from:"Martin"

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, May 12, 2015 5:42:57 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds

 Andrew Martin wrote:

 snip

  Joshua,

  As a mitigation for this problem, could I increase the timerb option in
  sip.conf
  to a large value, say 1 hour (instead of the default 32 seconds)? What
  other
  consequences would there be from this change?

 I don't know if chan_sip will allow this, but if it does... it'll keep
 transmitting over and over... it would be better to get to the bottom of
 the problem. Do a packet capture on the machine running Asterisk and see
 where the packet goes. That's the only thing left really. It's also
 possible something got fixed in relation to directmedia between your
 version and latest 11.

Joshua,

Looking at the packet capture from the asterisk server during this time,
I see the following sequence of SIP packets:
INVITE (102) - initial call connecting
RINGING (102) - initial call connecting
RINGING (102) - initial call connecting
OK (102) - initial call connecting
ACK (102) - initial call connecting
OK (102) - initial call connecting (seems like a duplicate OK)
INVITE (103) - re-INVITE to go to bypass mode
ACK (102) - initial call connecting (seems like a duplicate ACK)
INVITE (103) - re-INVITE to go to bypass mode (retry #1)
INVITE (103) - re-INVITE to go to bypass mode (retry #2)
INVITE (103) - re-INVITE to go to bypass mode (retry #3)
INVITE (103) - re-INVITE to go to bypass mode (retry #4)
INVITE (103) - re-INVITE to go to bypass mode (retry #5)

Looking at the logs from the yealink phone (http://pastebin.com/aAWs4j6i),
I see a few differences:
INVITE (102) - initial call connecting
TRYING (102) - initial call connecting
RINGING (102) - initial call connecting
INVITE (102) - initial call connecting (seems like a duplicate INVITE)
RINGING (102) - initial call connecting
OK (102) - initial call connecting
ACK (102) - initial call connecting
INVITE (103) - re-INVITE to go to bypass mode
TRYING (103) - re-INVITE to go to bypass mode
OK (103) - re-INVITE to go to bypass mode
ACK (102) - initial call connecting (seems like a duplicate ACK)
ACK (102) -initial call connecting (seems like a duplicate ACK)
INVITE (103) - re-INVITE to go to bypass mode (retry #1)
ACK (102) -initial call connecting (seems like a duplicate ACK)
INVITE (103) - re-INVITE to go to bypass mode (retry #2)
INVITE (103) - re-INVITE to go to bypass mode (retry #3)
INVITE (103) - re-INVITE to go to bypass mode (retry #4)
INVITE (103) - re-INVITE to go to bypass mode (retry #5)
INVITE (103) - re-INVITE to go to bypass mode
INVITE (103) - re-INVITE to go to bypass mode

Most noteworthy is that the phone seems to send the OK for cseq 103, but it
seems that the asterisk server never received this OK, which is why it kept
re-transmitting the INVITE (103). Is this OK supposed to go to the asterisk
server, or to the other phone? If it is supposed to go to the asterisk server,
I suppose the explanation could be network turbulence prevented this OK from 
getting back to the server - does this seem like what happened? If so, what
should be happening differently to ensure that this call doesn't get dropped?

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, May 13, 2015 10:50:02 AM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds

 Andrew Martin wrote:
  Since some packet loss is a possibility, I assume the protocol has
  mechanisms
  for dealing with it. What should be happening differently in the
  communication
  when packet loss occurs? Should the phone just be re-sending the OK,
  instead of
  printing 0  | ERROR | receive a request with same cseq?? to its log? Or
  should
  Asterisk be starting with a new cseq on each INVITE retry?

 The 200 OK should be retransmitted until an ACK is received. It honestly
 looks like the phone can't talk to Asterisk and it's just generally
 screwing up signaling.

Thanks for the clarification and help debugging this problem. I will work
with the phone vendor to see if they can resolve this from their end. If you
have any other ideas about how to disable re-INVITEs on the asterisk side,
beyond what I have done already, please let me know.

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, May 13, 2015 10:10:25 AM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds

 Andrew Martin wrote:
  - Original Message -

 snip

  Most noteworthy is that the phone seems to send the OK for cseq 103, but it
  seems that the asterisk server never received this OK, which is why it kept
  re-transmitting the INVITE (103). Is this OK supposed to go to the asterisk
  server, or to the other phone? If it is supposed to go to the asterisk
  server,
  I suppose the explanation could be network turbulence prevented this OK
  from
  getting back to the server - does this seem like what happened? If so, what
  should be happening differently to ensure that this call doesn't get
  dropped?

 The traffic is between the phone and Asterisk. As to why, I have no
 idea. The packets aren't getting to Asterisk - that's all I can say. I
 doubt it's network turbulence. Likely getting lost/blocked somewhere.

Since some packet loss is a possibility, I assume the protocol has mechanisms
for dealing with it. What should be happening differently in the communication
when packet loss occurs? Should the phone just be re-sending the OK, instead of
printing 0 | ERROR | receive a request with same cseq?? to its log? Or 
should
Asterisk be starting with a new cseq on each INVITE retry?

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

- Original Message -
 From: Steve Davies davies...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, May 13, 2015 11:39:29 AM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls after 32 seconds

 Hi,

 In my experience, all Yealink phones work just fine with Asterisk, we have
 hundreds (perhaps even low-thousands) out there with customers on Asterisk
 1.2, 1.6.2, 1.8 and 11.

 If you are accurately representing the SIP trace on the phone and the SIP
 trace on Asterisk, then I would strongly suggest a SIP ALG exists in the
 network between the two devices and that SIP ALG does not understand SIP
 properly. The two halves simply do not match, so something must surely be
 interfering.

 In my experience it is often an innocent looking Cisco router. Cisco's SIP
 implementation is SIP By Cisco rather than RFC compliant SIP. If that is
 the case Cisco call it a SIP fixup and you just need to disable it.

 Hope that helps,
 Steve

Steve,

That is an interesting point - the server and the phone are both connected to
Netgear switches where I have enabled their Auto-VoIP feature, which remarks
packets based on protocol (SIP, SCCP, etc) for better QoS:
http://kb.netgear.com/app/answers/detail/a_id/21758

I wonder if this remarking process is modifying another part of the packet too?
Both devices are on the same subnet, so although these switches do route 
traffic as well, that shouldn't be coming into play here.

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

2015-05-12 Thread Andrew Martin

- Original Message -
 From: Andrew Martin amar...@xes-inc.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, May 11, 2015 4:18:58 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds

 - Original Message -
  From: Andrew Martin amar...@xes-inc.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Monday, May 11, 2015 1:35:07 PM
  Subject: Re: [asterisk-users] Retransmission Timeout results in dropped
  calls   after 32 seconds

   That should be all that is required. If that were broken I'd expect
   issue reports to implode - what's the configuration?

  Here's the sip.conf (only showing a single extension since they're all the
  same):
  [general]
  directmedia=no
  directrtpsetup=no
  dtmfmode=rfc2833
  context=asterisk-internal
  allowsubscribe=no
  qualify=no
  disallow=all
  allow=ulaw
  allow=alaw
  allow=gsm
  localnet=10.10.32.0/255.255.248.0
  localnet=192.168.32.0/255.255.255.0

  [146]
  secret=
  host=dynamic
  type=friend

  From the aforementioned sip debug capture, 146 is on the 10.10.32.0/21
  network
  and 113 is on the 192.168.32.0/24 network (these are directly route-able so
  no
  NAT is involved). However, I have now been able to reproduce the problem
  between
  two devices directly on the 10.10.32.0/21 network as well.

 I've gathered the log for this dialog from the SIP phone:
 http://pastebin.com/aAWs4j6i

 What I see is that there's an INVITE for CSeq 103, then an OK for CSeq 103,
 then another INVITE is received for CSeq 103, at which point the phone
 reports an error:
 0 | ERROR | receive a request with same cseq??

 From the asterisk side, it never seems to receive this OK for CSeq 103, hence
 the reason it sends out the INVITE again.

Joshua,

As a mitigation for this problem, could I increase the timerb option in 
sip.conf
to a large value, say 1 hour (instead of the default 32 seconds)? What other
consequences would there be from this change?

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, May 11, 2015 1:24:53 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds

  Could this perhaps be because the phone doesn't support bypass or
  re-INVITEs?
  Is there a way to disable this functionality and instruct asterisk to just
  stay in the middle of the conversation (bridging or native-bridging) for
  the
  duration of the call? I thought that setting directmedia=no and
  directrtpsetup=no would disable re-INVITEs and force asterisk to use
  bridging
  mode, but perhaps something else is required?

 That should be all that is required. If that were broken I'd expect
 issue reports to implode - what's the configuration?

Here's the sip.conf (only showing a single extension since they're all the 
same):
[general]
directmedia=no
directrtpsetup=no
dtmfmode=rfc2833
context=asterisk-internal
allowsubscribe=no
qualify=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
localnet=10.10.32.0/255.255.248.0
localnet=192.168.32.0/255.255.255.0

[146]
secret=
host=dynamic
type=friend

From the aforementioned sip debug capture, 146 is on the 10.10.32.0/21 network
and 113 is on the 192.168.32.0/24 network (these are directly route-able so no
NAT is involved). However, I have now been able to reproduce the problem between
two devices directly on the 10.10.32.0/21 network as well.

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, May 11, 2015 12:32:06 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds

 Andrew Martin wrote:
  - Original Message -

 snip

  By doing a number of test calls today, I have managed to reproduce this
  while
  sip debugging was on, so I have that information available now as well:
  http://pastebin.com/ZJqzdvY3

  This was a call from 113 to 146 via a queue. Note that the asterisk server
  is
  at 10.10.32.251. I see the following:
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  SIP/2.0 180 Ringing
  SIP/2.0 180 Ringing
  SIP/2.0 200 OK
  ACK sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  SIP/2.0 200 OK
  ACK sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0

  This appears to start out with a successful SIP conversation (ending with
  the
  first ACK), so it is unclear to me why we have two new sets of INVITEs sent
  afterwards.

 Asterisk has sent a re-INVITE to have the media flow directly. The
 device (seems) to respond with the 200 OK (you can tell based on the
 CSeq) for the initial INVITE, and not for the re-INVITE. As Asterisk
 gets no response to its re-INVITE it gives up and terminates the dialog.

Could this perhaps be because the phone doesn't support bypass or re-INVITEs?
Is there a way to disable this functionality and instruct asterisk to just 
stay in the middle of the conversation (bridging or native-bridging) for the 
duration of the call? I thought that setting directmedia=no and 
directrtpsetup=no would disable re-INVITEs and force asterisk to use bridging
mode, but perhaps something else is required?

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

- Original Message -
 From: Andrew Martin amar...@xes-inc.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, May 11, 2015 1:35:07 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds

  That should be all that is required. If that were broken I'd expect
  issue reports to implode - what's the configuration?

 Here's the sip.conf (only showing a single extension since they're all the
 same):
 [general]
 directmedia=no
 directrtpsetup=no
 dtmfmode=rfc2833
 context=asterisk-internal
 allowsubscribe=no
 qualify=no
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 localnet=10.10.32.0/255.255.248.0
 localnet=192.168.32.0/255.255.255.0

 [146]
 secret=
 host=dynamic
 type=friend

 From the aforementioned sip debug capture, 146 is on the 10.10.32.0/21
 network
 and 113 is on the 192.168.32.0/24 network (these are directly route-able so
 no
 NAT is involved). However, I have now been able to reproduce the problem
 between
 two devices directly on the 10.10.32.0/21 network as well.

I've gathered the log for this dialog from the SIP phone:
http://pastebin.com/aAWs4j6i

What I see is that there's an INVITE for CSeq 103, then an OK for CSeq 103,
then another INVITE is received for CSeq 103, at which point the phone
reports an error:
0 | ERROR | receive a request with same cseq??

From the asterisk side, it never seems to receive this OK for CSeq 103, hence
the reason it sends out the INVITE again.

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

- Original Message -
 From: Andrew Martin amar...@xes-inc.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Friday, May 8, 2015 5:12:28 PM
 Subject: [asterisk-users] Retransmission Timeout results in dropped calls   
 after 32 seconds

 Hello,

 I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All
 the SIP clients are on a LAN, so no NAT is involved. I have been experiencing
 an intermittent problem where a call will be successfully answered, but then
 dropped by Asterisk 32 seconds after it is answered (with a Retransmission
 timeout reached on transmission error). Here is an example of this happening
 in the asterisk console:
 http://pastebin.com/7LDwHAJe

 This problem only happens a fraction of the time, so I have been unable to
 enable SIP debugging before it happens to get a capture. However, usually the
 caller will just call back immediately and then the call will work without a
 problem. It sounds like SIP Timer B is what causes the call to be dropped if
 an
 ACK to the INVITE is not received within 32 seconds. How can I determine if
 this is the case and how can I resolve this Retransmission timeout problem?

 Here is my sip.conf:
 general]
 directmedia=no
 directrtpsetup=no
 dtmfmode=rfc2833
 context=internal
 allowsubscribe=no
 qualify=no
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 localnet=10.10.32.0/255.255.248.0

 [123]
 secret=11
 host=dynamic
 type=friend

By doing a number of test calls today, I have managed to reproduce this while
sip debugging was on, so I have that information available now as well:
http://pastebin.com/ZJqzdvY3

This was a call from 113 to 146 via a queue. Note that the asterisk server is
at 10.10.32.251. I see the following:
INVITE sip:146@10.10.32.96:5062 SIP/2.0
SIP/2.0 180 Ringing
SIP/2.0 180 Ringing
SIP/2.0 200 OK
ACK sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
SIP/2.0 200 OK
ACK sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0

This appears to start out with a successful SIP conversation (ending with the
first ACK), so it is unclear to me why we have two new sets of INVITEs sent
afterwards. 

Also in case it is relevant, the asterisk server has two NICs set up in a bond
with bond-mode 1 (active/backup).

Does this additional debug information provide any clues to why this 
intermittent retransmission timeout error is occurring?

Thanks,

Andrew

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[asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

2015-05-08 Thread Andrew Martin

Hello,

I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All
the SIP clients are on a LAN, so no NAT is involved. I have been experiencing
an intermittent problem where a call will be successfully answered, but then
dropped by Asterisk 32 seconds after it is answered (with a Retransmission
timeout reached on transmission error). Here is an example of this happening
in the asterisk console: 
http://pastebin.com/7LDwHAJe

This problem only happens a fraction of the time, so I have been unable to
enable SIP debugging before it happens to get a capture. However, usually the
caller will just call back immediately and then the call will work without a
problem. It sounds like SIP Timer B is what causes the call to be dropped if an
ACK to the INVITE is not received within 32 seconds. How can I determine if
this is the case and how can I resolve this Retransmission timeout problem?

Here is my sip.conf:
general]
directmedia=no
directrtpsetup=no
dtmfmode=rfc2833
context=internal
allowsubscribe=no
qualify=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
localnet=10.10.32.0/255.255.248.0


[123]
secret=11
host=dynamic
type=friend


Thanks!

Andrew Martin

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Re: [asterisk-users] Phones don't stop ringing when queue is answered

2015-05-07 Thread Andrew Martin

James,

The WaitExten()s just provide a pause between the two Queue() calls to
let the first group of phones finish ringing. In this example I am ringing
the same group (queue_level_1) twice, however in a real-world scenario I 
would ring queue_level_1 and then ring queue_level_2 which each have a 
different list of phones.

Thanks,

Andrew

- Original Message -
 From: James Thomas jthomas...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, May 7, 2015 10:20:10 AM
 Subject: Re: [asterisk-users] Phones don't stop ringing when queue is answered
 
 What purpose do the WaitExten()s serve here? Are you really allowing the
 caller to connect to different extensions in the test-queue context? Have
 you tried without the WaitExten()s?
 
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Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

2015-05-05 Thread Andrew Martin



- Original Message -
 From: Guenther Boelter gboel...@gmail.com
 To: asterisk-users@lists.digium.com
 Sent: Tuesday, May 5, 2015 1:05:44 AM
 Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
 
  Looking into it further, in my case it does not appear to be a
  NATing issue, since running OpenVPN from pfSense means there's no
  NATing occurring between the clients or between the clients and the
  asterisk server.
  
  Although I was unable to reproduce the problems, I did notice some
  packet loss and jitter in sip show channelstats, here is a
  sample: Peer Call ID  Duration Recv: Pack  Lost
  ( %) Jitter Send: Pack  Lost   ( %) Jitter
  192.168.32.26446613544@1  00:03:03 94  004238
  (97.83%) 0. 00  000244 ( 0.00%) 0.
  192.168.32.385b2ebdc92fd  00:03:03 59  01 (
  1.67%) 0. 00  91 ( 0.00%) 0.0028
  
  I was unable to find documentation each of these columns, but the
  high percentage of loss for received packets for 192.168.32.26
  seems suspicious. Do these statistics indicate a problem?
  
  Thanks,
  
  Andrew
 
 Hi Andrew,
 
 is this a linux machine? If so, check your NIC with ifconfig for
 hardware errors.
 
 Guenther
 

Guenther,

Yes, this machine is running CentOS 6.4 (see my original post for more
details). This asterisk server has 2x gigabit NICs set up in a bond with
bond mode 1.

Both ifconfig and ethtool do not report any hardware errors,
although they do show a few checksum errors:
eth0  Link encap:Ethernet  HWaddr 00:11:22:33:44:55
  UP BROADCAST RUNNING SLAVE MULTICAST  MTU:1500  Metric:1
  RX packets:467927100 errors:0 dropped:0 overruns:1 frame:0
  TX packets:304724661 errors:0 dropped:0 overruns:0 carrier:0
  collisions:0 txqueuelen:1000
  RX bytes:131747094082 (122.6 GiB)  TX bytes:93869585242 (87.4 GiB)
  Memory:fb92-fb94

eth1  Link encap:Ethernet  HWaddr AA:BB:CC:DD:EE:FF
  UP BROADCAST RUNNING SLAVE MULTICAST  MTU:1500  Metric:1
  RX packets:41250363 errors:0 dropped:0 overruns:0 frame:0
  TX packets:3467 errors:0 dropped:0 overruns:0 carrier:0
  collisions:0 txqueuelen:1000
  RX bytes:5190889937 (4.8 GiB)  TX bytes:1594075 (1.5 MiB)
  Memory:fb90-fb92

From ethtool -S eth0:
 tx_smbus: 164709
 rx_smbus: 119082408
 dropped_smbus: 104036

 rx_queue_0_packets: 97532982
 rx_queue_0_bytes: 16800645524
 rx_queue_0_drops: 1
 rx_queue_0_csum_err: 0
 rx_queue_0_alloc_failed: 0

 rx_queue_7_packets: 53850556
 rx_queue_7_bytes: 12797600155
 rx_queue_7_drops: 0
 rx_queue_7_csum_err: 41
 rx_queue_7_alloc_failed: 0

Thanks,

Andrew

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Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

2015-05-04 Thread Andrew Martin

- Original Message -
 From: Administrator TOOTAI ad...@tootai.net
 To: asterisk-users@lists.digium.com
 Sent: Friday, May 1, 2015 6:42:38 AM
 Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

 Le 01/05/2015 00:05, Andrew Martin a écrit :
  - Original Message -
  From: Administrator TOOTAI ad...@tootai.net
  To: asterisk-users@lists.digium.com
  Sent: Thursday, April 30, 2015 4:43:33 PM
  Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call
  In

  I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and
  internal phones are located on the 10.10.32.0/21 LAN subnet. I have many
  internal SIP phones, which appear to be working correctly. I have a few
  external phones (Yealink SIP-T32G or other Yealink model) on
  192.168.32.0/24 which have an OpenVPN client configured on them that
  connects back to the LAN network through a pfSense gateway with OpenVPN
  configured on it.

  I faced problems with pfsense -no VPN involved- and finally installed
  siproxd on it. Also set the firewall mode to conservative.

  Daniel,

  Thanks for the information. Do you have an example or documentation on the
  siproxd configuration that you used?

 No, just follow the basis of the parameters given by the package. If I
 remember, SIP use the proxy siproxd and RTP is direct.

Looking into it further, in my case it does not appear to be a NATing issue,
since running OpenVPN from pfSense means there's no NATing occurring between
the clients or between the clients and the asterisk server.

Although I was unable to reproduce the problems, I did notice some packet loss
and jitter in sip show channelstats, here is a sample:
Peer Call ID  Duration Recv: Pack  Lost   ( %) Jitter 
Send: Pack  Lost   ( %) Jitter
192.168.32.26446613544@1  00:03:03 94  004238 (97.83%) 0. 
00  000244 ( 0.00%) 0.
192.168.32.385b2ebdc92fd  00:03:03 59  01 ( 1.67%) 0. 
00  91 ( 0.00%) 0.0028

I was unable to find documentation each of these columns, but the high 
percentage
of loss for received packets for 192.168.32.26 seems suspicious. Do these 
statistics
indicate a problem?

Thanks,

Andrew

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[asterisk-users] OpenVPN Clients Intermittently Cannot Call In

2015-04-30 Thread Andrew Martin

Hello,

I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal 
phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP 
phones, which appear to be working correctly. I have a few external phones 
(Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an 
OpenVPN client configured on them that connects back to the LAN network through 
a pfSense gateway with OpenVPN configured on it. 

Asterisk server LAN IP: 10.10.32.10
My internal test phone: 146 at 10.10.32.96
My external test phone: 265 at 192.168.32.10

My sip.conf for these external users is as follows:
http://pastebin.com/2b9YE7Dz


The dialplan uses this Dial() invocation when dialing either an internal or 
external phone. Note that the max timeout is 12 seconds:
exten = _[12]XX,1,Dial(SIP/${EXTEN},12)


These external phones register correctly, and internal users can call these 
external users, the phones ring immediately, and the call is normal. However, 
if the external users try to dial an internal phone, I've observed some 
different failure modes:
* operating normally: sometimes the call rings immediately, the internal user 
answers, and the audio is present immediately
* ringing delay and no connection even after pickup: sometimes there's a 
significant delay between when the call starts ringing on the external side 
and when it actually starts ringing on the internal user's phone. Consequently, 
the internal user only has 1 or 2 rings to answer. Even if they do answer 
during this time, the line is dead and it goes to voicemail (the next step in 
the dialplan)
* delay before audio is connected after answer: sometimes the internal user 
answers, but there's a delay of 3-10 seconds before either party can hear audio

I've enabled rtp and sip debug for this particular external phone 
(192.168.32.10) and attached console logs from both types of these failures:
* ringing delay and no connection even after pickup: 
http://pastebin.com/fe1khEmF
* delay before audio is connected after answer: http://pastebin.com/uZSMKczk

What else can I try to debug these problems? Since it is intermittent, I am not 
always able to reproduce (sometimes the calls work just fine).

Thanks,

Andrew Martin

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Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

2015-04-30 Thread Andrew Martin

- Original Message -
 From: Administrator TOOTAI ad...@tootai.net
 To: asterisk-users@lists.digium.com
 Sent: Thursday, April 30, 2015 4:43:33 PM
 Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

  I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and
  internal phones are located on the 10.10.32.0/21 LAN subnet. I have many
  internal SIP phones, which appear to be working correctly. I have a few
  external phones (Yealink SIP-T32G or other Yealink model) on
  192.168.32.0/24 which have an OpenVPN client configured on them that
  connects back to the LAN network through a pfSense gateway with OpenVPN
  configured on it.

 I faced problems with pfsense -no VPN involved- and finally installed
 siproxd on it. Also set the firewall mode to conservative.

Daniel,

Thanks for the information. Do you have an example or documentation on the
siproxd configuration that you used?

Thanks,

Andrew

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[asterisk-users] customizing Asterisk CLI

2015-01-25 Thread Martin Vegter

Hello,

when I am in the Asterisk CLI, I can exit with 'exit' or 'quit'. Ctrl+d
has no effect. Is there any way to bind Ctrl+d to exit/quit ?

Also, when I am in asterisk CLI, I can use command history and readline
functions such as CTRL+r to search. But not all functions are available.
For example, the alternate mappings for page up and page down to
search the history do not work. They work in everything else (bash,
mysql, ..)

$ cat /etc/inputrc
\e[5~: history-search-forward
\e[6~: history-search-backward

is there a way to make it work in asterisk ?

I am using Asterisk 11.13 on Debian Wheezy.

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Re: [asterisk-users] Do public wifi block IAX port 4569

2014-12-27 Thread Martin Lima

 But when I catch open wifi network in a mall or a Tim Horton zoiper fail
 to register.
 
 Do they block IAX port 4569 or IAX protocol?

Sometimes they use a Captive portal, you have to open a webpage first and 
confirm some agreement. Until you do, all the packets except those being 
redirected to this are dropped.
Sometimes they only allow some ports, eg. 80, 443, or protocols (http...)
Martin

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[asterisk-users] asterisk prompt

2014-11-25 Thread Martin Vegter

Hello,

I am trying to set up color prompt. In the documentation I have found this:

  %Cn[;n] Change terminal foreground (and optional background) color to
  specified. A full list of colors may be found in include/asterisk/term.h*

But nowhere could I find what format the color code should be. I have
tried all possible permutations, none of them works:

  ASTERISK_PROMPT=%Cn[COLOR_BLUE] %H:  asterisk -vvr
  ASTERISK_PROMPT=%Cn[32;128] %H:  asterisk -vvr
  ASTERISK_PROMPT=%Cn[32;] %H:  asterisk -vvr
  ASTERISK_PROMPT=%Cn[;32] %H:  asterisk -vvr
  ASTERISK_PROMPT=%Cn[;COLOR_CYAN] %H:  asterisk -vvr
  ASTERISK_PROMPT=%Cn[32|128] %H:  asterisk -vvr

can somebody please tell me how to make my asterisk prompt red, for example?
__

Also, I would like to piggyback a second question:

I am using Asterisk 11.13 on Debian Wheezy. When I am in asterisk CLI, I
can use command history and readline functions such as CTRL+r to search.
But not all functions are available. For example, the alternate mappings
for page up and page down to search the history do not work. They
work in everything else (bash, mysql, ..)

$ cat /etc/inputrc
\e[5~: history-search-forward
\e[6~: history-search-backward

is there a way to make it work in asterisk ?

thanks,
Martin


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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

2014-09-09 Thread Martin Lima

Dne St 3. září 2014 10:49:41, Khalid Touati napsal(a):
 I will be trying with H263+ and let you guys know, thank you for the good
 news! I almost lost hope in Asterisk enabling me to use video (with
 identical softphones using same codec :) )
 
It only worked for me when I had only one video codec allowed. (h.264 in my 
case) Not only there is no trnscoding available but also video codec 
negotiation doesn't (or didn't...) work properly.
Martin

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[asterisk-users] Internal calls without voice transport

2014-07-28 Thread martin f krafft

Hey,

we're experiencing a weird problem with Asterisk 1.8.13.1
(1:1.8.13.1~dfsg1-3+deb7). Calls that leave and enter Asterisk via
a PBX (sipgate.de) work perfectly fine, almost 100% of the time.

However, calls that are routed to sipgate.de, which then routes the
call back to our Asterisk instance are silent most of the time.

What I mean with that is that even though RTP traffic flows, neither
side can hear anything from the other.

This problem happens when people at site A dial someone at site
B using the number provided by sipgate.de, but also if people call
each other within a site through the external number, i.e. if I dial
089-1234567-100 from 089-1234567-200.

I have not been able to reproduce this problem with purely internal
calls, i.e. calling ext. 100 directly, so I am assuming there's
a problem due to sipgate's involvement. However, as far as
I understand, once the call is established (and both parties' phones
suggest that), the traffic flows only via Asterisk (directmedia
= update,nonat), so the problem is likely to be found there, no?

Before I shower you with debug logs and traces, I am wondering if
this sounds familiar to anyone…?

Thanks,

--
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we would have been born that way.

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Re: [asterisk-users] Internal calls without voice transport

2014-07-28 Thread martin f krafft

By chance, I managed to fig into this a bit and found the exact
moment when audio stops. It is exactly the moment when the
counterparty picks up and RTP debug output says:

Got RTP packet from46.244.255.146:8058 (type 00, seq 000680, ts
340914880, len 000160)
Sent RTP packet to 46.244.255.146:8058 (type 00, seq 026000, ts
3578986600, len 000160)
-- SIP/lehel-sipgate-3573 answered SIP/lehel-martin-3572
-- Remotely bridging SIP/lehel-martin-3572 and
SIP/lehel-sipgate-3573
Sent RTP P2P packet to 46.244.255.146:8058 (type 08, len 000160)
Sent RTP P2P packet to 46.244.255.146:8058 (type 08, len 000160)

so RTP switches to RTP P2P and no more packets are received from the
phone.

I did have a sniffer running on 46.244.255.146, and Wireshark really
rocks, so now I know that the gateway firewall is at fault, and
indeed, for some reason, nf_conntrack_sip and nf_nat_sip were not
loaded. Now I am wondering how it worked in the first place, but
that's that. Maybe this will fix things.

Anyway, I don't quite yet understand what RTP P2P packets are or why
they are sometimes used and not at other times. I assume they are
packets intended to be exchanged directly between the two clients,
but since I have MixMonitor() on Asterisk, this shouldn't actually
be possible as Asterisk should always force itself into the middle.

Thoughts?

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dies ist eine manuell generierte email. sie beinhaltet
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[asterisk-users] modify from field sip headers

2014-03-18 Thread Luis San Martin

Hello,

Im trying to modify the 'From' field in my sip headers in order to include
extra info (user=tel) as it follows:

From : sip:005114824403@200.91.0.146;user =tel

However asterisk is still doing this header:

 sip:111@1.1.1.1;tag=as167b4b82

Is there a way to accomplish this? Ive been also looking up AGI but without
any success. Ideas are welcome!

Kind regards
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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-23 Thread Martin




in the sip.conf i specified

[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes


There is a typo in the last line above. Should be canreinvite. AFAIK it's 
obsoleted in favor of directmedia. BTW, try to set it to NO.
BTW, what is the codec order? Fax detection doesn't work reliably over 
compressed codecs (g729 etc...), in my case didn't work at all...

try to add:
directmedia=no
disallow=all
allow=ulaw
allow=alaw

to your peer definition.

Martin 



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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

2013-12-13 Thread Martin

If I need to use SIP, from where to get the suitable firmware for these Cisco 
IP Phones 7942G?



Be careful, not all versions of SIP firmware work with asterisk. I do have 8-3-1 
(cmterm-7941_7961-sip.8-3-1)here and it works just fine with my 7961. Downloaded 
somewhere. Version 9.x is broken, SIP only works over TCP.



Where do u download the SIP firmware usually for your Cisco IP Phones?


Search for cmterm-7941_7961-sip.8-3-1.zip
I also have some other files here but I don't remember what was the reason for 
them :-(


Martin


Your kindly help is highly appreciated.
Regards
Bilal



I'm using the sip firmware.. It's alright.. I feel like I'm
not receiving
all the features I should.. But MWI works and multiple call
appearance..



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[asterisk-users] language specific email templates

2013-06-18 Thread Thomas Martin

Hi,

I am new to Asterisk. I'm using it behind a kamailio sip-router to provide
voicemail boxes to sip-users.
I followed these instruction:
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x
to set everything up, using ARA with a MySQL DB.

After a few tweaks everything is basically working, however, a few questions
remain that I could not find
clear answers for - anywhere. Maybe some of the experts can help a little ...

So far, my Asterisk DB features 1 table (voicemessages) and 2 views
(vmusers and sipusers).

vmusers shows the following columns: uniqueid, customer_id, context, mailbox,
password, fullname, email, pager, stamp
(sipusers shows: name, defaultuser, type, secret, host, callerid, context,
mailbox, nat, qualify, fromuser, fromuser, authuser, fromdomain, insecure,
canreinvite, disallow, allow, restrictcid, defaultip, ipaddr, port, regseconds)

Questions:

1) In vmusers, the content for column fullname is composed from 2 columns
(i.e. first_name, last_name) of the kamailio subscriber table
that do not even natively exist there, but were manually added by me (to
comply with the instructions).
Q1a: Is the fullname column in the vmusers view mandatory for Asterisk to
function properly? fullname might always stay empty...
Q1b: Is the stamp column in the vmusers view mandatory for Asterisk to
function properly?

2) in my world, sip-users have a language. At certain points, Asterisk will
communicate with my users regarding voicemail.
e.g.: a) Asterisk delivers the message left as a wav-attachment by email, b)
VoicePrompts guide the user through the voicemail-menu
So far, I am able to hard-code the language for the voiceprompts in
extension.conf with Set(CHANNEL(language)=fr). But this set the language for
everybody.
Also, you can customise the email template ($emailbody, etc) for voicemail
delivery in voicemail.conf. However, this is also covers one language only.
Q2a:How can configure Asterisk I pick the voicemail-prompts of the
respective sip-users language?
Q2b:Is it useful for that purpose, to add a language column to the
vmusers view ?
Q2c:Where can language-specific email-templates for voicemail delivery be
supplied and how can be achieved that the correct one is used ?

3) I would also like to use Asterisk as a SIP client (since Kamailio can't do
this). Each vmuser may have one or many SIP accounts with foreign registrars.
I would like Asterisk to register those accounts with their registrars and
forward incoming calls to the kamailio (parent) account that the vmuser owns.
(The kamailio (parent) account -in turn- shall forward this call to the user's
asterisk voice-box, if the user is busy/not-registered/etc ).
Q3a:How can I use ARA to configure Asterisk as a SIP client to act in the
described fashion ?
Q3b:Is it useful/necessary for that purpose, to host dial-plans in the
MySQL-DB as well ?

Q4: Can you point to some documentation that explains ARA a little more in
depth and possibly illustrates a few examples?

For any hint - thank you very much in advance!

Best regards,

-Thomas

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[asterisk-users] Detecting fax without Aswer()ing the call first?

2013-02-24 Thread Martin

Trying to make the fax detection work. My current setup (with no fax) is done 
without Answer(), so the call is answered only when someone actually picks-up 
the phone. But when the incoming call is fax, I can her the tone and call is 
never forwarded to Fax extension.

But... Strange thing happens when I (mistakenly) put a call on hold:

   -- Executing [youngandson-test@incoming:2] 
Gosub(SIP/66.193.176.35-00b8, process-callerid,s,1) in new stack
   -- Executing [s@process-callerid:1] Verbose(SIP/66.193.176.35-00b8, 
3,- Original CallerID: FREE CALL TOLL 18009806858 ) in new stack

   -- - Original CallerID: FREE CALL TOLL 18009806858
   -- Executing [s@process-callerid:2] GotoIf(SIP/66.193.176.35-00b8, 
1?4) in new stack

   -- Goto (process-callerid,s,4)
   -- Executing [s@process-callerid:4] GotoIf(SIP/66.193.176.35-00b8, 
0?8) in new stack
   -- Executing [s@process-callerid:5] GotoIf(SIP/66.193.176.35-00b8, 
0?8) in new stack
   -- Executing [s@process-callerid:6] GotoIf(SIP/66.193.176.35-00b8, 
1?7:8) in new stack

   -- Goto (process-callerid,s,7)
   -- Executing [s@process-callerid:7] Set(SIP/66.193.176.35-00b8, 
CALLERID(num)=18009806858) in new stack
   -- Executing [s@process-callerid:8] Return(SIP/66.193.176.35-00b8, ) 
in new stack
   -- Executing [youngandson-test@incoming:3] 
Macro(SIP/66.193.176.35-00b8, stdexten,210,sip/ra2501) in new stack
   -- Executing [s@macro-stdexten:1] Dial(SIP/66.193.176.35-00b8, 
sip/ra2501,360) in new stack

 == Using SIP RTP CoS mark 5
   -- Called sip/ra2501
   -- SIP/ra2501-00b9 is ringing
[2013-02-24 17:05:12] WARNING[6554]: chan_sip.c:8979 process_sdp: ignoring 
'video' media offer because port number is zero

   -- SIP/ra2501-00b9 answered SIP/66.193.176.35-00b8
   -- Locally bridging SIP/66.193.176.35-00b8 and SIP/ra2501-00b9
[2013-02-24 17:05:31] WARNING[6554]: chan_sip.c:8979 process_sdp: ignoring 
'video' media offer because port number is zero
[2013-02-24 17:05:31] WARNING[6554]: chan_sip.c:8945 process_sdp: ignoring 
'audio' media offer because port number is zero
   -- Started music on hold, class 'default', on channel 
'SIP/66.193.176.35-00b8'
[2013-02-24 17:05:31] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll() 
failed: Interrupted system call
[2013-02-24 17:05:31] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll() 
failed: Interrupted system call




[2013-02-24 17:05:34] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll() 
failed: Interrupted system call
[2013-02-24 17:05:34] WARNING[6532]: res_musiconhold.c:659 monmp3thread: poll() 
failed: Interrupted system call
 == Redirecting 'SIP/66.193.176.35-00b8' to fax extension due to CNG 
detection

   -- Stopped music on hold on SIP/66.193.176.35-00b8
 == Spawn extension (incoming, fax, 1) exited non-zero on 
'SIP/66.193.176.35-00b8' in macro 'stdexten'
 == Spawn extension (incoming, fax, 1) exited non-zero on 
'SIP/66.193.176.35-00b8'
   -- Executing [fax@incoming:1] Gosub(SIP/66.193.176.35-00b8, 
receive-fax,fax,1) in new stack
   -- Executing [fax@receive-fax:1] Verbose(SIP/66.193.176.35-00b8, 
3,Incoming fax from 18009806858) in new stack

   -- Incoming fax from 18009806858
   -- Executing [fax@receive-fax:2] Set(SIP/66.193.176.35-00b8, 
FAXDEST=/var/spool/fax/incoming) in new stack
   -- Executing [fax@receive-fax:3] Set(SIP/66.193.176.35-00b8, 
FAX-FILENAME=20130224-170534 Incoming Fax) in new stack
   -- Executing [fax@receive-fax:4] ReceiveFAX(SIP/66.193.176.35-00b8, 
/var/spool/fax/incoming/20130224-170534 Incoming Fax.tif) in new stack
   -- Channel 'SIP/66.193.176.35-00b8' receiving FAX 
'/var/spool/fax/incoming/20130224-170534 Incoming Fax.tif'

 == Using UDPTL CoS mark 5
 == Spawn extension (receive-fax, fax, 4) exited non-zero on 
'SIP/66.193.176.35-00b8'


Fax is suddenly detected and received! (Im not sure why all these warnings came 
up, something misconfigured in music on hold...)


Is there any way to make Asterisk listen for CNG tone during the connected 
call, eliminating the need for Answer() and Wait()?
Is the fax detection completely impossible when compressed codec (g729, gsm...) 
is in use? I've read its unreliable but does not work at all for me.

(Asterisk 1.8.13 installed from Debian repository)

Thanks 
Martin 



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Re: [asterisk-users] OT - Desktop SIP phone with OpenVPN client

2013-01-21 Thread Anselm Martin Hoffmeister


Am 21.01.2013 14:21, schrieb Olivier:

Hello,

I've seen some desktop SIP phones (Snom, Yealink) intregrate a VPN 
(OpenVPN ?) client.


Has someone experience to share about that particular feature ?
Is this experience rather successful ?

My underlying question is can one supervise and configure these 
desktop phones, in teleworking environment ?

Is DHCP required ?


With Snom phones, those need an underlying network connection (d'oh, you 
wouldn't guess :-). That can be configured just
like you are used to do it with snom phones - DHCP, fixed IP, whichever 
you like. They also need a reachable NTP server.


Then they will ( after booting) download the VPN config from your 
(hopefully protected) server and connect to the OpenVPN
server. Address assignment on the VPN link is done by the OpenVPN 
internal mechanism. You will be able to reach the
phone's web interface, afaik, both over its local address and the 
OpenVPN assigned one.


Make sure to either have your PBX on the machine with the OpenVPN daemon 
or add appropriate route configuration to the OpenVPN client config.


BR
AMH

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Re: [asterisk-users] Recorded reminders

2013-01-13 Thread Anselm Martin Hoffmeister


Am 13.01.2013 03:17, schrieb Adolphus Enaboifo:

Hi List Members ,
its been about one months since I built my first Asterisk server.
What I want to know is: are there ways to make Asterisk take recorded
reminders.
This is the scenario I have in mind.

1 You place a call to a specific extension say 350.

2 On recognizing the incoming extension the reminder application at
extension 350 prompts you to enter a number say 1 to record a message to
your profile as well as input the time   when the application will
call you to playback the message.

3 or enter another number say 0 to playback all your recorded reminders
in your profile.with options to add to the list or delete from the list.

and of course there will be a limit on the amount of messages per user.

Please if such applications exist can you guys show me how to configure it.


Hi Adolphus,

this sounds like something that a little scripting + call files can do. 
There may be better ways (and others may point those out in a flash), 
but this is what springs to my mind:


- Have one directory for recorded audio for each user. Name audio files
  for their target time, like 201301131500.wav
- When someone calls 350 and presses 1, check the number of files in
  that directory. If more than MAXMSGS, deny.
- If there is already a recording for the target time, deny.
- Else prompt for target day (today, tomorrow...) and time. If a file
  for that date/time exists, deny.
- Record file, move to the right directory
- Create a call file on the same filesystem as the spool directory
- touch it for the target date/time and move into call files directory

You'd need some nice scripting later on to handle that outgoing call.
If the messages is read (and possibly acknowledged by pressing 1 or
the like), the sound file should be deleted.

If either the call fails or the acknowledgement is not given, the
sound file should be re-named to a new time (say, one hour later)
and a new call file generated.

A few things that also should be thought about:
- To not have endless reminder calls over and over, you could have
  a failed delivery counter per user - once that reaches a
  certain threshold, say 5, the reminders can be emailed to the
  user and deleted from the spool. You can reset that counter if
  the counter file has a change date older than 2 days with a
  cronjob - so if no failed deliveries happen within 3 days or so,
  they will be activated again. Make sure the user is informed
  about this problem iff a file exists when he calls in to
  record a new message.
- Do sane error checking. When the call file is fired and fails to
  find the wav file it expects, this should not trigger another
  call. Perhaps an email avoid endless loops.
- It might be a good idea to have a variation in the touch to
  the call file such that the expected time is only precise in
  minutes. Like add a random number of seconds in the range
  (0...50, or even -180 to 180 if precision is not essential).
  Be sure to document that or users might complain that the
  telephone system clock is not space-age-precise (Lusers!)
  This should get around everyone wanting to be reminded of
  going home in time for the soccer match, and everyone typing in
  a reminder time of 1630.
- You should monitor usage; there can be still quite a lot of
  calls to interesting times. Same problem that automatic window
  blinds have: If everyone sets the DCF controlled clock to open
  the shutters at 8:00 precisely, the start current of possibly
  many motors may be _noticeable_ for the power company. That is
  why those devices do not have and do not need clocks with ultra-
  high precision - some even vary the morning/evening action
  time by several minutes on purpose.

BR
AMH




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Re: [asterisk-users] Playing music through VoIP handsets while on hook

2013-01-10 Thread Anselm Martin Hoffmeister


Am 11.01.2013 02:42, schrieb Christopher Harrington:


Wow, that seems wildly bandwidth inefficient. Is it possible to do
multicast VoIP?



Snom phones[*] do support multicast streaming. You can setup an
IP  port combination that the phone will accept audio at; once
stream data starts arriving, the phone will start playback.

[*] and possibly others as well, but that is what I have on my desk.

Reasonably multicast will be ignored during a call though.

AFAIK Asterisk supports Page to multicast. VLC or the like may
also be audio sources.

BR
AMH



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Re: [asterisk-users] Impromptu conferencing

2012-12-03 Thread martin f krafft

also sprach Brandon B. bran...@brellsystems.com [2012.12.03.0132 +0100]:
 [all-inbound-for-999]
 ; inbound extension through a conference room
 exten = 999,1,MeetMeCount(999,COUNT-999);
 exten = 999,2,GotoIf($[${COUNT-999}=1]?10);
 exten = 999,3,Dial(SIP/99,999,G(6));
 exten = 999,4,Hangup;
 exten = 999,6,MeetMe(999,FAqx);
 exten = 999,7,MeetMe(999,Fqx);
 
 ; bypass the conference room for multiple inbound calls
 exten = 999,10,Dial(SIP/999);

This is an interesting approach, but I am still not sure how to add
the third party. Sure, I can call them up and tell them to dial
a number, but I'd really rather be able to just switch them in.

What would need to be done for a user to e.g. suspend the
conference, dial another number and finally merge the channels? Do
I need the manager API for that, like this:

  http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe#Mergingconferences

?

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Re: [asterisk-users] Managing complex setups with Asterisk

2012-11-16 Thread martin f krafft

also sprach Raj Mathur (राज माथुर) r...@linux-delhi.org [2012.11.16.1005
+0100]:
Warning: Not a fan of using whitespace as semantic markup, so no Django
this side. Fine with Perl or Java, though.

As long as we can agree on using a database (i.e. no MySQL) or the
filesystem (Git…), then the question of which language to use for
a frontend is secondary. I wouldn't chose Java myself, but I suspect
that the job is enough text processing that Perl would actually be
a sensible choice — except I won't help since I don't know it well.

But shouldn't the first step be a mixture of database design and
requirement specification?

I would like a solution that keeps users, sites, and numbers
(belonging to trunks (hardware, as well as SIP)) separate and then
basically allows for free combinations.

User A might have a desk at site I, to which a range of numbers is
assigned, and in addition to an internal number (e.g. a one digit
site prefix followed by a two digit number, or a site-independent
number assigned per person), one of those externals rings at A's
desk.

User B might roam between sites I and II and either should have the
same internal/external numbers ringing at both desks, or require
some sort of login to let the system know where to ring.

User C might have a desk with a phone at site II, but is out most of
the time, and calls should also ring on his/her cell.

User D has a smart phone and wants both his desk and the smart phone
to ring.

All users want voicemail and be able to configure the time until
voicemail answers.

During vacation etc., a forwarding number should be configurable.

Some users might want their voicemail to say e.g. press 1 now to be
transferred to my cell.

We would also want to be able to specify per-user whether to use
UDP, TCP or IAX, who can transfer and park calls, who can record
them with mix monitor, who can create ad-hoc conferences, their
language, who has a video telephone…

… and of course there ought to be a way to set user-specific
sip.conf settings.

On top, it would be nice if there were some sort of group
inheritance. This sounds a bit like LDAP, except LDAP can't actually
do it. What I mean is that I'd really like to define a group of e.g.
managers who all have internal numbers beginning with 11 and
secretaries who can create conferences, and then associate users
with (multiple) groups, inheriting and merging the settings.

These are — I think — my base requirements. What would you add?

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[asterisk-users] Thankfully no longer with Puppet (was: Managing complex setups with Asterisk)

2012-11-15 Thread martin f krafft

also sprach Shaun Ruffell sruff...@digium.com [2012.11.08.1615 +0100]:
  My systems are already managed automatically, thankfully no longer
  with Puppet. ;)
 
 Just out of curiosity why do you say this?

Sorry for the late reply, I don't want to go into this on the list,
but if you are curious:
http://madduck.net/blog/2012.10.19:configuration-management/

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Re: [asterisk-users] Managing complex setups with Asterisk

2012-11-15 Thread martin f krafft

also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.08.2304
+0100]:
Either way, it sounds like you need to store your data some place and
start building it out.

To recap: given that Asterisk RealTime doesn't really provide
anything more than real-time access to data (i.e. the data in the
database are not any more structured that they are in
/etc/asterisk), any more logical and/or abstract approach to
Asterisk configuration means that the data have to come from
elsewhere and be brought into shape.

Either the abstraction happens in a relational database and Asterisk
accesses stored procedures or views (I would not use LDAP due to
childhood traumata), or the relational database is used to generate
Asterisk's configuration files, or some other data source is used to
generate these configuration files.

It's a shame that noone has done anything into this direction yet.
On the other hand, it means that there aren't already a dozen
PHP+MySQL hacks out there, and that's a good thing.

So if I design the database (PostgreSQL), anyone interested in
providing a frontend, e.g. using Django?

Are people interested in discussing the design here and making it
widely usable? I only have my own three use-cases to refer to, and
I would probably impose my own paradigms…

Does anyone already have something done into that domain?

--
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she is absolutely inadmissible into society. many a woman has a past,
but I am told that she has at least a dozen, and that they all fit.
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Re: [asterisk-users] Impromptu conferencing

2012-11-08 Thread martin f krafft

also sprach Administrator TOOTAI ad...@tootai.net [2012.11.08.0954 +0100]:
 Does anyone have a working example they would be willing to
 share?
 
 As said by James, you just have to transfer all parties in
 a conference room and then you call this conference.

The scenario is usually that we are in a discussion and need a third
party. I suppose I can tell the initial correspondent I will now
transfer you to a conference room, enter this PIN when asked, then
hang up, dial the next, and do the same.

What I would like to do is to convert the current channel into
a conference room, go on hold and dial a third party, and when
I come back to the conference room, I bring along the third party.

Put differently: I don't really want my correspondents to have to do
anything, just wait and listen to MOH.

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Re: [asterisk-users] Managing complex setups with Asterisk

2012-11-08 Thread martin f krafft

also sprach Jeff LaCoursiere j...@sunfone.com [2012.11.07.2049 +0100]:
 Just to chime in, if you REALLY want multi-tenant, it is super
 easy and surprisingly efficient to use kernel level virtualization
 to run multiple instances of asterisk (and even FreePBX).  We use
 LXC to do this.  The host runs an instance that has the dahdi
 hardware, drivers, and upstream connections.  The clients have
 SIP connections to the host for all inbound/outbound

Yes, separation into logical units is one way forward, but then you
will necessarily have redundant configuration between the instances.
It's nice to have clear separations (unless you cannot clearly
separate), but I am not convinced that this decreases complexity.

-- 
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toleranz heißt, die fehler der anderen entschuldigen.
 takt heißt, sie nicht bemerken.
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Re: [asterisk-users] Impromptu conferencing

2012-11-08 Thread martin f krafft

also sprach Administrator TOOTAI ad...@tootai.net [2012.11.08.1018 +0100]:
 For a 3 way conference, all those days phones are able to do this.

Yeah, except I want Asterisk to handle that, not my phone (which
might lose reception or run out of battery etc.).

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[asterisk-users] Managing complex setups with Asterisk

Hello,

we are finally going to redesign our Asterisk-Setup, which has grown
quite complex. We have five sites with a total of 400 users, 15 SIP
registrations and 3 IAX registrations. We do not use any
VoIP-hardware, so it's all software-based. But we make heavy use of
features, including voicemail, followme, conferencing,
call-recording, and queuing.

As I said, the configuration has grown quite complex — so complex
that we are all a bit scared to touch it. It works, but as we are
now adding a sixth site and upgrading the hardware, we thought it
would be a good opportunity to get the sixth site up and running on
a new box, then migrate the other sites.

Now we are trying to figure out how to organise sip.conf, iax.conf
and extensions.conf. I read about Realtime configuration, but I was
a bit disappoointed because it's really just moving the
section-key-value store from the flat files to a relational
database without really making use of any relational features. Sure,
it's realtime thereafter, but not any less complex.

So what to do? Does anyone have a similar setup and would like to
offer a glance into their configs? Are there best practices? Or is
there maybe even software (Linux) to manage setups?

Cheers,

--
martin | http://madduck.net/ | http://two.sentenc.es/

not the truth in whose possession any man is, or thinks he is, but
the honest effort he has made to find out the truth, is what
constitutes the worth of man.
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Re: [asterisk-users] Managing complex setups with Asterisk

Can Asterisk do virtual hosting? While I want/need the sites to be
hosted by the same instance (so that e.g. calls can be transferred
easily), I don't want to have to name my peers [site1-john], and
I want people to be able to SIP-dial j...@site1.example.org and
j...@site2.example.org and trust that Asterisk knows what to do.

Thanks,

-- 
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the human brain is like an enormous fish --
 it is flat and slimy
 and has gills through which it can see.
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Re: [asterisk-users] Managing complex setups with Asterisk

also sprach Joshua Colp jc...@digium.com [2012.11.07.1831 +0100]:
Peer names have to be distinct, this is just a fundamental design
element of chan_sip. What a lot of people end up doing is instead of
treating peers as people they treat them as devices. The peer name
becomes the MAC address of the device they have been assigned.

Especially in combination with users.conf, this can become quite
cumbersome.

Also, it solves the sip.conf problem, but in extensions.conf, your
contexts still need to encode the locality/domain, e.g.
[site1-phones], [site2-outgoing] and [incoming-to-site3]. This is
all doable, with prefixes and #includes, but it requires more
discipline than if Asterisk would simply learn to virtually host. ;)

Also, when users have multiple devices, then handing out two sets of
credentials is a bit of a pain. I realise that this is not specific
to your suggestion, but I do recall a university using Asterisk that
provided 10 logins for everyone, i.e. if my username was 12345, then
12345[0-9] would all be valid SIP login names using the same
password. Any idea how this was done? 10 stanzas? ;)

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brevity is the soul of wit.
-- polonius (hamlet)
brevity is ... wit.
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[asterisk-users] Impromptu conferencing

Dear list,

we would really like to be able to invite a third and fourth party
to our current one-on-one call. At the moment, we have to agree to
dial into MeetMe 10 minutes later, then make calls to the third
parties, and hope it all works out.

I have found a couple of examples on the Internet for converting
channels into conferences, but I could not get any of them working.

Does anyone have a working example they would be willing to share?

Thanks,

-- 
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picture yourself in a boat on a river
 with tangerine trees and marmelade skies...
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Re: [asterisk-users] Managing complex setups with Asterisk

also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.07.2340
+0100]:
What is your point of pain? Right now we do most of the
configuration, provisioning, and system management outside of
asterisk.

My systems are already managed automatically, thankfully no longer
with Puppet. ;)

I am only talking about configuration of Asterisk, whether in
/etc/asterisk or some sensible external data source. My point of
pain is the complexity due to a couple of special cases, e.g.

- Roaming users, i.e. no 1:n relation between sites and users;
- Multiple devices per user (some want them all to ring, some want
individual extensions but shared voicemail, …)
- Keeping track of the mappings between incoming calls (from SIP
providers) and extensions to ring (using incoming contexts and
extension groups for that)
- Keeping track of which extension uses which outgoing trunk
- …

With a logical naming scheme, a policy and include files, this is
all working. But it's very error-prone and there is a bit of
redundancy in the information, so I was wondering if there wasn't
a better way.

Either way, don't manually build your 6th machine. Start from
fresh using some sort of automated tool (chef / puppet). This
will help you get on the right path.

The new machine for the 6th site is up and running (provisioning
(not image-based) took less than half an hour). What now? ;)

--
martin | http://madduck.net/ | http://two.sentenc.es/

science without religion is lame,
religion without science is blind.
-- albert einstein

spamtraps: madduck.bo...@madduck.net

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Re: [asterisk-users] Managing complex setups with Asterisk