Re: [asterisk-users] seeding an originated number in a SIP phone [was: Re: Thunderbird extension using AMI to dial]

[Olle E. Johansson]

29 aug 2011 kl. 15:05 skrev Kevin P. Fleming:

 On 08/28/2011 01:56 AM, Tzafrir Cohen wrote:
 On Thu, Aug 25, 2011 at 07:36:53PM +0100, Chris Hastie wrote:
 Hi
 
 I've just added direct support for AMI to a forthcoming version of
 TBDialOut, a Thunderbird extension for dialling direct from
 Thunderbird's address book. If anyone fancies testing it I'd be grateful
 for any feedback. If you feel like casting a critical eye over the code,
 or doing some translating, even better.
 
 AMI support is available in TBDialOut 1.7.0pre1, which can be found
 either at http://www.oak-wood.co.uk/tbdialout/ or from the 'Development
 channel' at the bottom of the page at
 https://addons.mozilla.org/en-US/thunderbird/addon/tbdialout/
 
 We already have a dialer script (sent to this list a while ago) so it's
 good to see that this extension support that simpler option as well (I
 don't use ThunderBird, as you can see. Some others in the office do use
 it).
 
 One followup question: I originate a call from a SIP phone to some
 remote number. The problem is that the number will not show up properly
 in the list of outgoing calls for the phone. Any idea how to fix this
 (for whatever SIP phone)?
 
 You aren't originating a call *from* the phone (that would require some sort 
 of API into the phone itself to make it place a call). You are originating a 
 call *to* the phone and also to another endpoint; as far as the SIP phone is 
 concerned, this is an incoming call.
 
 I've never seen discussion of a desire to provide a method for an incoming 
 call to be treated as if the endpoint had placed the call itself in any of 
 the SIP discussion lists I frequent... so I'm pretty sure there's no standard 
 way to do this.
Oh, there is - REFER.

We could possibly implement sending a REFER request to the phone, something 
that is frequently used to do call setups from click-to-call apps. This is not 
something we do support in Asterisk today. I've implemented it using SIP 
libraries since Asterisk doesn't have to be involved in the REFER.

If you do ORIGINATE from the phone you have to be aware that Asterisk lacks 
some security framework here. An application that has ORIGINATE access can 
reach the whole dialplan. I have patches for that which needs to be moved 
forward. My proposal is to add a default context to manager accounts to put a 
limitation of destinations they can reach with originate and redirect AMI 
commands (which where the only ones I could come up with as dangerous in this 
regard).

/O
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[asterisk-users] FREE webinar video about Auto-Dialer Business Model (Telemarketing)

[Kolmisoft Marketing]

Hello,

We would like to share our webinar about Auto-Dialer Business Model
(Telemarketing).

It is educational video which we made for our clients and now we are sharing
it with you.

http://www.kolmisoft.com/how-to-start-a-VoIP-business/webinars/

Enjoy.

NOTE: This is not attempt to sell you anything. No product or service is
sold/marketed in the video.

Regards,
Kolmisoft Team
www.kolmisoft.com
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Re: [asterisk-users] Possible Bug? .call files executing multiple times

[Brandon Phelps]

Thanks Danny.  Changing the ownership of the .call files seems to have 
fixed the problem and I can now see that asterisk is adding a 
StartRetry line to the end of the file after it makes the first call, 
which it was unable to do before since the file was owned by root:


cp test5703.call /tmp/test.call  chown asterisk:asterisk 
/tmp/test.call  mv /tmp/test.call /var/spool/asterisk/outgoing/


Thanks,
Brandon

On 08/29/2011 05:41 PM, Danny Nicholas wrote:

Asterisk has to be able to execute and rewrite the file - the call file is
updated in place and when the call is considered successful, removed.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps
Sent: Monday, August 29, 2011 4:28 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple
times

Also I should note that we use the 'noatime' attribute on the /var
filesystem, would this cause the problem below?


On 08/29/2011 05:22 PM, Brandon Phelps wrote:

Here is the contents of the .call file. The file is the same before the
move as after (I did a cat on the file after the move, while the phone
was ringing a second time):

Channel: Local/5703@ext-main
Callerid: MyCompany8005551234
Set: TicketNumber=100
Set: CallerID_Num=8005551234
Set: CALLSTATUS=0
Context: ext-autodialer
MaxRetries: 0
WaitTime: 45
Extension: s
Priority: 1

We have tried using a SIP channel as well (as opposed to Local) with the
same results. The s extension of ext-autodialer runs an AGI script which
makes use of those Set: variables.

I can most easily reproduce the problem by simply not answering the
call. After 2 or 3 rings line 2 on the phone lights up indicating
another call. If I reject the first call and answer the second call,
it's the same script.

Also during my most recent test the following happened:

1. I moved file to /var/spool/asterisk/outgoing
2. Phone rang on line 1
3. I let phone continue to ring
4. After 3 rings, line 2 started ringing (another call from the same
.call file)
5. I rejected both calls, sending both to voicemail.
6. 6 or 7 seconds after rejecting both calls, the phone rang a 3rd time.
7. I let the phone ring until it was automatically moved to voicemail
and finally the .call file was removed.


On 08/29/2011 11:00 AM, Danny Nicholas wrote:

Can you post the .call file (with called number blacked out) before
call and
after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2
should
be from /v/s/a/o).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon
Phelps
Sent: Monday, August 29, 2011 8:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug? .call files executing
multiple
times

On 08/19/2011 09:14 AM, Brandon Phelps wrote:

Hello all,

We are setting up an auto-dialer to call customers based on the
opening of tickets in our internal ticketing system. Everything is
going fine so far except for one snag:

To test the system we are implementing I am manually moving .call
files into the /var/spool/asterisk/outgoing directory like this:

asterisk@dialerdev:~# cp test5703.call /tmp/test.call  mv
/tmp/test.call /var/spool/asterisk/outgoing/

This works great and the call is immediately started, however more
often than not (ie. not all the time, but most of the time) after
answering the call or rejecting it (sending it to voicemail), another
call is performed using the same file.

I notice that when a call is initiated the .call file is not removed
immediately. Instead, asterisk waits until the call is completed
before removing the call file, so it seems like 5-10 seconds into the
call since the .call file still exists another call is placed.

Any advice on how we can avoid this situation and ensure that only one
call is made per .call file?

The OS is Ubuntu 11.04 server and we're running Asterisk 1.8.

Thanks,



Sorry to bring this back up but I am still having this issue and
haven't had
any luck resolving it. It should be noted that the .call files in
question
are set to MaxRetries: 0, and simply connect the call to the 's'
extension
in a custom context. From there the context is pretty complicated,
running
some AGI scripts along with some dealing with user input, basically a
simple
IVR.

Any help would be appreciated.

Thanks,
Brandon

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Re: [asterisk-users] Wanted a modified SIP message body

[Jaime Lozano]

Hello,
I have been using wireshark to capture some traffic. I'm talking when the
PBX sends OK (200) connection accepted. 3Com PBX sends TZ=7200\n (an much
more things) in a SIP packet message body but Asterisk PBX sends packets
without message body, it only sends variables in the message header. So I
want Asterisk to send packets with a message body and its proper content.

I've been looking for, sure spending my time. Modifying the source code is
not very realistic, but i have to try. The file channels/chan_sip.c seems to
have interesting functions like add_header(), add_tcodec_to_sdp(),
handle_response_invite(), handle_response(), also struct cfalias. I know C
programming but it's really hard to understand the code.

Should I ask in the developers list?

have a nice day



2011/8/27 Jaime Lozano jaimelozan...@gmail.com



 -- Forwarded message --
 From: Olle E. Johansson o...@edvina.net
 Date: 2011/8/26
 Subject: Re: [asterisk-users] Wanted a modified SIP message body
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com



 26 aug 2011 kl. 14:06 skrev Jaime Lozano:

  Hello,
  In which file do I use SIPAddHeader()?
  Please consider that the packet goes from the PBX to the telephone, and
 what I want is not a header because the TZ: 7200\n is in the *message
 body* not in the *message header*.

 That's no longer a SIP header, it's part of the SDP you want to change. You
 can't do that without changing the source code.

 /O
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Re: [asterisk-users] Wanted a modified SIP message body

[Kevin P. Fleming]

On 08/30/2011 07:36 AM, Jaime Lozano wrote:


I have been using wireshark to capture some traffic. I'm talking when
the PBX sends OK (200) connection accepted. 3Com PBX sends TZ=7200\n
(an much more things) in a SIP packet message body but Asterisk PBX
sends packets without message body, it only sends variables in the
message header. So I want Asterisk to send packets with a message body
and its proper content.


This is extremely confusing, to say the least. 'a message body' and 'its 
proper content' are ambiguous, especially since Asterisk already works 
with pretty much every SIP UA on the planet and none of them require the 
things you are asking for.


Why don't you post an actual example of what you think Asterisk should 
be sending, and what it is actually sending, rather than trying to 
describe the differences (which is clearly not going well)?


In general, though, you can't just put random content in a SIP request 
or response message body; the message body is usually of a defined type 
(application/sdp, for example), and has rules about what it can and 
cannot contain.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] FREE webinar video about Auto-Dialer Business Model (Telemarketing)

[Paul Belanger]

On 11-08-30 05:16 AM, Kolmisoft Marketing wrote:

NOTE: This is not attempt to sell you anything. No product or service is
sold/marketed in the video.

That maybe the case, but this is still a non-commercial mailing list. 
Please use asterisk-biz in the future.


--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?

[Paul Hayes]

On 27/08/11 10:14, Gordon Henderson wrote:

On Sat, 27 Aug 2011, Alan Lord (News) wrote:


On 26/08/11 19:02, linux guy wrote:

I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home
asterisk system.


We've been using the Siemens Gigaset 685IP range for over three years
and I'm (still) very pleased with them:


+1



The current generation is the N300 or N300A (A = with answering 
machine).  These have the advantage of being able to do 3 SIP calls at 
once.  You also don't get a handset with it so you can choose whatever 
handset you want (in theory doesn't even have to be a Gigaset handset as 
long as it is GAP compatible but you'll have a better time if you do use 
Gigaset ones, or at least one).


These definitely work well with Asterisk for me.

So if you really need 6 calls at once then you can get 2 base stations 
and 6 handsets.  If you can live with 6 handsets but only 3 of them in 
use at once, then a single base will do the job.


cheers,
Paul.

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[asterisk-users] T.38 passthru on 1.8.5

[Fabian Borot]

 Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built 
by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 
21:31:22 UTC]

The call flow is:
quintum gateway -- asterisk -- Dialogic IMG 1010

the call starts as a voice call, the remote fax picks up and we hear the fax 
tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is 
passed back to the user side [quintum gateway] whcih reply with 200 OK with 
t.38 and the nothing else happens. After 20 secs of inactivity the quintum 
sends another Invite with voice only and then a BYE.

We do see that the quintum sends a lot of messages like this from the quintum's 
IP [192.168.1.18] but we do not see that asterisk sends the packages to the 
destination

UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, 
len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)

we have this settings on sip.conf
faxdetect = yes
t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations 
t38pt_udptl = yes;t38pt_udptl = yes,fec etc]

When we send the fax from the quintum to the Dialogic IMG the fax works 100% of 
the times.
I enabled fax set debug on and udptl set debug on but the console does not show 
almost anything but the udptl packets shown above.
What else should I do?Any ideas/help is greatly appreciated

txs a lot
fborot

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[asterisk-users] same sip peer as user and provider

[Fabian Borot]

Hello

Up to version 1.6.0 we have been able  to configure the same SIP device as a 
user [inbound trunk] and as a peer [outbound trunk] w/o issues.
After we switched to version 1.8 this setup wont work, apparently one can not 
have the same IP on 2 different trunks anymore. The trunk that is configured as 
user or friend is not choosen when the inbound call hits asterisk, instead the 
outbound trunk is and that trunk is usually w/o context and then asterisk can 
not find any call logic in the dialplan in the default extension, hence the 
call fails.

as a workaround I have been trying the SIP_CODEC variables [inbound and 
outbound] but it wont help me in all cases.

Also I can not set the ptime on the fly using those variables in the dialplan.

After reading in the forums and the books/guides apparently the users are 
matched by the From header and peers are matched by IP. 

Is this is the intended behavior now?

any help is greatly appreciated, txs a lot in advance
fborot 
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Re: [asterisk-users] same sip peer as user and provider

[A J Stiles]

On Tuesday 30 August 2011, Fabian Borot wrote:
 Up to version 1.6.0 we have been able  to configure the same SIP device as
 a user [inbound trunk] and as a peer [outbound trunk] w/o issues. After we
 switched to version 1.8 this setup wont work, apparently one can not have
 the same IP on 2 different trunks anymore.

Have you tried defining the device as type=friend ?

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] same sip peer as user and provider

[Fabian Borot]

yes, same thing

 

From: fbo...@hotmail.com
To: fbo...@hotmail.com
Subject: RE: same sip peer as user and provider
Date: Tue, 30 Aug 2011 10:35:01 -0400








yes my friend. same thing


 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: same sip peer as user and provider
Date: Tue, 30 Aug 2011 10:16:11 -0400








Hello

Up to version 1.6.0 we have been able  to configure the same SIP device as a 
user [inbound trunk] and as a peer [outbound trunk] w/o issues.
After we switched to version 1.8 this setup wont work, apparently one can not 
have the same IP on 2 different trunks anymore. The trunk that is configured as 
user or friend is not choosen when the inbound call hits asterisk, instead the 
outbound trunk is and that trunk is usually w/o context and then asterisk can 
not find any call logic in the dialplan in the default extension, hence the 
call fails.

as a workaround I have been trying the SIP_CODEC variables [inbound and 
outbound] but it wont help me in all cases.

Also I can not set the ptime on the fly using those variables in the dialplan.

After reading in the forums and the books/guides apparently the users are 
matched by the From header and peers are matched by IP. 

Is this is the intended behavior now?

any help is greatly appreciated, txs a lot in advance
fborot 

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[asterisk-users] Avaya to Asterisk Voice mail

[Dustin fails]

Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue
line. The issue I am having is Avaya is sending the originating caller id
not the station id so Asterisk see that originating id so I can't route the
call correctly in Asterisk.

Thanks!

Dustin
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Re: [asterisk-users] Avaya to Asterisk Voice mail

[Robert Huddleston]

Search the forum - I believe I remember a recent exchange on this subject

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dustin fails
Sent: Tuesday, August 30, 2011 10:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Avaya to Asterisk Voice mail

 

Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue
line. The issue I am having is Avaya is sending the originating caller id
not the station id so Asterisk see that originating id so I can't route the
call correctly in Asterisk. 

Thanks!

Dustin

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Re: [asterisk-users] T.38 passthru on 1.8.5

[Karsten Wemheuer]

Hi,

Am Dienstag, den 30.08.2011, 09:44 -0400 schrieb Fabian Borot:
 
  Hello
 We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk
 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running
 Linux on 2011-08-26 21:31:22 UTC]
 
 The call flow is:
 quintum gateway -- asterisk -- Dialogic IMG 1010
 
 the call starts as a voice call, the remote fax picks up and we hear
 the fax tone, the we see the re-invite from the IMG asking for t.38,
 the RE-Invite is passed back to the user side [quintum gateway] whcih
 reply with 200 OK with t.38 and the nothing else happens. After 20
 secs of inactivity the quintum sends another Invite with voice only
 and then a BYE.
 
 We do see that the quintum sends a lot of messages like this from the
 quintum's IP [192.168.1.18] but we do not see that asterisk sends the
 packages to the destination
 
 UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0,
 seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
  UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 
 we have this settings on sip.conf
 faxdetect = yes
 t38pt_udptl = yes,maxdatagram=400 [I have tested with several
 combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc]
 
 When we send the fax from the quintum to the Dialogic IMG the fax
 works 100% of the times.
 I enabled fax set debug on and udptl set debug on but the console does
 not show almost anything but the udptl packets shown above.
 What else should I do?Any ideas/help is greatly appreciated

I assume there is some NAT/firewall on the way? (The 192.30.189.146 is
public internet and 192.168.1.18 is RFC1918). Be sure to let udptl pass
through your firewall/NAT. Udptl is using a port range of its own
(different from the range used for audio). 

Audio is always symmetric. Audio in one directions implies audio in the
opposite direction. A NAT Gateway open for one direction is normally no
problem for audio. 
T.38 is not symmetric. One side sends and waits for some kind of ACK.
The other side waits for data. If the sending side gets through the NAT
Gateway all is fine. If the sending side is not getting through, the
transmission is aborted (after timeout).

HTH,

Karsten




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Re: [asterisk-users] USB or Ethernet based FXO device ?

[Carlos Chavez]

On Tue, 2011-08-30 at 01:31 +0200, Gilles wrote:
 On Sat, 27 Aug 2011 09:31:12 -0600, linux guy linuxguy...@gmail.com
 wrote:
 I'm looking for an FXO device to connect to a POTS line that communicates
 via USB or Ethernet.
 
 For USB, AFAIK, there's only the one from Sangoma. All others are
 Ethernet-based.
 
 www.voip-info.org/wiki/view/VoIP+Gateways
 
Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] T.38 passthru on 1.8.5

[Fabian Borot]

both endpoints use public Ips, I just changed the real ones for the privates 
ones to protect our ips but made a mistake and left the dest as a pub and the 
orig as private, my bad.
but for the record, both are public IPs, there is no nat and iptables is off

 also, I see that the quintum sends a lot of these packages but asterisk sends 
only 1 or 2 to the other side.




 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 09:44:15 -0400










 Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built 
by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 
21:31:22 UTC]

The call flow is:
quintum gateway -- asterisk -- Dialogic IMG 1010

the call starts as a voice call, the remote fax picks up and we hear the fax 
tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is 
passed back to the user side [quintum gateway] whcih reply with 200 OK with 
t.38 and the nothing else happens. After 20 secs of inactivity the quintum 
sends another Invite with voice only and then a BYE.

We do see that the quintum sends a lot of messages like this from the quintum's 
IP [192.168.1.18] but we do not see that asterisk sends the packages to the 
destination

UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, 
len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)

we have this settings on sip.conf
faxdetect = yes
t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations 
t38pt_udptl = yes;t38pt_udptl = yes,fec etc]

When we send the fax from the quintum to the Dialogic IMG the fax works 100% of 
the times.
I enabled fax set debug on and udptl set debug on but the console does not show 
almost anything but the udptl packets shown above.
What else should I do?Any ideas/help is greatly appreciated

txs a lot
fborot


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[asterisk-users] dahdi with isdn nt_mode, phone no signal still.

[Tamer Higazi]

Hi people!
I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on
asterisk. On which DAHDI tells me also properly that both of my boards
are registered, one in TE and the other on in NT mode.

Calls do successfully come inside, but I want to connect my ISDN phone
at the board, but the phone is death. What did I make wrong?!


dahdi_scan:
office tamer # dahdi_scan
[1]
active=yes
alarms=OK
description=HFC-S PCI A ISDN card 0 [TE]
name=ZTHFC1
manufacturer=Cologne Chips
devicetype=HFC-S PCI-A ISDN
location=PCI Bus 01 Slot 01
basechan=1
totchans=3
irq=17
type=digital-TE
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI
framing_opts=CCS
coding=AMI
framing=CCS
[2]
active=yes
alarms=OK
description=HFC-S PCI A ISDN card 1 [NT]
name=ZTHFC2
manufacturer=Cologne Chips
devicetype=HFC-S PCI-A ISDN
location=PCI Bus 01 Slot 02
basechan=4
totchans=3
irq=21
type=digital-NT
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI
framing_opts=CCS
coding=
framing=CAS
office tamer #


and dahd status in asterisk:

office*CLI dahdi show status
Description  Alarms  IRQbpviol CRC4  
Fra Codi Options  LBO
HFC-S PCI A ISDN card 0 [TE] OK  0  0  0 
CCS AMI   0 db (CSU)/0-133 feet (DSX-1)
HFC-S PCI A ISDN card 1 [NT] OK  0  0  0 
CAS Unk   0 db (CSU)/0-133 feet (DSX-1)

TE-mode works fine so far.

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Re: [asterisk-users] T.38 passthru on 1.8.5

[Fabian Borot]

will installing spandsp help with t.38 pass-through?


 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 11:42:41 -0400








both endpoints use public Ips, I just changed the real ones for the privates 
ones to protect our ips but made a mistake and left the dest as a pub and the 
orig as private, my bad.
but for the record, both are public IPs, there is no nat and iptables is off

 also, I see that the quintum sends a lot of these packages but asterisk sends 
only 1 or 2 to the other side.




 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 09:44:15 -0400










 Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built 
by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 
21:31:22 UTC]

The call flow is:
quintum gateway -- asterisk -- Dialogic IMG 1010

the call starts as a voice call, the remote fax picks up and we hear the fax 
tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is 
passed back to the user side [quintum gateway] whcih reply with 200 OK with 
t.38 and the nothing else happens. After 20 secs of inactivity the quintum 
sends another Invite with voice only and then a BYE.

We do see that the quintum sends a lot of messages like this from the quintum's 
IP [192.168.1.18] but we do not see that asterisk sends the packages to the 
destination

UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, 
len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)

we have this settings on sip.conf
faxdetect = yes
t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations 
t38pt_udptl = yes;t38pt_udptl = yes,fec etc]

When we send the fax from the quintum to the Dialogic IMG the fax works 100% of 
the times.
I enabled fax set debug on and udptl set debug on but the console does not show 
almost anything but the udptl packets shown above.
What else should I do?Any ideas/help is greatly appreciated

txs a lot
fborot


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Re: [asterisk-users] T.38 passthru on 1.8.5

[Steve Underwood]

On 08/31/2011 01:15 AM, Fabian Borot wrote:

will installing spandsp help with t.38 pass-through?
The only part of spandsp which is relevant to T.38 passthrough is its 
modem tone detection module, and I don't think the standard Asterisk 
distribution can make use of that. Some people do use it, to overcome 
the limitations in Asterisk's own tone detection, but I don't think they 
make their patches available.


Steve



From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 11:42:41 -0400

both endpoints use public Ips, I just changed the real ones for the 
privates ones to protect our ips but made a mistake and left the dest 
as a pub and the orig as private, my bad.
but for the record, both are public IPs, there is no nat and iptables 
is off


 also, I see that the quintum sends a lot of these packages but 
asterisk sends only 1 or 2 to the other side.







From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 09:44:15 -0400


 Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 
1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running 
Linux on 2011-08-26 21:31:22 UTC]


The call flow is:
quintum gateway -- asterisk -- Dialogic IMG 1010

the call starts as a voice call, the remote fax picks up and we hear 
the fax tone, the we see the re-invite from the IMG asking for t.38, 
the RE-Invite is passed back to the user side [quintum gateway] whcih 
reply with 200 OK with t.38 and the nothing else happens. After 20 
secs of inactivity the quintum sends another Invite with voice only 
and then a BYE.


We do see that the quintum sends a lot of messages like this from the 
quintum's IP [192.168.1.18] but we do not see that asterisk sends the 
packages to the destination


UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, 
seq 0, len 6)

 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)

we have this settings on sip.conf
faxdetect = yes
t38pt_udptl = yes,maxdatagram=400 [I have tested with several 
combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc]


When we send the fax from the quintum to the Dialogic IMG the fax 
works 100% of the times.
I enabled fax set debug on and udptl set debug on but the console does 
not show almost anything but the udptl packets shown above.

What else should I do?Any ideas/help is greatly appreciated

txs a lot
fborot


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Re: [asterisk-users] T.38 passthru on 1.8.5

[Fabian Borot]

txs a lot for your explanation steve
so, it should work w/o spandsp fairly fine if we do not have a bad connection. 
I see that this version has a lot of fixes related to t.38
 but is the implementation already mature enough to guarantee a decent success 
rate with fax calls?


 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 13:15:19 -0400








will installing spandsp help with t.38 pass-through?


 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 11:42:41 -0400








both endpoints use public Ips, I just changed the real ones for the privates 
ones to protect our ips but made a mistake and left the dest as a pub and the 
orig as private, my bad.
but for the record, both are public IPs, there is no nat and iptables is off

 also, I see that the quintum sends a lot of these packages but asterisk sends 
only 1 or 2 to the other side.




 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 09:44:15 -0400










 Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built 
by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 
21:31:22 UTC]

The call flow is:
quintum gateway -- asterisk -- Dialogic IMG 1010

the call starts as a voice call, the remote fax picks up and we hear the fax 
tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is 
passed back to the user side [quintum gateway] whcih reply with 200 OK with 
t.38 and the nothing else happens. After 20 secs of inactivity the quintum 
sends another Invite with voice only and then a BYE.

We do see that the quintum sends a lot of messages like this from the quintum's 
IP [192.168.1.18] but we do not see that asterisk sends the packages to the 
destination

UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, 
len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)

we have this settings on sip.conf
faxdetect = yes
t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations 
t38pt_udptl = yes;t38pt_udptl = yes,fec etc]

When we send the fax from the quintum to the Dialogic IMG the fax works 100% of 
the times.
I enabled fax set debug on and udptl set debug on but the console does not show 
almost anything but the udptl packets shown above.
What else should I do?Any ideas/help is greatly appreciated

txs a lot
fborot



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Re: [asterisk-users] dahdi with isdn nt_mode, phone no signal still.

[Patrick Lists]

On 08/30/2011 06:32 PM, Tamer Higazi wrote:

Hi people!
I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on
asterisk. On which DAHDI tells me also properly that both of my boards
are registered, one in TE and the other on in NT mode.

Calls do successfully come inside, but I want to connect my ISDN phone
at the board, but the phone is death. What did I make wrong?!


Afaik ISDN phones need power to work. You would need a HFC-S card with a 
power plug where you can hook up power from the computer's power supply 
so the card can power the phone. Like Sangoma's B700. Alternatively 
perhaps you could try to power the phone through an NT1?


Would you mind sharing which version of DAHDI you used and where you got 
the HFC-S patch?


Regards,
Patrick

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Re: [asterisk-users] dahdi with isdn nt_mode, phone no signal still.

[Tamer Higazi]

Hi Patrick!
Now i got it.

I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1.

The patches are automatically integrated at Gentoo. I didn't have to
patch anything. That did the community.

Another question, I really don't like to buy a new ISDN phone with
external power connector, can I make the power supply for the phone
somehow?!

Tamer

Am 30.08.2011 20:05, schrieb Patrick Lists:
 On 08/30/2011 06:32 PM, Tamer Higazi wrote:
 Hi people!
 I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on
 asterisk. On which DAHDI tells me also properly that both of my boards
 are registered, one in TE and the other on in NT mode.

 Calls do successfully come inside, but I want to connect my ISDN phone
 at the board, but the phone is death. What did I make wrong?!
 
 Afaik ISDN phones need power to work. You would need a HFC-S card with a
 power plug where you can hook up power from the computer's power supply
 so the card can power the phone. Like Sangoma's B700. Alternatively
 perhaps you could try to power the phone through an NT1?
 
 Would you mind sharing which version of DAHDI you used and where you got
 the HFC-S patch?
 
 Regards,
 Patrick
 
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Re: [asterisk-users] T.38 passthru on 1.8.5

[Kevin P. Fleming]

On 08/30/2011 12:53 PM, Fabian Borot wrote:

txs a lot for your explanation steve
so, it should work w/o spandsp fairly fine if we do not have a bad
connection. I see that this version has a lot of fixes related to t.38
but is the implementation already mature enough to guarantee a decent
success rate with fax calls?


Since you are using T.38, nothing is going to guarantee that... there 
are many obstacles to achieving it. We can try, though :-)


There's really no way to figure out what is happening here without a 
packet capture of all the traffic and an Asterisk console log at high 
verbose/debug levels. Is there a way you can produce that and provide it 
to us without having to reveal confidential information? If not, we can 
create a private issue on the issue tracker for you to have a place to 
upload your files without them being visible to the public.


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Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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[asterisk-users] MOH making calls appear hung up

[Kevin Oravits]

Greetings,

I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones. 
With one of my sites, we're having an issue where when a call is transferred, 
the MOH is not playing and all the caller is hearing is silence. The caller of 
course thinks they have been hung up on, but the call is actually still in 
progress and gets successfully transferred if they wait until the person 
answers.

I have researched online and even consulted our 3rd party vendor but no one 
seems to know how to fix it.

Anyone have any advice? Any help would be appreciated.

Thanks,

Stivaro
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Re: [asterisk-users] MOH making calls appear hung up

[Danny Nicholas]

It seems a reasonable likelihood that your moh at the offending site does
not match the codec of the call (IE your MOH is wav and your call codec is
SLIN).  Set your verbosity and debug up to 15 and try a call to verify this.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 1:53 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] MOH making calls appear hung up

 

Greetings,

 

I'm have asterisk servers at about 10 sites, all using Polycom IP 450
phones. With one of my sites, we're having an issue where when a call is
transferred, the MOH is not playing and all the caller is hearing is
silence. The caller of course thinks they have been hung up on, but the call
is actually still in progress and gets successfully transferred if they wait
until the person answers.

 

I have researched online and even consulted our 3rd party vendor but no one
seems to know how to fix it.

 

Anyone have any advice? Any help would be appreciated.

 

Thanks,

 

Stivaro

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Re: [asterisk-users] dahdi with isdn nt_mode, phone no signal still.

[Patrick Lists]

On 08/30/2011 08:37 PM, Tamer Higazi wrote:

Hi Patrick!
Now i got it.

I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1.

The patches are automatically integrated at Gentoo. I didn't have to
patch anything. That did the community.


Thanks for the info.


Another question, I really don't like to buy a new ISDN phone with
external power connector, can I make the power supply for the phone
somehow?!


You can read more about it here:
http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html

Scroll all the way down to 2.2 Connect ISDN telephones to your ISDN 
card to read howto power your ISDN phone with an NT1 (aka NTBA). You 
can find a 2nd hand NT1 on ebay or try your local e-market site. I saw 
several on ebay for $10.


Regards,
Patrick

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Re: [asterisk-users] MOH making calls appear hung up

[Kevin Oravits]

I noticed the CLI shows that the music on hold actually stops for some reason?

Here's the output of my CLI:
Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)
Verbosity is at least 28
-- Executing [s@ivr-boi-ntc-day:3] Answer(SIP/gw1-05d6, ) in new 
stack
-- Executing [s@ivr-boi-ntc-day:4] Wait(SIP/gw1-05d6, 1) in new 
stack
-- Executing [s@ivr-boi-ntc-day:5] Dial(SIP/gw1-05d6, SIP/1021,20) 
in new stack
  == Using SIP RTP CoS mark 5
-- Called 1021
-- SIP/1021-05d7 is ringing
-- SIP/1021-05d7 answered SIP/gw1-05d6
-- Packet2Packet bridging SIP/gw1-05d6 and SIP/1021-05d7
-- Started music on hold, class 'default', on SIP/gw1-05d6
  == Using SIP RTP CoS mark 5
-- Executing [6937@from-sip:1] Macro(SIP/1021-05d8, 
stdexten,6937,sip/6937) in new stack
-- Executing [s@macro-stdexten:1] Wait(SIP/1021-05d8, 1) in new 
stack
-- Executing [s@macro-stdexten:2] Dial(SIP/1021-05d8, sip/6937,20) 
in new stack
  == Using SIP RTP CoS mark 5
-- Called 6937
-- SIP/6937-05d9 is ringing
-- Stopped music on hold on SIP/gw1-05d6
  == Spawn extension (ivr-boi-ntc-day, s, 5) exited non-zero on 
'SIP/1021-05d8ZOMBIE'
-- Nobody picked up in 2 ms
-- Executing [s@macro-stdexten:3] Goto(SIP/gw1-05d6, s-NOANSWER,1) 
in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail(SIP/gw1-05d6, 
6937,u) in new stack
-- SIP/gw1-05d6 Playing 
'/var/spool/asterisk/voicemail/default/6937/unavail.slin' (language 'en')
  == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 
'SIP/gw1-05d6' in macro 'stdexten'
  == Spawn extension (from-sip, 6937, 1) exited non-zero on 'SIP/gw1-05d6'

Thanks!
Kevin Oravits

From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, August 30, 2011 11:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up

It seems a reasonable likelihood that your moh at the offending site does not 
match the codec of the call (IE your MOH is wav and your call codec is SLIN).  
Set your verbosity and debug up to 15 and try a call to verify this.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 1:53 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] MOH making calls appear hung up

Greetings,

I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones. 
With one of my sites, we're having an issue where when a call is transferred, 
the MOH is not playing and all the caller is hearing is silence. The caller of 
course thinks they have been hung up on, but the call is actually still in 
progress and gets successfully transferred if they wait until the person 
answers.

I have researched online and even consulted our 3rd party vendor but no one 
seems to know how to fix it.

Anyone have any advice? Any help would be appreciated.

Thanks,

Stivaro
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Re: [asterisk-users] MOH making calls appear hung up

[Danny Nicholas]

Your Dial command stops the MOH - if the command were Dial(SIP/1021,20,m)
the music would continue until connected or timed-out.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 2:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up

 

I noticed the CLI shows that the music on hold actually stops for some
reason?

 

Here's the output of my CLI:

Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)

Verbosity is at least 28

-- Executing [s@ivr-boi-ntc-day:3] Answer(SIP/gw1-05d6, ) in new
stack

-- Executing [s@ivr-boi-ntc-day:4] Wait(SIP/gw1-05d6, 1) in new
stack

-- Executing [s@ivr-boi-ntc-day:5] Dial(SIP/gw1-05d6,
SIP/1021,20) in new stack

  == Using SIP RTP CoS mark 5

-- Called 1021

-- SIP/1021-05d7 is ringing

-- SIP/1021-05d7 answered SIP/gw1-05d6

-- Packet2Packet bridging SIP/gw1-05d6 and SIP/1021-05d7

-- Started music on hold, class 'default', on SIP/gw1-05d6

  == Using SIP RTP CoS mark 5

-- Executing [6937@from-sip:1] Macro(SIP/1021-05d8,
stdexten,6937,sip/6937) in new stack

-- Executing [s@macro-stdexten:1] Wait(SIP/1021-05d8, 1) in new
stack

-- Executing [s@macro-stdexten:2] Dial(SIP/1021-05d8,
sip/6937,20) in new stack

  == Using SIP RTP CoS mark 5

-- Called 6937

-- SIP/6937-05d9 is ringing

-- Stopped music on hold on SIP/gw1-05d6

  == Spawn extension (ivr-boi-ntc-day, s, 5) exited non-zero on
'SIP/1021-05d8ZOMBIE'

-- Nobody picked up in 2 ms

-- Executing [s@macro-stdexten:3] Goto(SIP/gw1-05d6,
s-NOANSWER,1) in new stack

-- Goto (macro-stdexten,s-NOANSWER,1)

-- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail(SIP/gw1-05d6,
6937,u) in new stack

-- SIP/gw1-05d6 Playing
'/var/spool/asterisk/voicemail/default/6937/unavail.slin' (language 'en')

  == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/gw1-05d6' in macro 'stdexten'

  == Spawn extension (from-sip, 6937, 1) exited non-zero on
'SIP/gw1-05d6'

 

Thanks!

Kevin Oravits  

 

From: Danny Nicholas [mailto:da...@debsinc.com] 
Sent: Tuesday, August 30, 2011 11:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up

 

It seems a reasonable likelihood that your moh at the offending site does
not match the codec of the call (IE your MOH is wav and your call codec is
SLIN).  Set your verbosity and debug up to 15 and try a call to verify this.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 1:53 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] MOH making calls appear hung up

 

Greetings,

 

I'm have asterisk servers at about 10 sites, all using Polycom IP 450
phones. With one of my sites, we're having an issue where when a call is
transferred, the MOH is not playing and all the caller is hearing is
silence. The caller of course thinks they have been hung up on, but the call
is actually still in progress and gets successfully transferred if they wait
until the person answers.

 

I have researched online and even consulted our 3rd party vendor but no one
seems to know how to fix it.

 

Anyone have any advice? Any help would be appreciated.

 

Thanks,

 

Stivaro

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Re: [asterisk-users] USB or Ethernet based FXO device ?

[Gilles]

On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
cur...@telecomabmex.com wrote:
   Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.

Thanks for the tip. It looks like the smallest option is 8 FXO ports:

www.xorcom.com/telephony-interfaces/astribank-models.html


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Re: [asterisk-users] Polycoms rebooting themselves

[Mike Diehl]

Well, we've taken the time to check out the wiring.  It's only 3 years old and 
looks like the people who did it knew what they were doing.  Nice work.

Rebooting the cable modem, router, and switch didn't fix the problem.

Also, we had an instance today where ALL of the phones went down within 
minutes of each other.  The Internet connection was still active.

Looks like more often than not, all of the phones die at the same time.

Any other ideas?

Mike.


On Thursday 18 August 2011 7:58:03 am Justin Sherrill wrote:
 I've had mystery reboots with Polycom IP550s - the culprit in both cases
 was the network connection.  Replacing the cat5 cable to the phone or
 changing the attached port fixed it both times.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
 Sent: Wednesday, August 17, 2011 6:01 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Polycoms rebooting themselves
 
 Hi all,
 
 I've got a customer with 10 Polycom 335's and the latest(ish) firmware. 
 For the most part, things are working well.
 
 However, about once a day, a given phone will just reboot.  They don't do
 it all at once, and they don't do it along any pattern that I can discern.
 
 I've got a tcpdump running against one of the phones on my server, but so
 far, it's not rebooted, so I've got nothing to look at.
 
 Any other ideas?

-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Polycoms rebooting themselves

[Tim Nelson]

- Original Message -
 Well, we've taken the time to check out the wiring. It's only 3 years
 old and
 looks like the people who did it knew what they were doing. Nice work.
 
 Rebooting the cable modem, router, and switch didn't fix the problem.
 
 Also, we had an instance today where ALL of the phones went down
 within
 minutes of each other. The Internet connection was still active.
 
 Looks like more often than not, all of the phones die at the same
 time.
 
 Any other ideas?
 


If they're all powered via PoE on the same switch, look to diagnosing the 
switch itself. Look for issues with heat (not enough cooling or circulation), 
or depending on the switch, you could be pulling too much power from the PoE 
module contained within. Does your switch's PoE module put out enough power for 
'X' number of phones at 'Y' number of watts each?

Either of these problems would lead to the switch shutting down or resetting 
the PoE module which causes your phone reboots.

--Tim

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Re: [asterisk-users] Polycoms rebooting themselves

[Mike Diehl]

On Tuesday 30 August 2011 3:14:50 pm Tim Nelson wrote:
 - Original Message -
 
  Well, we've taken the time to check out the wiring. It's only 3 years
  old and
  looks like the people who did it knew what they were doing. Nice work.
  
  Rebooting the cable modem, router, and switch didn't fix the problem.
  
  Also, we had an instance today where ALL of the phones went down
  within
  minutes of each other. The Internet connection was still active.
  
  Looks like more often than not, all of the phones die at the same
  time.
  
  Any other ideas?
 
 If they're all powered via PoE on the same switch, look to diagnosing the
 switch itself. Look for issues with heat (not enough cooling or
 circulation), or depending on the switch, you could be pulling too much
 power from the PoE module contained within. Does your switch's PoE module
 put out enough power for 'X' number of phones at 'Y' number of watts each?
 
 Either of these problems would lead to the switch shutting down or
 resetting the PoE module which causes your phone reboots.

All of the phones are AC powered.  Either via an injector or wall outlet; I 
don't remember which.  Definitely NOT POE.


-- 

Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] MOH making calls appear hung up

[Kevin Oravits]

Thanks Danny. I tried that but all that did is make it so when I call the site, 
I get hold music instead of ringing. Still has no affect on the call transfer 
MOH.   :/

Interestingly, the music is playing for about 3-5 seconds before stopping 
during the transfer.

I've built all of my phone servers the same at my sites and I'm still sorta 
green on some of this stuff. When doing a call transfer, is it using the 
macro-stdexten or does it go to the IVR dial plan? Because the entry you noted 
is in the IVR dialplan, not the macro-stdexten.

Here's my macro-stdexten:
[macro-stdexten]

exten = s,1,wait(1)
exten = s,2,Dial(${ARG2},20)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s,4,Dial(${ARG2},15)  ; Ring phone for 15 seconds

exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy 
announce
exten = s-BUSY,2,Hangup
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail
exten = s-NOANSWER,2,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

exten = a,1,VoicemailMain(${CALLERIDNUM})
exten = a,2,Hangup

Here's my IVR Dialplan:
exten = s,1,Set(TIMEOUT(digit)=4)
exten = s,n,Wait(1)
exten = s,n,Answer
exten = s,n,Dial(SIP/1021,20)
exten = s,n,Wait(1)
exten = s,n,Dial(SIP/6909,15)
exten = s,n,Dial(SIP/6904,15)
exten = s,n,Voicemail(1021,u)
exten = s,n,Hangup

Note: I removed the ,m because it was only affecting the new incoming calls.

Here's my CLI output:
  == Using SIP RTP CoS mark 5
-- Executing [1021@from-PRI:1] Goto(SIP/gw1-0066, ivr-boi-ntc,s,1) 
in new stack
-- Goto (ivr-boi-ntc,s,1)
-- Executing [s@ivr-boi-ntc:1] GotoIfTime(SIP/gw1-0066, 
07:00-17:00,mon-fri,*,*?ivr-boi-ntc-day,s,1) in new stack
-- Goto (ivr-boi-ntc-day,s,1)
-- Executing [s@ivr-boi-ntc-day:1] Set(SIP/gw1-0066, 
TIMEOUT(digit)=4) in new stack
-- Digit timeout set to 4.000
-- Executing [s@ivr-boi-ntc-day:2] Wait(SIP/gw1-0066, 1) in new 
stack
-- Executing [s@ivr-boi-ntc-day:3] Answer(SIP/gw1-0066, ) in new 
stack
-- Executing [s@ivr-boi-ntc-day:4] Dial(SIP/gw1-0066, 
SIP/1021,20,m) in new stack
  == Using SIP RTP CoS mark 5
-- Called 1021
-- Started music on hold, class 'default', on SIP/gw1-0066
-- SIP/1021-0067 is ringing
-- SIP/1021-0067 answered SIP/gw1-0066
-- Stopped music on hold on SIP/gw1-0066
-- Packet2Packet bridging SIP/gw1-0066 and SIP/1021-0067
  == Using SIP RTP CoS mark 5
-- Executing [12086597642@from-sip:1] Set(SIP/6908-0068, 
CALLERID(num)=2084331021) in new stack
-- Executing [12086597642@from-sip:2] Dial(SIP/6908-0068, 
SIP/gw1/12086597642,60) in new stack
  == Using SIP RTP CoS mark 5
-- Called gw1/12086597642
-- SIP/gw1-0069 is ringing
-- Started music on hold, class 'default', on SIP/gw1-0066
-- SIP/gw1-0069 is making progress passing it to SIP/6908-0068
  == Using SIP RTP CoS mark 5
-- Executing [6911@from-sip:1] Macro(SIP/1021-006a, 
stdexten,6911,sip/6911) in new stack
-- Executing [s@macro-stdexten:1] Wait(SIP/1021-006a, 1) in new 
stack
-- Executing [s@macro-stdexten:2] Dial(SIP/1021-006a, sip/6911,20) 
in new stack
  == Using SIP RTP CoS mark 5
-- Called 6911
-- SIP/6911-006b is ringing
-- Stopped music on hold on SIP/gw1-0066
  == Spawn extension (ivr-boi-ntc-day, s, 4) exited non-zero on 
'SIP/1021-006aZOMBIE'
  == Spawn extension (macro-stdexten, s, 2) exited non-zero on 
'SIP/gw1-0066' in macro 'stdexten'
  == Spawn extension (from-sip, 6911, 1) exited non-zero on 'SIP/gw1-0066'
-- SIP/gw1-0069 answered SIP/6908-0068
-- Packet2Packet bridging SIP/6908-0068 and SIP/gw1-0069

Thanks,

Kevin Oravits

From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, August 30, 2011 12:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up

Your Dial command stops the MOH - if the command were Dial(SIP/1021,20,m) the 
music would continue until connected or timed-out.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 2:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MOH making calls appear hung up

I noticed the CLI shows that the music on hold actually stops for some reason?

Here's the output of my CLI:
Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)
Verbosity is at least 28
-- Executing [s@ivr-boi-ntc-day:3] Answer(SIP/gw1-05d6, ) in new 
stack
-- Executing [s@ivr-boi-ntc-day:4] Wait(SIP/gw1-05d6, 1) in new 
stack
-- Executing [s@ivr-boi-ntc-day:5] Dial(SIP/gw1-05d6, SIP/1021,20) 
in new stack
  == Using SIP RTP CoS mark 5
-- Called 1021
-- SIP/1021-05d7 is ringing
-- SIP/1021-05d7 answered SIP/gw1-05d6
-- 

[asterisk-users] subscriptions from ekiga to asterisk

[rhododendronbusch]

Hello List!

I have small but strange problem with ekiga 3.2.7 (on Debian Squeeze as
well as on Win XP). For every contact I have listed in ekiga that has a
corresponding hint in the dialplan ekiga tries to subscribe to the hint
but fails. In asterisk the a message like this is shown for every
contact in ekiga:

[Aug 30 23:25:55] NOTICE[30755]: chan_sip.c:21489
handle_request_subscribe: Failed to authenticate device Michael
Hauptfeld sip:michi@10.8.0.6;tag=867d4550-bcd1-e011-94fe-f4ec38ab651d
for SUBSCRIBE

I wonder why there is ... sip:michi@...  as I have michi nowhere in
my ekiga setup, nor do I in the asterisk configuration.

In the extension.conf I have the following for most numbers:
[...]
exten = 210,hint,SIP/210
exten = 210,1,Dial(SIP/210,${INT_RINGTIME},tTxX)
exten = 210,n,VoiceMail(210,u)
[...]

In sip.conf:
[general]
...
;allowsubscribe=no  ; (Default is yes)
;notifyringing = no ; (default: yes)
notifyhold = yes;
notifycid = yes ;
callcounter = yes   ;
...

[int-std](!)
type=friend
secret=somesecret
host=dynamic
nat=no
subscribecontext=default
call-limit=10
canreinvite=no
callgroup=8
pickupgroup=8

[210](int-std)
callerid=Hennes 210
secret=...
[...]

I'm running Asterisk on Debian Squeeze from the debian repositories.

I don't know whether this is a problem of asterisk or ekiga. Seems to be
ekiga, but I hope somebody can help me here.

Sincerly,
Michael

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[asterisk-users] Transfer to VoiceMail Asterisk 1.6

[motty.cruz]

Hello,
I'm using Asterisk 1.6 with Polycom SoundPoint 650, everything is running
fine except that I can't program a button on Polycom to transfer inbound
call to Voicemail directly. 

I have the following in my extension.conf 

exten = _547xx,1,Voicemail(${EXTEN:1}@default,u)

Reception can transfer directly to VoiceMail when dialing digit 5 I want to
make a softkey on Polycom 650 does anybody know how to accomplish tranfering
directly to VoiceMail? 

Thanks, 
Motty


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Re: [asterisk-users] Polycoms rebooting themselves

[Gord Urquhart]

Take a look at the network traffic, things like arp storms etc. A lot of
noise on the net can cause reboots.  Even if you don't find anything try
turning on the storm filter (if it is not on already), its in the Settings
- Advanced- Administration - Network settings - Ethernet I think.
g

On Tue, Aug 30, 2011 at 2:20 PM, Mike Diehl mdi...@diehlnet.com wrote:

 On Tuesday 30 August 2011 3:14:50 pm Tim Nelson wrote:
  - Original Message -
 
   Well, we've taken the time to check out the wiring. It's only 3 years
   old and
   looks like the people who did it knew what they were doing. Nice work.
  
   Rebooting the cable modem, router, and switch didn't fix the problem.
  
   Also, we had an instance today where ALL of the phones went down
   within
   minutes of each other. The Internet connection was still active.
  
   Looks like more often than not, all of the phones die at the same
   time.
  
   Any other ideas?
 
  If they're all powered via PoE on the same switch, look to diagnosing the
  switch itself. Look for issues with heat (not enough cooling or
  circulation), or depending on the switch, you could be pulling too much
  power from the PoE module contained within. Does your switch's PoE module
  put out enough power for 'X' number of phones at 'Y' number of watts
 each?
 
  Either of these problems would lead to the switch shutting down or
  resetting the PoE module which causes your phone reboots.

 All of the phones are AC powered.  Either via an injector or wall outlet; I
 don't remember which.  Definitely NOT POE.


 --

 Take care and have fun,
 Mike Diehl.

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Re: [asterisk-users] Polycoms rebooting themselves

[isrlgb]

I'm just throwing in my 2c (I don't have polycom)

Are your phones auto provisioned then maybe the provisioning server is sending 
a reboot for some reason or maybe something on the server is sending a sip 
notify of reboot 
-Original Message-
From: Gord Urquhart gord...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 30 Aug 2011 15:26:59 
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Polycoms rebooting themselves

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Re: [asterisk-users] T.38 passthru on 1.8.5

[Fabian Borot]

Txs a lot Kevin.
I had just created and account on https://issues.asterisk.org/jira
Let me know if this is the right place to post both the pcap capture and the 
sip logs. If not please help me out creating the account in the right place so 
that I can provide all the information you need.
The sip debug logs I can post here but I need to change the real IPs, which is 
easy to do because it will be a text file.
I appreciate your time and effort in helping us find the roout cause.
Fborot

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 13:53:25 -0400




txs a lot for your explanation steve
so, it should work w/o spandsp fairly fine if we do not have a bad connection. 
I see that this version has a lot of fixes related to t.38
 but is the implementation already mature enough to guarantee a decent success 
rate with fax calls?


 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 13:15:19 -0400








will installing spandsp help with t.38 pass-through?


 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 11:42:41 -0400








both endpoints use public Ips, I just changed the real ones for the privates 
ones to protect our ips but made a mistake and left the dest as a pub and the 
orig as private, my bad.
but for the record, both are public IPs, there is no nat and iptables is off

 also, I see that the quintum sends a lot of these packages but asterisk sends 
only 1 or 2 to the other side.




 

From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 09:44:15 -0400










 Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built 
by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 
21:31:22 UTC]

The call flow is:
quintum gateway -- asterisk -- Dialogic IMG 1010

the call starts as a voice call, the remote fax picks up and we hear the fax 
tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is 
passed back to the user side [quintum gateway] whcih reply with 200 OK with 
t.38 and the nothing else happens. After 20 secs of inactivity the quintum 
sends another Invite with voice only and then a BYE.

We do see that the quintum sends a lot of messages like this from the quintum's 
IP [192.168.1.18] but we do not see that asterisk sends the packages to the 
destination

UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, 
len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)
 UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6)

we have this settings on sip.conf
faxdetect = yes
t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations 
t38pt_udptl = yes;t38pt_udptl = yes,fec etc]

When we send the fax from the quintum to the Dialogic IMG the fax works 100% of 
the times.
I enabled fax set debug on and udptl set debug on but the console does not show 
almost anything but the udptl packets shown above.
What else should I do?Any ideas/help is greatly appreciated

txs a lot
fborot



  
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Re: [asterisk-users] Polycoms rebooting themselves

[Michael L. Young]

- Original Message -
 From: Mike Diehl mdi...@diehlnet.com
 To: asterisk-users@lists.digium.com
 Sent: Tuesday, August 30, 2011 5:13:22 PM
 Subject: Re: [asterisk-users] Polycoms rebooting themselves
 
 Well, we've taken the time to check out the wiring.  It's only 3
 years old and
 looks like the people who did it knew what they were doing.  Nice
 work.
 
 Rebooting the cable modem, router, and switch didn't fix the problem.
 
 Also, we had an instance today where ALL of the phones went down
 within
 minutes of each other.  The Internet connection was still active.
 
 Looks like more often than not, all of the phones die at the same
 time.
 
 Any other ideas?
 
 Mike.

How latest(ish) is the firmware?  I see in the release notes for 3.3.2 under 
corrections:

61147: SoundPoint IP 331, 335, 450, 550, 560, 650, 670: SoundStation IP 5000:
Phone reboots when a GET request is sent to the phone to
/TA/getParam?paramName=reg.1.ringType.

68063: Phone reboots when DHCP failover occurs.

(I know you have 335s but someone else mentioned they had issues with 550s)
70988: SoundPoint IP 550: Phone when powered by external AC power reboots
during playing of certain audio on full volume.

Just some things that came to mind when I saw your email.  I had just recently 
reviewed the release notes and they were fresh in my mind.

Hope you find a fix soon.

Regards,

Michael
(elguero)

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