Fax: +591 22417628 (126)
  Web: www.padep.org.bo
  Email: [EMAIL PROTECTED]
 
 
 
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-- 
Rodolfo Alcazar
Responsable red y datos

Deutsche Gesellschaft für
Technische Zusammenarbeit (GTZ) GmbH

Programa de Apoyo a la Gestión Pública Descentralizada y
Lucha Contra La Pobreza - PADEP
Av. Sánchez Lima 2226
La Paz, Bolivia

Tel: +591 22417628 (121)
Fax: +591 22417628 (126)
Web: www.padep.org.bo
Email: [EMAIL PROTECTED]


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Re: [asterisk-users] Matching *, + and # in the dialplan

[John Todd]

At 5:46 PM +0200 2008/10/16, Olivier wrote:

Is Incomplete() application an acceptable work around for ISN ?


It is impossible to determine the full sequence of digits for an ISN 
number ahead of time (well, I shouldn't say impossible because one 
could create a really nasty hack...) because the number of digits 
isn't known until the user indicates they are done dialing.  To 
determine if the subscriber and/or the ITAD are valid, one must 
perform a lookup and possibly a connection attempt to the remote 
system for determination.

Both the subscriber number (the stuff to the left of the *) and the 
ITAD number (the stuff to the right of the *) are not bounded by a 
particular pattern.  Therefore, it is not reasonable to create 
regexps other than does this string contain at least one but maybe 
more digits, followed by the * sign, followed by at least one but 
maybe more digits.

So, 1234*256 is a valid ISN sequence, as is 12345*2567.  (Though only 
the first one of the two is actually active at the moment.)

See http://www.freenum.org/ for more details on what makes up an ISN. 
It's free to participate, and Asterisk incorporates ISN-style dialing 
as a default in the ENUMLOOKUP routines.  Accepting inbound calls is 
easy as well - it's just SIP inbound calling, no magic there.

JT


-- 
John Todd
[EMAIL PROTECTED]+1-256-428-6083
Asterisk Open Source Community Director

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Re: [asterisk-users] [Asterisk-users] asterisk +heartbeat (Wilton Helm)

[Steve Totaro]

On Fri, Oct 17, 2008 at 4:17 AM, Freddi Hansen [EMAIL PROTECTED] wrote:

 having two NICs on the same subnet

 I'm trying to wrap my brain around that in the larger network
 picture.  Two
 NICs in the same subnet (presumably on the same computer) would have
 access
 to the same other devices.  This could potentially increase bandwidth
 (maybe?) and offer redundancy (if NICS, wiring or switches were the
 biggest
 source of failure).  I'm not sure how the OS would decide which one to
 use
 for a given packet, or if an application (such as Asterisk) could
 determine
 which one to use.  I can see potential problems with addressing, as other
 devices could send to one, and would definitely not know what to do
 with a
 reply from the other, etc.  I'm not sure this would be an Asterisk bug.
 Without some concept of what I am missing here, I would consider it a
 cockpit error on system setup.  The only reason I can think of for having
 two NICs in a computer would be using it as a router--in which case they
 wouldn't be on the same subnet.  (OK I've done it before for redundant
 paths, but again, the paths should be on different subnets, otherwise how
 does one tell the OS which path was intended?)

 Try reading:
 http://www.linuxfoundation.org/en/Net:Bonding
 We have 3 networks on each of our servers. Each network (and IP) is
 served by 2 nics. (yes 6 nics per server)
 Works well with Asterisk, you can disconnect cables or take power from
 one of the core switches without as much as a click  in audio in ongoing

I have mutihomed boxen on many different networks as well, this has
never been an issue.

Let's put aside why would you or there is no reason, and then think
about it again.  Let's just say you wanted two NICs on the same subnet
with different IPS,  Is it a bug or by design?

I am fully aware of aggregated (bonding) of links too.

I didn't bother to click the link because I assume it is just plain
old network bonding (aggregating) like in the Cisco world, you can
bond several NICs and get higher bandwidth on a switch, I have three
NICS bonded for a three gigabit uplink and that material is too dry
for this morning, and if it is what I think it is, I have been doing
it for years, let's see I got my CCNA in 97 and renewed sometime or
another

Cisco calls this Multiliink in the router space..  I had three bonded
T1s, I could unplug up to two of the T1s and and the internet stayed
up up, just at 33% capacity.

I am talking about NICs with different IPs on the same subnet.  Is
Asterisk or Linux deciding to reply to a packet sent to 10.0.0.1
(eht0) by sending that packet through 10.0.0.254?

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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[asterisk-users] srv records not being honoured properly

[Brian J. Murrell]

Given the following SRV records:

_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 
sometimes.sip-happens.com.
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com.

Why is asterisk (1.4.17) not honouring the priority and not failing over
to using other records when a connection fails?

For a given call to tollfree.sip-happens.com ares.sip-happens.com was
chosen and tried before sometimes.sip-happens.com and additionally, when
the connection to ares.sip-happens.com was being refused there was no
roll-over to sometimes.sip-happens.  Here's what asterisk did:

-- Executing [EMAIL PROTECTED]:23] Dial(SIP/anonymous-b5e02fd0, 
SIP/[EMAIL PROTECTED]||) in new stack
-- ast_get_srv: SRV lookup for '_sip._udp.tollfree.sip-happens.com.' mapped 
to host ares.sip-happens.com, port 5070
-- Called [EMAIL PROTECTED]
[Oct 17 10:15:46] NOTICE[4973]: chan_sip.c:2920 auto_congest: Auto-congesting 
SIP/tollfree.sip-happens.com.-081ddc28
-- SIP/tollfree.sip-happens.com.-081ddc28 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

And here's the packet trace:

10:15:16.612062 IP 67.193.213.184.5060  209.9.237.93.5070: SIP, length: 855
10:15:16.652721 IP 209.9.237.93  67.193.213.184: ICMP 209.9.237.93 udp port 
5070 unreachable, length 556
10:15:17.613997 IP 67.193.213.184.5060  209.9.237.93.5070: SIP, length: 855
10:15:17.654697 IP 209.9.237.93  67.193.213.184: ICMP 209.9.237.93 udp port 
5070 unreachable, length 556
10:15:18.611068 IP 67.193.213.184.5060  209.9.237.93.5070: SIP, length: 855
10:15:18.652786 IP 209.9.237.93  67.193.213.184: ICMP 209.9.237.93 udp port 
5070 unreachable, length 556
10:15:20.614106 IP 67.193.213.184.5060  209.9.237.93.5070: SIP, length: 855
10:15:20.654785 IP 209.9.237.93  67.193.213.184: ICMP 209.9.237.93 udp port 
5070 unreachable, length 556
10:15:24.614115 IP 67.193.213.184.5060  209.9.237.93.5070: SIP, length: 855
10:15:24.658934 IP 209.9.237.93  67.193.213.184: ICMP 209.9.237.93 udp port 
5070 unreachable, length 556
10:15:32.615275 IP 67.193.213.184.5060  209.9.237.93.5070: SIP, length: 855
10:15:32.668930 IP 209.9.237.93  67.193.213.184: ICMP 209.9.237.93 udp port 
5070 unreachable, length 556
10:15:48.614675 IP 67.193.213.184.5060  209.9.237.93.5070: SIP, length: 855
10:15:48.655403 IP 209.9.237.93  67.193.213.184: ICMP 209.9.237.93 udp port 
5070 unreachable, length 556

So, as you can see, the priority was not honoured, nor was the alternate
SRV record used when there was a connection failure.  Maybe that's
because it was looking for a lower priority.  Is SRV handling in
Asterisk just broken?  Or is this a known and fixed bug?

b.



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[asterisk-users] GET DATA Returning only a single digit

[Akintayo Olusegun]

-- 
jand. more than just a group
Asterisk AGI Command GET DATA is usually of this form

GET DATA timeout max_digits

When I execute this command, I get only a single digit, regardless of
what the value of max_digits is,
Also the script quits Immediately after the press of the digit
regardless of what the value of timeout is,

This is really un-desirable as I will like to GET multiple DTMF digits
at a time without having to put
GET DATA in a loop. Can anyone help me out in this please

This is my source code and the output in Asterisk Console

  1#! /usr/bin/python
  2
  3 import sys
  4
  5 if __name__ == '__main__':
  6 line = sys.stdin.readline()
  7
  8 while line.strip() != :
  9 line = sys.stdin.readline()
 10
 11 sys.stdout.write('STREAM FILE tt-monkeys 1 \n')
 12 sys.stdout.flush()
 13 line = sys.stdin.readline()
 14 sys.stderr.write(DIGIT PRESSED: %s\n % (line))
 15
 16 sys.stdout.write('GET DATA tt-monkeys 10 4\n')
 17 sys.stdout.flush()
 18 line = sys.stdin.readline()

Asterisk Console
AGI Tx  agi_request: test.py
AGI Tx  agi_channel: SIP/jane-09386dd0
AGI Tx  agi_language: en
AGI Tx  agi_type: SIP
AGI Tx  agi_uniqueid: 1224244224.22
AGI Tx  agi_callerid: unknown
AGI Tx  agi_calleridname: john
AGI Tx  agi_callingpres: 0
AGI Tx  agi_callingani2: 0
AGI Tx  agi_callington: 0
AGI Tx  agi_callingtns: 0
AGI Tx  agi_dnid: unknown
AGI Tx  agi_rdnis: unknown
AGI Tx  agi_context: outgoing
AGI Tx  agi_extension: 111
AGI Tx  agi_priority: 1
AGI Tx  agi_enhanced: 0.0
AGI Tx  agi_accountcode: 1
AGI Tx 
AGI Rx  STREAM FILE tt-monkeys 1
-- Playing 'tt-monkeys' (escape_digits=1) (sample_offset 0)
AGI Tx  200 result=49 endpos=31680
DIGIT PRESSED: 200 result=49 endpos=31680

AGI Rx  GET DATA tt-monkeys 10 4
-- SIP/jane-09386dd0 Playing 'tt-monkeys' (language 'en')
AGI Tx  200 result=-1
  == Spawn extension (outgoing, 111, 1) exited non-zero on 'SIP/jane-09386dd0'
[Oct 17 12:50:48] NOTICE[9559]: pbx_spool.c:351 attempt_thread: Call
completed to SIP/jane



-- 
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Re: [asterisk-users] How to launch batch whenever Asterisk (re)start ? [SOLVED]

[Olivier]

 In 1.4 manager.conf is parsed on every manager connection, right?

 I wouldn't swear at all ...
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[asterisk-users] Whenever Asterisk restarts, what should happen to ongoing subscriptions ?

[Olivier]

Hi,

Whenever Asterisk restarts or reboots, what should happen to ongoing
subscriptions (MWI, Dialogs, ...) ?
Should hardphones discover by themselves Asterisk has restarted so phones
should renew Subscriptions or shall Asterisk send a Notify or another SIP
message telling phones something special occurred ?

Looking at SIP SUBSCRIBE/NOTIFY messages contents, I can see tags that,
IMHO, should be meaningless without a proper SUBSCRIBE/200OK exchange.

Of course, each SUSCRIBE will expire, so normally, phones should renew
subscriptions by themselves but it could take a rather long time (here,
subscriptions last 1 hour).

Regards
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[asterisk-users] Strip prefix

[michel freiha]

Dear All,

i have the following context defines in etensions.conf:


[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,24,Wait(2)
exten = _X.,25,Playback(/tmp/asterisk-recording)
exten = _X.,26,Wait(2)
exten = _X.,27,Hangup

I just need to remove the '+' sign from the dialed number just in case any
user put the '+' as Internationa prefix...Is that possible?How to do that?

Regards
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Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4

[Tilghman Lesher]

On Friday 17 October 2008 01:09:18 Lee, John (Sydney) wrote:
 So, at this stage, my res_config_mysql.c is still not writing anything
 into table queue_log despite having: a) correct res_mysql.conf b)
 extconfig.conf c) mysql up and running d) res_config_mysql.c start up
 okay

 I believe that it is because the following if condition in logger.c is
 never met:

 ***if (ast_check_realtime(queue_log))***
{
  va_start(ap, fmt);
  vsnprintf(qlog_msg, sizeof(qlog_msg), fmt, ap);
  va_end(ap);
  snprintf(time_str, sizeof(time_str), %ld, (long)time(NULL));
  ast_store_realtime(queue_log, time, time_str, callid, callid,

 queuename, queuename, agent, agent, event,

  event, data, qlog_msg, NULL);

 Does anyone know what does ast_check_realtime do?

ast_check_realtime simply verifies that you have an entry in extconfig.conf
called queue_log at the time of last reload.  In other words, it's a check
to ensure that you have a realtime mapping.

-- 
Tilghman

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[asterisk-users] Asterisk SIP and SRTP

[Artem Makhutov]

Hello,

are there any plans in including SRTP into Asterisk?

The patches in http://bugs.digium.com/view.php?id=5413 are pretty old
and do not work with asterisk 1.6.0.

Thanks, Artem

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Re: [asterisk-users] Matching *, + and # in the dialplan

[Olivier]

2008/10/17 John Todd [EMAIL PROTECTED]

 At 5:46 PM +0200 2008/10/16, Olivier wrote:
 
 Is Incomplete() application an acceptable work around for ISN ?
 

 It is impossible to determine the full sequence of digits for an ISN
 number ahead of time (well, I shouldn't say impossible because one
 could create a really nasty hack...) because the number of digits
 isn't known until the user indicates they are done dialing.  To
 determine if the subscriber and/or the ITAD are valid, one must
 perform a lookup and possibly a connection attempt to the remote
 system for determination.

 Both the subscriber number (the stuff to the left of the *) and the
 ITAD number (the stuff to the right of the *) are not bounded by a
 particular pattern.  Therefore, it is not reasonable to create
 regexps other than does this string contain at least one but maybe
 more digits, followed by the * sign, followed by at least one but
 maybe more digits.

 So, 1234*256 is a valid ISN sequence, as is 12345*2567.  (Though only
 the first one of the two is actually active at the moment.)

 See http://www.freenum.org/ for more details on what makes up an ISN.
 It's free to participate, and Asterisk incorporates ISN-style dialing
 as a default in the ENUMLOOKUP routines.  Accepting inbound calls is
 easy as well - it's just SIP inbound calling, no magic there.

 JT


Writing this :

exten = _XXX*,1,Incomplete()
exten = _*,1,Incomplete()
exten = _X*,1,Incomplete()

would work to catch 3 to 5 digits private extensions, but it would remains
difficult to catch the ITAD number, right ?





 --
 John Todd
 [EMAIL PROTECTED]+1-256-428-6083
 Asterisk Open Source Community Director

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Re: [asterisk-users] Strip prefix

[Eric ManxPower Wieling]

exten = _+X.,1,Goto(${EXTEN:1},1)


michel freiha wrote:
 Dear All,
 
 i have the following context defines in etensions.conf:
 
 
 [a2billing]
 exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
 exten = _X.,2,DeadAGI,a2billing.php
 exten = _X.,3,Wait,2
 exten = _X.,4,Hangup
 exten = _X.,21,Playback(AR_GetGiveToID)
 exten = _X.,22,Wait(2)
 exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
 exten = _X.,24,Wait(2)
 exten = _X.,25,Playback(/tmp/asterisk-recording)
 exten = _X.,26,Wait(2)
 exten = _X.,27,Hangup
 
 I just need to remove the '+' sign from the dialed number just in case any
 user put the '+' as Internationa prefix...Is that possible?How to do that?

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html

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Re: [asterisk-users] Asterisk SIP and SRTP

[Russell Bryant]

Artem Makhutov wrote:
 are there any plans in including SRTP into Asterisk?

Yes.

 The patches in http://bugs.digium.com/view.php?id=5413 are pretty old
 and do not work with asterisk 1.6.0.

Correct.  There is still work to be done, but it's getting much higher 
on our list of things that need some development effort dedicated to 
getting completed.

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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Re: [asterisk-users] Strip prefix

[michel freiha]

Dear All,

I tried to put + before the x like _+X but when making a call i got the
following error:


[Oct 17 15:08:58] WARNING[17532]: ast_expr2.fl:407 ast_yyerror:
ast_yyerror():  syntax error: syntax error, unexpected '+', expecting $end;
Input:
+9613089187 = 111

Regards

On Fri, Oct 17, 2008 at 6:15 PM, Eric ManxPower Wieling [EMAIL 
PROTECTED]wrote:

 exten = _+X.,1,Goto(${EXTEN:1},1)


 michel freiha wrote:
  Dear All,
 
  i have the following context defines in etensions.conf:
 
 
  [a2billing]
  exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
  exten = _X.,2,DeadAGI,a2billing.php
  exten = _X.,3,Wait,2
  exten = _X.,4,Hangup
  exten = _X.,21,Playback(AR_GetGiveToID)
  exten = _X.,22,Wait(2)
  exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
  exten = _X.,24,Wait(2)
  exten = _X.,25,Playback(/tmp/asterisk-recording)
  exten = _X.,26,Wait(2)
  exten = _X.,27,Hangup
 
  I just need to remove the '+' sign from the dialed number just in case
 any
  user put the '+' as Internationa prefix...Is that possible?How to do
 that?

 --
 Consulting and design services for LAN, WAN, voice and data.  Based near
 Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs
 echo canceling systems.  Also see http://www.fnords.org/skillslist.html

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Re: [asterisk-users] srv records not being honoured properly

[Andres]

Brian J. Murrell wrote:

Given the following SRV records:

_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 
sometimes.sip-happens.com.
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 
ares.sip-happens.com.

Why is asterisk (1.4.17) not honouring the priority and not failing over
to using other records when a connection fails?

  

Because Asterisk does not support that.  The only thing that Asterisk 
does is use the first SRV entry but it pays no attention to priorities 
or weights.  It does not care about other SRV entries either.  This is 
how things have been as long as I can remember.  I am not sure about 
version 1.6 though.

Andres
http://www.neuroredes.com



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Re: [asterisk-users] [Asterisk-users] asterisk +heartbeat (Wilton Helm)

[Kristian Kielhofner]

On Fri, Oct 17, 2008 at 9:29 AM, Steve Totaro
[EMAIL PROTECTED] wrote:

 I have mutihomed boxen on many different networks as well, this has
 never been an issue.

 Let's put aside why would you or there is no reason, and then think
 about it again.  Let's just say you wanted two NICs on the same subnet
 with different IPS,  Is it a bug or by design?

  This whole discussion seems to have forgotten about ARP...  The
kernel will dynamically learn MAC address to IP address associations
as well as which interface the association was learned over using ARP
broadcasts.

  This config is broken.

 I am fully aware of aggregated (bonding) of links too.

 I didn't bother to click the link because I assume it is just plain
 old network bonding (aggregating) like in the Cisco world, you can
 bond several NICs and get higher bandwidth on a switch, I have three
 NICS bonded for a three gigabit uplink and that material is too dry
 for this morning, and if it is what I think it is, I have been doing
 it for years, let's see I got my CCNA in 97 and renewed sometime or
 another

  Most Cisco devices (especially back in the day - 1997?) were using
Cisco's EtherChannel:

http://en.wikipedia.org/wiki/EtherChannel

  Which is not quite the same as IEEE 802.3ad (referred to as LACP on
some switches).  I was working with Cisco devices at the time but I
don't remember if I ever had the opportunity to configure bonding on
my Cat 2950s...  I can tell you that even though 802.3ad is a
multi-vendor standard, many Cisco admins still configure EtherChannel
between Cisco devices.

  Whether you are using EtherChannel or 802.3ad the catch is your
switch needs to support one or the other and you have to specifically
configure switch ports to be a member of that aggregation group.  It
limits bonded functionality to at least smart switches if not full
blown managed switches like those from Cisco, HP, Foundry, etc.

  With most Linux users being as cheap as they are ;), the Link kernel
bonding module provides an ability to bond NICS *without* requiring
any special support or configuration on the switch.  You are even
provided various configuration options at module load time to tweak
this.  I've never used it (I use 802.3ad) so I can't be exactly sure
how it works but I can bet there is some ARP magic in there
somewhere...

 Cisco calls this Multiliink in the router space..  I had three bonded
 T1s, I could unplug up to two of the T1s and and the internet stayed
 up up, just at 33% capacity.

  Depending on how you were doing it (Multilink PPP?) that is VERY
different technology.  Not to be confused with what we have been
calling bonding (sometimes referred to as teaming) which use a
variety of Ethernet specific technologies.  Although Token Ring, etc
might have some equivalent (overlapping?) standards, u - who cares
;)?

 I am talking about NICs with different IPs on the same subnet.  Is
 Asterisk or Linux deciding to reply to a packet sent to 10.0.0.1
 (eht0) by sending that packet through 10.0.0.254?


  Understood.  When you up an interface with an IP address and netmask
the kernel automatically inserts a route for that network in the route
table (using that interface):

ip route show:
10.16.5.0/24 dev eth0  proto kernel  scope link  src 10.16.5.233  metric 1
default via 10.16.5.1 dev eth0  proto static

  As you can see I've also added a default route here.  Now, if I ping
my default route the kernel's ARP cache learns which MAC address that
IP has and over which interface:

arp -an:
? (10.16.5.1) at 00:13:72:26:36:b7 [ether] on eth0

  My guess is that if you had two NICs on the same subnet with
different IPs the kernel route table and ARP cache would get pretty
confused.  This seems so incredibly broken to me I've never tried

  Something else that seems strange about this arrangement, why would
you want to bother to configure other hosts on the LAN differently?
You're not really adding bandwidth/reliability (if you could call it
that) unless you configure other machines on the LAN to use the
different addresses...  Weird.

  In short: If you want to have two NICs on the same network, run them
through bonding.ko - PLEASE! ;)  If you need other IPs, add an alias
to your bonded interface!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] srv records not being honoured properly

[Brian J. Murrell]

On Fri, 2008-10-17 at 10:32 -0500, Andres wrote:
 
 Because Asterisk does not support that.

Which is just another way of saying Asterisk is broken then.  SRV
records have requirements for their correct use.  If those requirements
are ignored, that is a broken implementation.

 The only thing that Asterisk 
 does is use the first SRV entry

First in terms of what was returned, not sorted by priority and weight,
right?

 but it pays no attention to priorities 
 or weights.  It does not care about other SRV entries either.

Tsk tsk tsk.

 This is 
 how things have been as long as I can remember.

Wonderful.  Nothing like half implementing standards.

b.



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Re: [asterisk-users] prective dialer

[Richard Lyman]

There are a few options.

He should probably start on the wiki.

http://www.voip-info.org/wiki/view/Predictive+dialer

Steve Totaro wrote:
 If you can figure out how to generate .call files from your DB
 entries, you have it made.

 Vicidial needs alot of work as far as I am concerned, for free it is
 OK I guess.  I think using meetme conference rooms for everything is a
 kludgy hack, and the UI is less than nice (if you are into UIs).

 I suggest you continue on your own custom development if you have the
 time.  Check out Aheeva for inspiration.

 Thanks,
 Steve Totaro

 On Fri, Oct 17, 2008 at 1:31 AM, ram [EMAIL PROTECTED] wrote:
   
 look at Vicidial

 ram

 On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim [EMAIL PROTECTED] wrote:
 
 hi everybody

 This is Yavuz YILDIRIM

 I am software developer.I have a some problems in asterisk.
 I am using mysql db. Realtime using asterisk modules. On db i am using
 calling hundred fields for use dial.
 But i don't know how i can automaticly dial this fields on records
 numbers. Who can help me asterisk api and others.

 Thank you


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[asterisk-users] Snom M3 firmware Update

[Joseph L. Casale]

I started this at 4pm yesterday, its 10am and the handsets still say they are 
in progress?
Is that normal?

Thanks!
jlc
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Re: [asterisk-users] Strip prefix

[Tilghman Lesher]

On Friday 17 October 2008 10:32:30 michel freiha wrote:
 I tried to put + before the x like _+X but when making a call i got the
 following error:

 
 [Oct 17 15:08:58] WARNING[17532]: ast_expr2.fl:407 ast_yyerror:
 ast_yyerror():  syntax error: syntax error, unexpected '+', expecting $end;
 Input:
 +9613089187 = 111

  michel freiha wrote:
   Dear All,
  
   i have the following context defines in etensions.conf:
  
  
   [a2billing]
   exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)

Change this to:
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)

-- 
Tilghman

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Re: [asterisk-users] srv records not being honoured properly

[Eric ManxPower Wieling]

It should be fairly easy to write an AGI script that does the SRV query, 
  do whatever you want with the response, set a channel variable with 
the results and use that in your dialplan.

Brian J. Murrell wrote:
 On Fri, 2008-10-17 at 10:32 -0500, Andres wrote:
 Because Asterisk does not support that.
 
 Which is just another way of saying Asterisk is broken then.  SRV
 records have requirements for their correct use.  If those requirements
 are ignored, that is a broken implementation.
 
 The only thing that Asterisk 
 does is use the first SRV entry
 
 First in terms of what was returned, not sorted by priority and weight,
 right?
 
 but it pays no attention to priorities 
 or weights.  It does not care about other SRV entries either.
 
 Tsk tsk tsk.
 
 This is 
 how things have been as long as I can remember.
 
 Wonderful.  Nothing like half implementing standards.


-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html

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Re: [asterisk-users] srv records not being honoured properly

[Brian J. Murrell]

On Fri, 2008-10-17 at 11:18 -0500, Eric ManxPower Wieling wrote:
 It should be fairly easy to write an AGI script that does the SRV query, 
   do whatever you want with the response, set a channel variable with 
 the results and use that in your dialplan.

Maybe.  If I were an AGI hacker.  But really, should I (and every other
Asterisk user) have to?

b.



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Re: [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!

[C F]

On Fri, Oct 17, 2008 at 8:02 AM, Rodolfo Alcazar Portillo
[EMAIL PROTECTED] wrote:
 You are argueing with things like I can do it with panasonic, but it's
 not documented anywhere, documentation is a mess but not poor, sorry
 for underestimate your abilities. Sorry but I do [completely
 understanding Panasonic PBXs]. Not technical. Worst even: that's the
 propietary software culture.

 Many thanks for your advice, LED's issue was an imprtant feature to keep
 the eye on. I'll stick with asterisk, for many reasons *.

 Thanks again. Good luck.

 R

 * Only one (have more examples): even though I have old pana-PBXs, I
 bought a TDA100 (new model as provider offered). Has some SMDR bug,
 CERTIFIED tech tried to upgrade firmware twice, can not, ended
 programming it three times, costs us tens hours of service, until
 guarantee is lost, now works worst as initially: has noise on one line.
 Surely there is a fix (though not documented, as you wrote)...


Both the fact tech couldn't update it and the noise indicate the tech
didn't do it right. There is specific well documented procedure how to
do it. The reset precess should take care of both problems.

 ... and bug stills strong as ever.

 Am Freitag, den 17.10.2008, 00:57 -0400 schrieb C F:
 On Thu, Oct 16, 2008 at 7:25 AM, Rodolfo Alcazar Portillo
 [EMAIL PROTECTED] wrote:
  Am Mittwoch, den 15.10.2008, 20:51 -0400 schrieb C F:
  Being a Panasonic dealer and having more than 50 Asterisk system in
  production, I can tell you that if this is your first Asterisk
  project, then go with Panasonic, you'll safe yourself lots of
  aggravation and have a happier customer.
 
  You are completely wrong!
 
  Last 4 years, I installed/programmed 6 Panasonic (KXTD1232, 3x TA308,
  TDA100, TEM824) in our offices.

 TD1232 has been discontinued for at least 5 years. Don't know about
 the the TA308 since the last and only one I installed was in 1998, but
 I have not seen them advertised in the last 5 years. Which makes me
 think they are discontinued as well.

  Until now, I don't completely understand

 Sorry but I do.

  them. Their GUI software is really bad. The functions are awfully
  limited. Manuals are poor. Mailing lists with helpful people there is
  not.

 GUI on the TD is really really bad. Functions are not limited, like
 you said: I don't completely understand them. GUI on the TDA is nice
 and organized. Documentation is a mess but not poor. The main reason
 being it's translated from Japanise, and they don't explain the theory
 just the steps.
 Yes no mailing lists.

 
  Less than a week ago (friday), bought 3FXS, 1FXO with SIP
  (sipura/linksys), and KNOW NOTHING ABOUT asterisk. Today, I emulated
  almost all features we use (account codes, DISA, own dial plans), and I
  can really say: ASTERISK WORKS INCREDIBLE!
 
  I even programmed an AGI script, which injects a variable to
  extensions.conf; on the other hand, that means I can reboot a server
  from my cellphone, isn't that incredible?

 I can do that with Panasonic as well, no it's not documented anywhere
 in Panasonic docs.

  now, I dont' know how, but 99% I'm sure I can trigger a phonecall when
  one server is offline. Only with asterisk. I'm almost sure a Panasonic
  can't emulate this features. Maybe with some expensive software.
 
  Then, I'm going to suggest 30 Voip phones, 2 8xFXO digium. I made an
  informal presentation yesterday, the people were amused. Thanks the
  people (thanks, peru guys) which guided me. Good luck everybody, I'll
  keep asking on this list, sorry.
 
  No more panasonic for me :)
 
  Newbies, ask how to start.

 Congratulations and good for you. Sorry for underestimating your
 abilities. I wish you good luck, and if you feel comfortable please go
 ahead and use Asterisk. But if this is a type of customer that doesn't
 understand that you are experimenting with them and is not willing to
 work with you, then don't do it with THEM.

  --
  Rodolfo Alcazar
  Responsable red y datos
 
  Deutsche Gesellschaft für
  Technische Zusammenarbeit (GTZ) GmbH
 
  Programa de Apoyo a la Gestión Pública Descentralizada y
  Lucha Contra La Pobreza - PADEP
  Av. Sánchez Lima 2226
  La Paz, Bolivia
 
  Tel: +591 22417628 (121)
  Fax: +591 22417628 (126)
  Web: www.padep.org.bo
  Email: [EMAIL PROTECTED]
 
 

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 --
 Rodolfo Alcazar
 Responsable red y datos

 Deutsche Gesellschaft für
 Technische Zusammenarbeit (GTZ) GmbH

 Programa de Apoyo a la Gestión Pública Descentralizada y
 Lucha Contra La Pobreza - PADEP
 Av. Sánchez Lima 2226
 La Paz, Bolivia

 Tel: +591 22417628 (121)
 Fax: +591 22417628 (126)
 Web: www.padep.org.bo
 Email: [EMAIL PROTECTED]


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Re: [asterisk-users] srv records not being honoured properly

[Eric ManxPower Wieling]

Brian J. Murrell wrote:
 On Fri, 2008-10-17 at 11:18 -0500, Eric ManxPower Wieling wrote:
 It should be fairly easy to write an AGI script that does the SRV query, 
   do whatever you want with the response, set a channel variable with 
 the results and use that in your dialplan.
 
 Maybe.  If I were an AGI hacker.  But really, should I (and every other
 Asterisk user) have to?
 

If you fight Asterisk's oddities then you will have a depressing and 
miserable life.  If you embrace Asterisk's oddities then you will have a 
joyous and enlightened life. 8-)

I agree that if Asterisk has SRV support it should work in the way 
expected.  The reason Asterisk's SRV support has not been fixed is 
because nobody with the skills has thought the issue was important 
enough to fix.

Do you know any programming languages?

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html

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Re: [asterisk-users] Matching *, + and # in the dialplan

[Tilghman Lesher]

On Friday 17 October 2008 10:15:22 Olivier wrote:
 2008/10/17 John Todd [EMAIL PROTECTED]

  At 5:46 PM +0200 2008/10/16, Olivier wrote:
  Is Incomplete() application an acceptable work around for ISN ?
 
  It is impossible to determine the full sequence of digits for an ISN
  number ahead of time (well, I shouldn't say impossible because one
  could create a really nasty hack...) because the number of digits
  isn't known until the user indicates they are done dialing.  To
  determine if the subscriber and/or the ITAD are valid, one must
  perform a lookup and possibly a connection attempt to the remote
  system for determination.
 
  Both the subscriber number (the stuff to the left of the *) and the
  ITAD number (the stuff to the right of the *) are not bounded by a
  particular pattern.  Therefore, it is not reasonable to create
  regexps other than does this string contain at least one but maybe
  more digits, followed by the * sign, followed by at least one but
  maybe more digits.
 
  So, 1234*256 is a valid ISN sequence, as is 12345*2567.  (Though only
  the first one of the two is actually active at the moment.)
 
  See http://www.freenum.org/ for more details on what makes up an ISN.
  It's free to participate, and Asterisk incorporates ISN-style dialing
  as a default in the ENUMLOOKUP routines.  Accepting inbound calls is
  easy as well - it's just SIP inbound calling, no magic there.
 
  JT

 Writing this :

 exten = _XXX*,1,Incomplete()
 exten = _*,1,Incomplete()
 exten = _X*,1,Incomplete()

 would work to catch 3 to 5 digits private extensions, but it would remains
 difficult to catch the ITAD number, right ?

There's no need to do anything like that.  You're not seeing the
possibilities:

exten = _X.,1,Set(main=${CUT(EXTEN,*,1)})
exten = _X.,n,Set(itad=${CUT(EXTEN,*,2)})
exten = _X.,n,GotoIf($[${itad}=]?incomplete) ; or some other test
exten = _X.,n,Dial(...)
exten = _X.,n+1000(incomplete),Incomplete()

-- 
Tilghman

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Re: [asterisk-users] srv records not being honoured properly

[Brian J. Murrell]

On Fri, 2008-10-17 at 11:35 -0500, Eric ManxPower Wieling wrote:
 If you fight Asterisk's oddities then you will have a depressing and 
 miserable life.  If you embrace Asterisk's oddities then you will have a 
 joyous and enlightened life. 8-)

I just want something that works.  :-)

 I agree that if Asterisk has SRV support it should work in the way 
 expected.  The reason Asterisk's SRV support has not been fixed is 
 because nobody with the skills has thought the issue was important 
 enough to fix.

~sigh~

 Do you know any programming languages?

I do and I'm pretty sure I could fix the problem.  I just have so much
other stuff to do that I didn't really want to be an Asterisk hacker on
top of all that.

b.



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Re: [asterisk-users] Snom M3 firmware Update

[Tim Litwiller]

The wiki says it should take about 20 minutes per handset.

Joseph L. Casale wrote:

 I started this at 4pm yesterday, its 10am and the handsets still say 
 they are in progress?
 Is that normal?

 Thanks!
 jlc

 

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Re: [asterisk-users] One way voice after call transfer (bugs 9305, 13120)

[Yehavi Bourvine]

Yes, the problem is there as well...

 __Yehavi:

 2008/10/17 broadband Voice [EMAIL PROTECTED]

 Is anyone seeing it in 1.4.22? I plan to upgrade to that version, my fear
 is zaptel compatibility.

 On Fri, Oct 17, 2008 at 1:08 AM, Yehavi Bourvine 
 [EMAIL PROTECTED] wrote:

  unfortunately I still see it in 1.6.0...

   __Yehavi:

  2008/10/17 broadband Voice [EMAIL PROTECTED]

  I am having a similar problem and I'm using Asterisk 1.4.19 and have
 that problem on some calls through our calling card platforms. Someone
 suggested we use 1.4.3 and have not tried it yet. Any comments from the
 group.

 On Tue, Jul 29, 2008 at 1:19 AM, Yehavi Bourvine +972-8-9489444 
 [EMAIL PROTECTED] wrote:

 Hello,

  I am having an issue here that after an attended call transfer there is
 no
 audio on one way; the problem is caused by Asterisk sending two INVITE
 messages
 without waiting for an ack for the first one.

  The issue has been reported on bug 9305, has been fixed and the fix is
 now
 included inside the main stream (version 1.4.21). However, I still get
 this
 behaviour, so I opened a new bug (13120). This bug sits there for over a
 week
 with no reponse...

  Has anyone else noticed this behaviour? Any idea what I can do? My
 users are
 angry on me...

Thanks! __Yehavi:

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Re: [asterisk-users] srv records not being honoured properly

[Tilghman Lesher]

On Friday 17 October 2008 11:46:09 Brian J. Murrell wrote:
 On Fri, 2008-10-17 at 11:35 -0500, Eric ManxPower Wieling wrote:
  If you fight Asterisk's oddities then you will have a depressing and
  miserable life.  If you embrace Asterisk's oddities then you will have a
  joyous and enlightened life. 8-)

 I just want something that works.  :-)

  I agree that if Asterisk has SRV support it should work in the way
  expected.  The reason Asterisk's SRV support has not been fixed is
  because nobody with the skills has thought the issue was important
  enough to fix.

 ~sigh~

  Do you know any programming languages?

 I do and I'm pretty sure I could fix the problem.  I just have so much
 other stuff to do that I didn't really want to be an Asterisk hacker on
 top of all that.

Have you considered upgrading to 1.6?  I believe it is fixed in that branch.

-- 
Tilghman

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Re: [asterisk-users] [Asterisk-users] +heartbeat

[Wilton Helm]

My guess is that if you had two NICs on the same subnet with
different IPs the kernel route table and ARP cache would get pretty
confused.  This seems so incredibly broken to me I've never tried


That was my guess and point to begin with.  I was not aware of or thinking 
about bonding.  Without something out of the ordinary in the protocol stack, 
there is no way to determine which NIC the OS will use for a given destination 
IP, since either can get there.  That is why the hi-rel stuff I do has two 
parallel LANS with different subnets.  It places more of the burden on the 
program to know how to do backup, but I control the code in the projects I'm 
doing, so its not a problem.

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Re: [asterisk-users] Snom M3 firmware Update

[Joseph L. Casale]

The wiki says it should take about 20 minutes per handset.

yeah I just found that, and so I called tech support
and they said to reset the gateway, and if needed to pull
the battery out of the phones and power them on. I have done
this and they restarted the firmware download so I will wait
and see. The tech suggested sometimes the handsets can loose
connectivity with the base and this happens...

jlc

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Re: [asterisk-users] Snom M3 firmware Update

[Torbjörn Abrahamsson]

Our M3's also got stuck somewhere in the middle, so we rebooted the base by
pressing the reset button. After this the handset moved along and continued
with the upgrade. This happended on all our 5 handsets, connected to three
different bases, one with fw 1.01 and two with 1.07, upgrading to 1.16.

Very logical! I did not feel very good pressing the reset button during a
firmware upgrade. And that it should take 20 minutes anyway, even if working
ok, is quite absurd. Just about an eternity... :)

// T

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Tim Litwiller
 Sent: den 17 oktober 2008 19:01
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Snom M3 firmware Update
 
 The wiki says it should take about 20 minutes per handset.
 
 Joseph L. Casale wrote:
 
  I started this at 4pm yesterday, its 10am and the handsets 
 still say 
  they are in progress?
  Is that normal?
 
  Thanks!
  jlc
 
  
 --
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Re: [asterisk-users] srv records not being honoured properly

[Brian J. Murrell]

On Fri, 2008-10-17 at 12:11 -0500, Tilghman Lesher wrote:
 
 Have you considered upgrading to 1.6?

Not to this point, no.  1.4 does everything I want and if it ain't
broke, don't fix it.  Well, now it's broke I guess.  Still, Ubuntu still
uses 1.4 and I don't like having to maintain my own packages.

   I believe it is fixed in that branch.

I think I'd sooner (backport a) fix (it) and offer it back upstream
before I'd go to maintaining my own packages.  I can do it, I just have
better things to do.

b.



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Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

[Juan E. Rodríguez]

I do, I am planning to have little more than 1000. Right now I had 
managed little more than 700 SIP channels + 100 IAX channels.


Do you think this can cause any problem?? --I mean, having this RTP 
ports range--



Tzafrir Cohen wrote:

On Fri, Oct 17, 2008 at 01:24:35AM -0400, Juan Rodríguez wrote:
  

Tzafrir:

Following the comments on your post, I started checking (after breaking my
head 'googling') the UDP ports in use, and found out that the script that my
Asterisk is running was using UDP connection too. This caused that ports
from 10,000 to 20,000 could not be used by Asterisk.

I change the port range from 10,000 to 40,, and now everything looks OK.



Why not change it to 9000- ?

Do you actually need more than 1000 sockets at a time?

  


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Re: [asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)

[Tzafrir Cohen]

On Fri, Oct 17, 2008 at 03:11:17PM -0400, Juan E. Rodríguez wrote:
 I do, I am planning to have little more than 1000. Right now I had 
 managed little more than 700 SIP channels + 100 IAX channels.
 
 Do you think this can cause any problem?? --I mean, having this RTP 
 ports range--

If you never had anything close to the order of magnitude of 1 SIP
channels, the range of 1 RTP ports should have been well over
enough. Unless your scripts have done very funny things (using over 5
sockets per Asterisk channel. Which is funny indeed, becuase even
chan_h323 isn't that bad).

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] One Way Audio Problem

[Jeff LaCoursiere]

On Thu, 16 Oct 2008, GNUbie wrote:

 Hello,

 On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
 
  A packet trace will probably show exactly what is happening.  Try:
 
  tcpdump -nlXs 8192 -i eth0 port 5060
 
  You should be able to see the SIP information going back and forth and
  will probably show you that your NAT rules are applying when they
  shouldn't.  I agree with first turning off your firewall and testing...
  but if that actually solves the problem you need to know why.  This should
  tell why.

 Why eth0 when in fact it is not being used AFAIK? My Asterisk box is
 connected to the LAN via its eth1 interface and the SIP phone is
 calling from the LAN to the analog telephone via FXO/POTS. Again,
 below is the call scenario diagram:

 [SNOM] ==LAN== eth1 [ASTERISK] fxo ==POTS== [ANALOG_TELEPHONE]
 eth0
   ||
 INTERNET

You should try on both interfaces.  If you see packets on eth0 then your
NAT rules are leaking!  Try on eth1 to see the SIP headers and tell if
your NAT rules are doing what you expect.

This is always my first attack...

j


 Please advice.  Thank you in advance.

 Regards,

 GNUbie

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[asterisk-users] Transfering Calls back on the same PRI

[Ron Joffe]

Here is my hardware configuration

TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk

The PBX is a Siemens Hicom 200 EX (Model 80)

We are connecting between the PBX and Asterisk using QSIG switch type.

What I want to do is the following:

1. Call comes from TELCO via PRI1 and enters PBX
2. PBX Routes call to Asterisk via PRI2
3. Asterisk does some call handling (IVR)
4. Call needs to be transfered to an extension on the PBX.

I can easily set up a dial command to pass the call back to the PBX from 
Asterisk along PRI2 but this uses 2 B Channels.

How do I tell asterisk to send a transfer request to the PBX so Asterisk is 
out of the loop?

Thanks,

Ron



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[asterisk-users] anoyingly answers already in use pstn line

[Jack Bates]

I am using Asterisk and an X101P card as a glorified answering machine.
We have a residential PSTN line with about six phones connected to it.
Like an answering machine, I want Asterisk answer the line *only* when
an incoming call is not answered after four rings.

This mostly works. My extensions.conf is at the end of this message.

The problem is that Asterisk will sometimes answer the line when someone
is already talking on one of the six phones connected to it. Sometimes
Asterisk will answer the line and start playing the greeting in the
middle of a conversation! This is especially a problem when I am talking
on the phone to an automated system, because although I hang up the
phone I am talking on, neither the automated system nor Asterisk will
hang up.

I have not yet discovered a pattern to when Asterisk answers the line.
It always answers after four rings, but it sometimes answers when
someone is already talking on one of the phones connected to the line.

In a perfect world, Asterisk would be the only thing connected to the
line, and all our phones would be Asterisk extensions. Unfortunately we
do not currently have the required VoIP phones or FXS interface...

Is there any way to make Asterisk less flaky, and answer the line *only*
when an incoming call is not answered after four rings?

---

[default]

exten = s,1,Wait(20)
exten = s,n,Answer
exten = s,n,Background(recordings/coop-greeting)
exten = s,n(instruct),Background(recordings/leave-message)
exten = s,n,Background(recordings/enter-extension)
exten = s,n,Background(recordings/dial-by-name)
exten = s,n,Background(recordings/visit-website)
exten = s,n,WaitExten

; General delivery mailbox
exten = #,1,Voicemail(6000)
exten = #,n,Goto(s,instruct)

; Dial by name
exten = a,1,Directory(default)

; Entering an invalid extension replays the instructions
exten = i,1,Playback(invalid)
exten = i,n,Goto(s,instruct)

; Timeout goes to voicemail
exten = t,1,Goto(#,1)

exten = 6003,1,Macro(stdexten,6003,SIP/cstewart)
exten = 6004,1,Macro(stdexten,6004,SIP/mhockley)
exten = 6005,1,Macro(stdexten,6005,SIP/jbates)
[...]


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Re: [asterisk-users] anoyingly answers already in use pstn line

[Gleim, Jason]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jack Bates
 Sent: Friday, October 17, 2008 4:48 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] anoyingly answers already in use pstn line
 
 I am using Asterisk and an X101P card as a glorified answering
machine.
 We have a residential PSTN line with about six phones connected to it.
 Like an answering machine, I want Asterisk answer the line *only* when
 an incoming call is not answered after four rings.
 
 This mostly works. My extensions.conf is at the end of this message.
 
 The problem is that Asterisk will sometimes answer the line when
 someone
 is already talking on one of the six phones connected to it. Sometimes
 Asterisk will answer the line and start playing the greeting in the
 middle of a conversation! This is especially a problem when I am
 talking
 on the phone to an automated system, because although I hang up the
 phone I am talking on, neither the automated system nor Asterisk will
 hang up.
 
 I have not yet discovered a pattern to when Asterisk answers the line.
 It always answers after four rings, but it sometimes answers when
 someone is already talking on one of the phones connected to the line.
 
 In a perfect world, Asterisk would be the only thing connected to the
 line, and all our phones would be Asterisk extensions. Unfortunately
we
 do not currently have the required VoIP phones or FXS interface...
 
 Is there any way to make Asterisk less flaky, and answer the line
 *only*
 when an incoming call is not answered after four rings?
 
 ---
 
 [default]
 
 exten = s,1,Wait(20)
 exten = s,n,Answer
 exten = s,n,Background(recordings/coop-greeting)
 exten = s,n(instruct),Background(recordings/leave-message)
 exten = s,n,Background(recordings/enter-extension)
 exten = s,n,Background(recordings/dial-by-name)
 exten = s,n,Background(recordings/visit-website)
 exten = s,n,WaitExten
 
 ; General delivery mailbox
 exten = #,1,Voicemail(6000)
 exten = #,n,Goto(s,instruct)
 
 ; Dial by name
 exten = a,1,Directory(default)
 
 ; Entering an invalid extension replays the instructions
 exten = i,1,Playback(invalid)
 exten = i,n,Goto(s,instruct)
 
 ; Timeout goes to voicemail
 exten = t,1,Goto(#,1)
 
 exten = 6003,1,Macro(stdexten,6003,SIP/cstewart)
 exten = 6004,1,Macro(stdexten,6004,SIP/mhockley)
 exten = 6005,1,Macro(stdexten,6005,SIP/jbates)
 [...]
 
 
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Others may wish to chime in and confirm or deny this but the card is
probably getting confused by you loading the line with the other phones.
I know most of the analog cards I've worked with (which does not include
the X101P) really get cranky if there is anything else hanging off that
line. The only solution I've seen to the problem is to change things
around so that the card is the only thing on the line.

In know you said you haven't switched to IP or FXS but is there a reason
why? Your problem would go away and you would be able to leverage all
the features of Asterisk if you just got a single ATA. Something like a
Linksys PAP2T-NA can be had for around $55 USD. Disconnect your PSTN
line at the entrance bridge, run it into the X101P, and plug the PAP2T
into the house. It is convenient and doesn't require any changes in
internal wiring. (You might have to run a few wires if the bridge is on
the back of your house.) No need for new phones or anything. Granted,
all the internal phones would be on one extension but you have that
situation now... And with the ATA you've solved your problem. As the
need arises, get more ATAs or IP phones or whatever and build out your
internal phone network.

Jason

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Re: [asterisk-users] One Way Audio Problem

[Brent Davidson]

GNUbie wrote:
 What particular configs are you looking for? Below is my current setup
 and scenario:

 [snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone]

 SNOM is using the 192.168.101.102 IP address
 Asterisk is using 192.168.101.1 IP address for its eth1 interface
 FXO port is connected to the POTS
 SNOM doesn't need to go out to the Internet in this scenario, AFAIK.

 Below is my current NAT rules:

 # iptables -L -v -t nat
 Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes)
  pkts bytes target prot opt in out source
 destination
 11460  760K RETURN 0--  anyany 192.168.101.0/24
 !192.168.101.0/24

 Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes)
  pkts bytes target prot opt in out source
 destination
 11408  757K MASQUERADE  0--  anyeth0192.168.101.0/24
 anywhere

 Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes)
  pkts bytes target prot opt in out source   
 destination

 Please advice if you need more information from me.

 Regards,

 GNUbie
Having had many years of experience working with iptables I can tell you 
that when IP Forwarding is enabled on a Linux machine things can get a 
bit tricky. In my experience using a Masquerade rule can cause some 
major weirdness.  Try doing this:

Instead of the Masquerade rule use:

iptables -t nat -A POSTROUTING -i eth1 -o eth0 -j SNAT --to-source 
public ip of eth0

Also, in the general section of your sip.conf make sure you have:

bindaddr=192.168.101.1

to make sure asterisk is not sending sip packets using the public IP 
then effectively trying to communicate with the phone by Masquerading 
the packets coming in over the eth1 to eth0.  This is more than likely 
what is happening. (It's normlly bindaddr=0.0.0.0)

Good luck,
Brent


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[asterisk-users] Transfering Calls back on the same PRI

[Ron Joffe]

Here is my hardware configuration

TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk

The PBX is a Siemens Hicom 200 EX (Model 80)

We are connecting between the PBX and Asterisk using QSIG switch type.

What I want to do is the following:

1. Call comes from TELCO via PRI1 and enters PBX
2. PBX Routes call to Asterisk via PRI2
3. Asterisk does some call handling (IVR)
4. Call needs to be transfered to an extension on the PBX.

I can easily set up a dial command to pass the call back to the PBX from 
Asterisk along PRI2 but this uses 2 B Channels.

How do I tell asterisk to send a transfer request to the PBX so Asterisk is 
out of the loop?

Thanks,

Ron



-- 
Ron Joffe
Siena Tech, Inc.
3319 Willow Glen Drive
Oak Hill, VA 20171
(919) 928-0404

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Re: [asterisk-users] Transfering Calls back on the same PRI

[Steve Totaro]

On Fri, Oct 17, 2008 at 5:24 PM, Ron Joffe [EMAIL PROTECTED] wrote:
 Here is my hardware configuration

 TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk

 The PBX is a Siemens Hicom 200 EX (Model 80)

 We are connecting between the PBX and Asterisk using QSIG switch type.

 What I want to do is the following:

 1. Call comes from TELCO via PRI1 and enters PBX
 2. PBX Routes call to Asterisk via PRI2
 3. Asterisk does some call handling (IVR)
 4. Call needs to be transfered to an extension on the PBX.

 I can easily set up a dial command to pass the call back to the PBX from
 Asterisk along PRI2 but this uses 2 B Channels.

 How do I tell asterisk to send a transfer request to the PBX so Asterisk is
 out of the loop?

 Thanks,

 Ron



 --
 Ron Joffe
 Siena Tech, Inc.
 3319 Willow Glen Drive
 Oak Hill, VA 20171
 (919) 928-0404

I would engineer the system so that Asterisk is in the middle rather
than the far end.  Is there a reason why you don't want to or cannot
do that?

TELCO --- PRI1 --- Asterisk --- PRI2 --- PBX

I have done dozens and dozens of this type of implementation,
sometimes you have to be very creative, but I have never failed.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] anoyingly answers already in use pstn line

[Tzafrir Cohen]

On Fri, Oct 17, 2008 at 05:04:32PM -0400, Gleim, Jason wrote:
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Jack Bates
  Sent: Friday, October 17, 2008 4:48 PM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] anoyingly answers already in use pstn line
  
  I am using Asterisk and an X101P card as a glorified answering
 machine.
  We have a residential PSTN line with about six phones connected to it.
  Like an answering machine, I want Asterisk answer the line *only* when
  an incoming call is not answered after four rings.
  
  This mostly works. My extensions.conf is at the end of this message.
  
  The problem is that Asterisk will sometimes answer the line when
  someone
  is already talking on one of the six phones connected to it. Sometimes
  Asterisk will answer the line and start playing the greeting in the
  middle of a conversation! This is especially a problem when I am
  talking
  on the phone to an automated system, because although I hang up the
  phone I am talking on, neither the automated system nor Asterisk will
  hang up.
  
  I have not yet discovered a pattern to when Asterisk answers the line.
  It always answers after four rings, but it sometimes answers when
  someone is already talking on one of the phones connected to the line.
  
  In a perfect world, Asterisk would be the only thing connected to the
  line, and all our phones would be Asterisk extensions. Unfortunately
 we
  do not currently have the required VoIP phones or FXS interface...
  
  Is there any way to make Asterisk less flaky, and answer the line
  *only*
  when an incoming call is not answered after four rings?
  
  ---
  
  [default]
  
  exten = s,1,Wait(20)
  exten = s,n,Answer
  exten = s,n,Background(recordings/coop-greeting)
  exten = s,n(instruct),Background(recordings/leave-message)
  exten = s,n,Background(recordings/enter-extension)
  exten = s,n,Background(recordings/dial-by-name)
  exten = s,n,Background(recordings/visit-website)
  exten = s,n,WaitExten
  
  ; General delivery mailbox
  exten = #,1,Voicemail(6000)
  exten = #,n,Goto(s,instruct)
  
  ; Dial by name
  exten = a,1,Directory(default)
  
  ; Entering an invalid extension replays the instructions
  exten = i,1,Playback(invalid)
  exten = i,n,Goto(s,instruct)
  
  ; Timeout goes to voicemail
  exten = t,1,Goto(#,1)
  
  exten = 6003,1,Macro(stdexten,6003,SIP/cstewart)
  exten = 6004,1,Macro(stdexten,6004,SIP/mhockley)
  exten = 6005,1,Macro(stdexten,6005,SIP/jbates)
  [...]
  
  
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 Others may wish to chime in and confirm or deny this but the card is
 probably getting confused by you loading the line with the other phones.
 I know most of the analog cards I've worked with (which does not include
 the X101P) really get cranky if there is anything else hanging off that
 line. The only solution I've seen to the problem is to change things
 around so that the card is the only thing on the line.

The cranky card here is not the issue. It would be the same with any
other card.

 
 In know you said you haven't switched to IP or FXS but is there a reason
 why? 

That would require rewiring.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Phones lose contact

[Jerry Jones]

On Oct 17, 2008, at 5:14 PM, Paul Douglas Franklin wrote:

 When off site, our IP phones lose contact after a few minutes of
 inactivity.  They no longer receive calls, though they can call out.
 Asterisk acts as if it is ringing the phone, but the phone does not  
 ring.
 The phones are behind a NAT/firewall.
 What is the most reasonable solution?

qualify=yes


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Re: [asterisk-users] Cisco 7960 not always receiving incoming calls

[Stephen Reese]

On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED] wrote:
 I've searched around and found a few similar situations where the
 phone will call out when using a Asterisk server but not receive
 inbound calls. My issue is a little stranger. If I call out from the
 phone then the phone will receive the next inbound call. The phone
 will not receive another inbound call until a call out again from it
 first. Any ideas?

 I am using SIP and am using the latest phone image from Cisco to date.
 I am also using a Cisco router at the gateway. Is there anything
 special I should to to make this work? Note my soft phone does not
 have any issues using the same dialing rules and extension
 information. Here is some of my config stuff:

 ns1*CLI sip show peers
 Name/username  HostDyn Nat ACL Port Status
 vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
 vitel-inbound/rsreese  64.2.142.1165060 Unmonitored
 101/10168.156.63.118D   N  1038 Unmonitored
 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]


 Inbound call in progress when the SIP Cisco phone doesn't ring

 Verbosity is at least 5
  == Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
 default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
 SIP/101SIP/[EMAIL PROTECTED],30) in new stack
  == Using SIP RTP CoS mark 5
-- Called 101
  == Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
 SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
  == Spawn extension (default, 101, 1) exited non-zero on 
 'SIP/rsreese-082a8358'

 Inbound call in progress when the SIP Cisco does ring after I first
 make an outbound call

  == Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
 default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
 SIP/101SIP/[EMAIL PROTECTED],30) in new stack
  == Using SIP RTP CoS mark 5
-- Called 101
  == Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
 SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
  == Spawn extension (default, 101, 1) exited non-zero on 
 'SIP/rsreese-082a8358'

 Extensions.conf, which I don't think is relevent, I've changed it to
 just a simple dial the sip phone and it still fails.

 exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
 exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
 exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
 exten = 101,n(lbl_default_0),Hangup()
 exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
 exten = 101,n,Goto(lbl_default_0)

 Cisco phone stuff from a Cisco 7960:

 SIPDefault.cnf
 image_version: P0S3-08-9-00
 proxy1_address: neocipher.net; Can be dotted IP or FQDN
 proxy_register: 1
 messages_uri:   100
 phone_password: cisco ; Limited to 31 characters (Default - cisco)
 sntp_server:10.10.10.1
 time_zone:  EST
 dial_template: DIALPLAN
 nat_enable: 1
 nat_address: 172.16.2.1
 nat_received_processing: 1

 outbound_proxy_port: 5060
 outbond_proxy: ns1.neocipher.net

 SIP0112B9EAFF72.cnf
 image_version: P0S3-08-9-00

 # Line 1 Setup
 line1_name: 101
 line1_authname: 101
 line1_shortname: Line 101
 line1_password: test
 line1_displayname: Stephen Reese; # Line 1 Display Name (Display
 name to use for SIP messaging)

 # Line 2 Setup
 #line2_name: scott
 #line2_authname: scott
 #line2_shortname: 201
 #line2_password: tiger
 #line2_displayname: Larry Ellison; # Line 2 Display Name (Display
 name to use for SIP messaging)

 # Phone Label (Text desired to be displayed in upper right corner)
 phone_label: Stephen Reese ; Has no effect on SIP messaging
 # Phone Password (Password to be used for console or telnet login)
 phone_password: goaway ; Limited to 31 characters (Default - cisco)
 # User classifcation used when Registering [ none(default), phone, ip ]
 user_info: none
 telnet_level: 2

 Any ideas or help would be great, thanks.


I'm still unable to wrap my head around this problem. I can recieve a
call after I first call out from the line/phone. I didn't think it's a
NAT issue since it kind of works.

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Re: [asterisk-users] Cisco 7960 not always receiving incoming calls

[Darryl Dunkin]

It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device, allowing calls to come
in until it expires (default on many devices is 60 seconds). You may
also receive inbound calls when the phone reregisters regularly. Try
'qualify=yes' in your phones section in sip.conf to send keepalives
(option packets in this case) every two seconds to the phone to keep it
from going idle. You can see the state of the phone from the console
with a 'sip show peers', if unreachable, your NAT device has killed the
NAT forward.

Should look like one of these:
xxx/xxx x.x.x.x   D   N  5060 OK (46 ms)   
xxx/xxx x.x.x.x   D   N  5060 UNREACHABLE

As another troubleshooting step, you can telnet to the phone and have it
reregister with Asterisk manually (register line 1 1) to see if that
brings it back to life.

If qualify doesn't do it, see if you can increase UDP timeouts in your
firewall/NAT device.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Reese
Sent: Friday, October 17, 2008 17:04
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls

On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED]
wrote:
 I've searched around and found a few similar situations where the
 phone will call out when using a Asterisk server but not receive
 inbound calls. My issue is a little stranger. If I call out from the
 phone then the phone will receive the next inbound call. The phone
 will not receive another inbound call until a call out again from it
 first. Any ideas?

 I am using SIP and am using the latest phone image from Cisco to date.
 I am also using a Cisco router at the gateway. Is there anything
 special I should to to make this work? Note my soft phone does not
 have any issues using the same dialing rules and extension
 information. Here is some of my config stuff:

 ns1*CLI sip show peers
 Name/username  HostDyn Nat ACL Port Status
 vitel-outbound/rsreese 64.2.142.22 5060
Unmonitored
 vitel-inbound/rsreese  64.2.142.1165060
Unmonitored
 101/10168.156.63.118D   N  1038
Unmonitored
 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0
offline]


 Inbound call in progress when the SIP Cisco phone doesn't ring

 Verbosity is at least 5
  == Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
 default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
 SIP/101SIP/[EMAIL PROTECTED],30) in new stack
  == Using SIP RTP CoS mark 5
-- Called 101
  == Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
 SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
  == Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'

 Inbound call in progress when the SIP Cisco does ring after I first
 make an outbound call

  == Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
 default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
 SIP/101SIP/[EMAIL PROTECTED],30) in new stack
  == Using SIP RTP CoS mark 5
-- Called 101
  == Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
 SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
  == Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'

 Extensions.conf, which I don't think is relevent, I've changed it to
 just a simple dial the sip phone and it still fails.

 exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
 exten = 101,n,GotoIf($[${DIALSTATUS} =
CHANUNAVAIL]?lbl_default_1:)
 exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
 exten = 101,n(lbl_default_0),Hangup()
 exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
 exten = 101,n,Goto(lbl_default_0)

 Cisco phone stuff from a Cisco 7960:

 SIPDefault.cnf
 image_version: P0S3-08-9-00
 proxy1_address: neocipher.net; Can be dotted IP or FQDN
 proxy_register: 1
 messages_uri:   100
 phone_password: cisco ; Limited to 31 characters (Default - cisco)
 sntp_server:10.10.10.1
 time_zone:  EST
 dial_template: DIALPLAN
 nat_enable: 1
 nat_address: 172.16.2.1
 nat_received_processing: 1

 outbound_proxy_port: 5060
 outbond_proxy: ns1.neocipher.net

 SIP0112B9EAFF72.cnf
 image_version: P0S3-08-9-00

 # Line 1 Setup
 line1_name: 101
 line1_authname: 101
 line1_shortname: Line 101
 line1_password: test
 line1_displayname: Stephen Reese; # Line 1 Display Name (Display
 name to use for SIP messaging)

 # Line 2 Setup
 #line2_name: 

Re: [asterisk-users] Cisco 7960 not always receiving incoming calls

[Darryl Dunkin]

Sorry, I missed the Cisco router bit.

As a last resort (if qualify doesn't help), you could enter this
(global) to increase the timeout on UDP translations:
ip nat translation udp-timeout 300 (or greater if you prefer)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darryl
Dunkin
Sent: Friday, October 17, 2008 17:28
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls

It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device, allowing calls to come
in until it expires (default on many devices is 60 seconds). You may
also receive inbound calls when the phone reregisters regularly. Try
'qualify=yes' in your phones section in sip.conf to send keepalives
(option packets in this case) every two seconds to the phone to keep it
from going idle. You can see the state of the phone from the console
with a 'sip show peers', if unreachable, your NAT device has killed the
NAT forward.

Should look like one of these:
xxx/xxx x.x.x.x   D   N  5060 OK (46 ms)   
xxx/xxx x.x.x.x   D   N  5060 UNREACHABLE

As another troubleshooting step, you can telnet to the phone and have it
reregister with Asterisk manually (register line 1 1) to see if that
brings it back to life.

If qualify doesn't do it, see if you can increase UDP timeouts in your
firewall/NAT device.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Reese
Sent: Friday, October 17, 2008 17:04
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls

On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED]
wrote:
 I've searched around and found a few similar situations where the
 phone will call out when using a Asterisk server but not receive
 inbound calls. My issue is a little stranger. If I call out from the
 phone then the phone will receive the next inbound call. The phone
 will not receive another inbound call until a call out again from it
 first. Any ideas?

 I am using SIP and am using the latest phone image from Cisco to date.
 I am also using a Cisco router at the gateway. Is there anything
 special I should to to make this work? Note my soft phone does not
 have any issues using the same dialing rules and extension
 information. Here is some of my config stuff:

 ns1*CLI sip show peers
 Name/username  HostDyn Nat ACL Port Status
 vitel-outbound/rsreese 64.2.142.22 5060
Unmonitored
 vitel-inbound/rsreese  64.2.142.1165060
Unmonitored
 101/10168.156.63.118D   N  1038
Unmonitored
 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0
offline]


 Inbound call in progress when the SIP Cisco phone doesn't ring

 Verbosity is at least 5
  == Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
 default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
 SIP/101SIP/[EMAIL PROTECTED],30) in new stack
  == Using SIP RTP CoS mark 5
-- Called 101
  == Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
 SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
  == Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'

 Inbound call in progress when the SIP Cisco does ring after I first
 make an outbound call

  == Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
 default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
 SIP/101SIP/[EMAIL PROTECTED],30) in new stack
  == Using SIP RTP CoS mark 5
-- Called 101
  == Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
 SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
  == Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'

 Extensions.conf, which I don't think is relevent, I've changed it to
 just a simple dial the sip phone and it still fails.

 exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
 exten = 101,n,GotoIf($[${DIALSTATUS} =
CHANUNAVAIL]?lbl_default_1:)
 exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
 exten = 101,n(lbl_default_0),Hangup()
 exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
 exten = 101,n,Goto(lbl_default_0)

 Cisco phone stuff from a Cisco 7960:

 SIPDefault.cnf
 image_version: P0S3-08-9-00
 proxy1_address: neocipher.net; Can be dotted IP or FQDN
 proxy_register: 1
 messages_uri:   100
 phone_password: cisco ; Limited to 31 characters (Default - cisco)
 

[asterisk-users] SER + Asterisk

[Joseph]

I am running Asterisk and would like to add SER to register my (sip) DID and 
connect it to asterisk; 
but I'm not sure if this is the correct forum. 

I have as DID, sip account with one VoIP provider; currently Im using just 
stand alone SIP phone and register with the VoIP provider via: 
stun.fwdnet.net

Is it possible to use SER to register with the provider and forward the call 
Asterisk.
Can anybody provide a link to practical example.

I'm comfortable with Asterisk but I just install SER and can not find 
appropriate example to follow on www.iptel.org web-page.
There are a lot explanations but not enough practical examples to follow.

-- 
#Joseph

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Re: [asterisk-users] Transfering Calls back on the same PRI

[Ron Joffe]

On Friday 17 October 2008 17:38, Steve Totaro wrote:
 I would engineer the system so that Asterisk is in the middle rather
 than the far end.  Is there a reason why you don't want to or cannot
 do that?

 TELCO --- PRI1 --- Asterisk --- PRI2 --- PBX

Steve, 

I have also done this same method in the past. In this case the number of 
PRI's entering the PBX far outweigh the number of PRI's in the Asterisk 
server, so it is not an option. I tried to simplify the example.

Any other suggestions ?

Ron


-- 
Ron Joffe
Siena Tech, Inc.
3319 Willow Glen Drive
Oak Hill, VA 20171
(919) 928-0404

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Re: [asterisk-users] SER + Asterisk

[Alex Balashov]

SER is defunct.  Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.

On Fri, October 17, 2008 9:36 pm, Joseph wrote:

 I am running Asterisk and would like to add SER to register my (sip) DID
 and connect it to asterisk;
 but I'm not sure if this is the correct forum.

 I have as DID, sip account with one VoIP provider; currently Im using
 just stand alone SIP phone and register with the VoIP provider via:
 stun.fwdnet.net

 Is it possible to use SER to register with the provider and forward the
 call Asterisk.
 Can anybody provide a link to practical example.

 I'm comfortable with Asterisk but I just install SER and can not find
 appropriate example to follow on www.iptel.org web-page.
 There are a lot explanations but not enough practical examples to follow.

 --
 #Joseph

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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599


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Re: [asterisk-users] SER + Asterisk

[Kristian Kielhofner]

On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote:

  SER is defunct.  Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.


  Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the
thing to do.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] SER + Asterisk

[Joseph]

I'm using Gentoo and the only package I was able to find in portage was SER;
I could compile manually but it is harder to upgrade and keep track of 
dependencies.

--
#Joseph

On 10/17/08 22:42, Alex Balashov wrote:

SER is defunct.  Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.

On Fri, October 17, 2008 9:36 pm, Joseph wrote:

 I am running Asterisk and would like to add SER to register my (sip) DID
 and connect it to asterisk;
 but I'm not sure if this is the correct forum.

 I have as DID, sip account with one VoIP provider; currently Im using
 just stand alone SIP phone and register with the VoIP provider via:
 stun.fwdnet.net

 Is it possible to use SER to register with the provider and forward the
 call Asterisk.
 Can anybody provide a link to practical example.

 I'm comfortable with Asterisk but I just install SER and can not find
 appropriate example to follow on www.iptel.org web-page.
 There are a lot explanations but not enough practical examples to follow.



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Re: [asterisk-users] SER + Asterisk

[Joseph]

On 10/17/08 23:23, Kristian Kielhofner wrote:
On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote:

  SER is defunct.  Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.


  Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the
thing to do.

I would gladly go with any of the newer packages if I only could.
I'm just working with what I can find in portage; I'm sure it will be 
eventually available.  It will first show up via overlay.

What I'm trying to do is to register SER to my VoIP provider via 
stun.fwdnet.net and connect SER with Asterisk, I just need some simple 
practical example; and 
upgrade will come with time.
I'm sure it is possible even with old SER.

Suggesting what is newer is not going to help me much :-)

-- 
#Joseph
GPG KeyID: ED0E1FB7

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Re: [asterisk-users] SER + Asterisk

[Grey Man]

As far as I'm aware SER (and it's derivatives) cannot initiate
outbound registraitions. They can do the opposite and act as a SIP
Registrar. For outbound registrations you should be able to use
Asterisk.

Regards,

Greyman.

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Re: [asterisk-users] SER + Asterisk

[ram]

On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] wrote:

 On 10/17/08 23:23, Kristian Kielhofner wrote:
 On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote:
 
   SER is defunct.  Kamailio / OpenSIPS (formerly OpenSER) is the thing to
 do.
 
 
   Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the
 thing to do.

 I would gladly go with any of the newer packages if I only could.
 I'm just working with what I can find in portage; I'm sure it will be
 eventually available.  It will first show up via overlay.

 What I'm trying to do is to register SER to my VoIP provider via 
 stun.fwdnet.net and connect SER with Asterisk, I just need some simple
 practical example; and
 upgrade will come with time.
 I'm sure it is possible even with old SER.

 Suggesting what is newer is not going to help me much :-)



Hi Joseph

you can use UAC Module to register with provider and make calls using
SER/Openser/OpensSIPs

or you can do other way is

SER as registrar and Asterisk act a b2bua ( you can register with provider)

let me know if it helps your need

Ram
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