Re: [asterisk-users] Asterisk inside network. What phone works well?

[Ryan Wagoner]

On Thu, Oct 13, 2016 at 12:06 PM,  wrote:

> > I have Asterisk running well inside our network. I did some
> > experiments exposing it to internet but had some issues:
> > 1. NAT issues (voice one way, etc). From what I understand double-
> > NAT users will always have something like this
> > 2. Immediately I see people trying to hack into. I did configure
> > Fail2Ban and it works somewhat, but not 100%. Erroneous logs, etc
> >
> > So.. I ended up closing network. Currently most users inside
> > network. My home router have GRE tunnel to office so phone works just
> fine.
> > Another user uses VPN and soft phone and it works good too.
> >
> > Now I need to setup some users with actual phone devices and none of
> > those solutions will work. So, I did some research and found
> > that some phones have VPN capability built in.
> >
> > Right now I use Cisco SPA504G phones. We have auto-provisioning for
> > them, works well. But I don’t think they have VPN capability.
> >
> >
> > What I found it that Cisco 525g2 has AnyConnect functionality (SSL
> > VPN) but not sure if this is what I need.
> >
> > We have Mikrotik router. Can I setup VPN on router and have this
> > Cisco phone auto-dial VPN and then connect to Asterisk? I’m asking
> > to see if this will work before I go in and buy that phone.
> > Or maybe there is other devices/solutions you suggest? I’d like to
> > stay with Cisco because I’m somewhat familiar with provisioning those..
>
> I haven't done this myself, but I think what you need to look at is phones
> that can do IPSEC vpn setups.
>
> For the Mikrotik router, this may be helpful to start investigating:
> http://wiki.mikrotik.com/wiki/L2TP_%2B_IPSEC_between_
> Mikrotik_router_and_a_PC
>

I have Asterisk installs behind Vyatta (linux iptables) and pfSense
(freebsd pf) NAT routers and majority of the time there are no issues with
phones outside the network. My go to phones are Polycom VVX series or
X-Lite / Bria softphones. The key is to make sure you have configured
Asterisk sip.conf with the externip= and nat=yes settings. Additionally on
the NAT routers that the outside phones are behind SIP ALG should be
disabled.

Ryan
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Re: [asterisk-users] small pbx for the office [it was: small homebrew pbx]

[Ryan Wagoner]

On Wed, Jun 17, 2015 at 9:07 AM, lu...@sulweb.org wrote:

 Lukasz Sokol wrote:

 but have you considered a web-managed config-builder such as FreePBX?
 Instead of building your dialplan from scratch ?


 I've never used FreePBX, but, after having looked at its website, I think
 I have a general understanding of what it can do. What I don't understand
 is how FreePBX answers my question about the Linksys SPA3102 being good for
 a mission critical solution or not.


I've used the SPA3102 and would recommend it for home use. For business
look at the Patton SmartNode 4110 series devices or a Cisco router with FXO
card and DSP modules. I have deployed both and haven't had any complaints.
They just work once configured.

Ryan
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Re: [asterisk-users] Strange Polycom Issue

[Ryan Wagoner]

On Mon, Mar 9, 2015 at 9:40 AM, David Wessell da...@ringfree.biz wrote:

 Welcome to our hell.

 We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally
 got Polycom to issue a hotfix firmware version. I'll be happy to share it
 with you offlist, just email me.

 Officially Polycom will fix the issue in 5.3 in a few months..

 Thanks
 David


Could this be a 5.2.x issue only? I have a hundred of the VVX 400 phones
running 4.1.7 and haven't heard of this issue yet from our users.

Thanks,
Ryan
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Re: [asterisk-users] chan_sip and 2 devices under same extension - transferring call endpoint(s)

[Ryan Wagoner]

On Mon, Dec 29, 2014 at 7:26 AM, Lukasz Sokol el.es...@gmail.com wrote:

 As the handsets have no LCD's to show the dialled number,
 I want to give the workforce the ability to dial OUT using the softphone,
 (as in, copy/paste the number from the CRM software into softphone then
 *immediately* transfer the originated call 'endpoint' to the handset of
 the same 'user' extension, somehow,
 the question is, HOW ?


We use FreePBX and a custom CRM. What we do is use the Asterisk Manager
interface to create a call using the originate command. Asterisk dials the
users handset, once they answer Asterisk then dials the outbound number. No
need for any transferring. You could also look at Asterisk call files to
originate the call.

Ryan
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Re: [asterisk-users] Asterisk LTS segment faults

[Ryan Wagoner]

On Wed, Oct 8, 2014 at 9:35 AM, Grant Bagdasarian g...@cm.nl wrote:

 Hello,



 Does anyone know how frequent segment faults occur in the current LTS
 release (version 11) and in the future LTS release (version 13)?

 We are currently using 1.6, which frequently throws unexplained segment
 faults, that’s why we are considering to upgrade to the latest LTS version.





I was having crashes at least once a month with Asterisk 1.6. Each time I
would upgrade to fix one issue another would appear. I moved to Asterisk
1.8 LTS when it was released and haven't looked back. I have around 700
endpoints registered and we handle over 10k calls per day. Even with
Asterisk 1.8 I was running into a hung channels every few months when using
a Sangoma card with chan_dahdi. About 6 months ago I switched over to a
Cisco gateway for the PRIs and am only using chan_sip with Asterisk. The
result has been rock solid performance.

Ryan
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[asterisk-users] Certified Asterisk 11.6 Menuselect

[Ryan Wagoner]

Has there been a change in the way certified Asterisk is being packaged?
Starting with certified Asterisk 11.6 has all the extended options are
checked by default in menuslect? Certified Asterisk 11.2 does not have them
checked and neither does certified Asterisk 1.8.15?

Thanks,
Ryan
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Re: [asterisk-users] DUNDi with SIP Mapping

[Ryan Wagoner]

On Wed, Apr 16, 2014 at 10:20 AM, Kevin Larsen 
kevin.lar...@pioneerballoon.com wrote:


 You are a bit outside of what I have done, but this looks like it might be
 what you want to do with SIP:
 http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP


I had looked at that guide before, but couldn't get it working. I could do
SIP without authentication. This would have worked if I only wanted to
terminate calls to extensions. For future purposes I wanted to include PSTN
routes. In the end I went with IAX and have it up and running. It was
actually simple to integrate with FreePBX. The important piece was setting
ttl to 1 to prevent DUNDi lookup loops, which would cause the box to
sometimes see its own DUNDi extensions.

The one FreePBX box with the PRI will try 10 digits numbers on DUNDi
private then go out the PRI. The other FreePBX boxes try to dial 10 digit
numbers on DUNDi private then use DUNDi to reach the PSTN. This allows me
to add additionally FreePBX boxes with PSTN connections and use weights.
Additionally providing a separate mapping for the PSTN allows toll free to
first try DUNDi private, then a VoIP provider, then the DUNDi PSTN.

cd /var/lib/asterisk/keys
astgenkey -n `hostname -f`
chown asterisk:asterisk *

share .pub keys between all servers

vim /etc/asterisk/dundi.conf
cachetime=60
ttl=1

priv = dundi-extens,0,IAX2,dundi:${
SECRET}@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
priv = dundi-dids,100,IAX2,dundi:${
SECRET}@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
pstn = dundi-via-pstn,400,IAX2,dundi:${
SECRET}@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial

;[EID of remote DUNDi peer]
;model = symmetric
;host = IP or FQDN of remote DUNDi peer
;inkey = public key of remote DUNDi peer, without .pub
;outkey = private key of local PBX, without .key
;include = all
;permit = all
;qualify = yes

vim /etc/asterisk/extensions_custom.conf
[dundi-local]
include = dundi-extens
include = dundi-dids
include = dundi-via-pstn

[dundi-local-keepcid]
exten = _X.,1,Set(KEEPCID=TRUE)
exten = _X.,n,Goto(dundi-local,${EXTEN},1)

[dundi-extens]
include = ext-queues
include = ext-findmefollow
include = ext-group
include = ext-local

[dundi-dids]
include = ext-did-0002

[dundi-via-pstn]
include = outbound-allroutes

FreePBX Trunks
Type: DUNDi
Trunk Name: DUNDi Private
DUNDi Mapping: priv

Type: DUNDi
Trunk Name: DUNDi Pstn
DUNDi Mapping: pstn

Type: IAX
Trunk Name: DUNDi
Outgoing Settings:
Trunk Name: dundi
PEER Details:
type=friend
dbsecret=dundi/secret
disallow=all
context=dundi-local-keepcid
allow=ulawg729

FreePBX Outbound Routes
Route Name: dundi
Route Type: Intra-Company
Dial Pattern: NXXX
Trunk: DUNDi Private

Route Name: outbound
Dial Pattern: 1NXXNXX
Dial Pattern: NXXNXX
Trunk: DUNDi Private
Trunk: PRI or DUNDi Pstn
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Re: [asterisk-users] Live Recording on the Storage Server?

[Ryan Wagoner]

On Thu, Apr 17, 2014 at 11:52 AM, Bryant Zimmerman brya...@zktech.comwrote:

A simple way that we use to do the move is to create a cron job that looks
 for a .move file.
 It has the same name as the recorded file. asterisk writes the .move file
 which is just a text file with some stats in it.
 The .move file is written from the dial plan  at the end of the recording.
 In the exten = h we write a .delete file for an abandon call.

 The cron then processes the .move and .delete files at a given interval.
 We actually write special instructions into our .move files that the cron
 parses and can then act accordingly. So we have a single smart cron job
 handling moves for each type of task. In some cases our .delete files are
 processed as moves to an abandon cache for recovery if a customer did not
 intend to abandon it.

 The sky's the limit on how complex you want to make it, but in the long
 run it is fairly simple and it just works.

 Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


We record locally and move the files to the storage server with a cron job
once a minute. The script uses lsof to check to see if Asterisk is writing
to the file.

/usr/sbin/lsof | grep filename | wc -l

Thanks,
Ryan
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[asterisk-users] DUNDi with SIP Mapping

[Ryan Wagoner]

From the reading and testing I have done it doesn't look like SIP supports
a username and password in the Dial string. I currently have the following
mapping.

priv = dundi-extens,0,SIP,
dundi:pass@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial

On the sending side I see

NOTICE[31598] chan_sip.c: Conflicting extension values given. Using 'dundi'
and not '1001'

On the receiving side it will not match the SIP dundi user and tries to
call dundi instead of 1001.

-- Executing [dundi@from-sip-external:1] NoOp(SIP/1.1.1.2-,
Received incoming SIP connection from unknown peer to dundi) in new stack


Is there a way to configure DUNDi to use SIP or does it only work with IAX?

Thanks,
Ryan
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Re: [asterisk-users] DUNDi with SIP Mapping

[Ryan Wagoner]

On Wed, Apr 16, 2014 at 9:06 AM, Kevin Larsen 
kevin.lar...@pioneerballoon.com wrote:


 I am using DUNDi with SIP to do some least cost routing amongst my various
 locations. My mapping is close to what you have:

 priv = dundi-extens,0,SIP,trunk_name/number_to_dial

 Where trunk_name is replaced with the actual name of my trunk as defined
 in sip.conf and number_to_dial is the number they should dial on that
 trunk. I have not tried to define the SIP username/password in the DUNDi
 config itself, so I don't know if what you are trying to do is possible or
 not.


I was trying to avoid having to define the SIP trunks on all systems. I
currently have three FreePBX systems connected by SIP trunks with 800 DIDs.
Each system has SIP trunks defined to both other systems and routes
defining the extensions / DIDs. As I add more DID blocks and FreePBX
systems maintaining the trunks and routes is going to become cumbersome.

I wanted to move to DUNDi to simplify the setup. It looks like I need to
switch to IAX trunks to be able to do this.

Thanks,
Ryan
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Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450

[Ryan Wagoner]

On Thu, Jan 2, 2014 at 11:13 AM, Eric Wieling ewiel...@nyigc.com wrote:

 Which firmware version?  4.1.x is only for use with MS Link server.  A
 symptom of running 4.1.x firmware with a non-MS server is the phone will
 not show buddies.


I'm running 4.1.0 on a Polycom IP 335 and IP 550 and version 4.1.5 on a
Polycom VVX 400. Buddies work on all three phones. The firmware is for both
SIP and Lync. You change the base profile option accordingly. Look in the
Polycom UC Software Admin Guide for more information.

Ryan
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Re: [asterisk-users] Maximum number of users

[Ryan Wagoner]

On Wed, Dec 18, 2013 at 9:45 AM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hello;

 Can someone advise me what is the maximum number of users (IP Phones) that
 can be supported by asterisk 1.8 or later?

 Regards
 Bilal


The number of devices and concurrent calls is dependent on many factors.
Dialplan complexity, new call rate, features enabled, and transcoding all
play a factor in these numbers.

To give you an example I have a Dell R710 with two quad core E5520
processors running Asterisk 1.8 and FreePBX 2.11. I have around 1,000 SIP
device registrations, 50-80 concurrent calls for the majority of the day,
and a total of 8-10k calls processed per day. A few times a week I will see
the last minute load at 20 and the 5 min load at 7. This seem to happen
when there are a high volume of new calls as the FreePBX dialplan is
complex.

Ryan
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Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

[Ryan Wagoner]

On Sun, Dec 15, 2013 at 5:07 AM, jg webaccou...@jgoettgens.de wrote:

 I think the order or elements is relevant:

 [100]
 disallow=all
 allow=ulaw
 allow=g722
 or
 [100]
 allow=!all,ulaw,g722

 should work.

 jg


If I choose that order and the phone supports both ulaw and g722 only ulaw
will be used. I want to use g722 when available on both devices, fallback
to ulaw without transcoding if both devices support it, or transcode if
only one device supports ulaw.

I looked at the code more and here is what happens. Device 100 dials 101.
The sip_new function is called and AST_CODEC_CHOOSE g722 is set as the
read/write format.

[2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7911 sip_new:
*** Our native formats are (g722)
[2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7912 sip_new:
*** Joint capabilities are (ulaw|g722)
[2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7913 sip_new:
*** Our capabilities are (ulaw|g722)
[2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7914 sip_new:
*** AST_CODEC_CHOOSE formats are g722

Dial 101 is executed in the dialplan, sip_request_call is called, which in
turn calls sip_new. The AST_CODEC_CHOOSE g722 from above becomes the
incoming preferred format. We can only have one preferred format as
sip_request_call takes in struct ast_format_cap *cap.

[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7911 sip_new:
*** Our native formats are (ulaw)
[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7912 sip_new:
*** Joint capabilities are (nothing)
[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7913 sip_new:
*** Our capabilities are (ulaw)
[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7914 sip_new:
*** AST_CODEC_CHOOSE formats are ulaw
[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7916 sip_new:
*** Our preferred formats from the incoming channel are (g722)

Asterisk tries to find a common codec between this channels capabilities
and the incoming channel preferred format. Of course there are none (g722
and ulaw don't match) so we pick ulaw and transcode. What I am proposing is
Asterisk passes fallback formats to sip_request_call. If the joint
capabilities are none, then check the fallback formats. In this case it
would be ulaw and ulaw. If there is a match switch the incoming channel to
that format (ulaw) and AST_CODEC_CHOOSE would be ulaw this for channel.
However I'm not sure how to make this change as I don't know my way around
the interaction with the Asterisk core and the channels.

Ryan
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Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

[Ryan Wagoner]

On Sun, Dec 15, 2013 at 7:20 AM, jg webaccou...@jgoettgens.de wrote:

 I see, you do want something like picking g722 provided there is no
 transcoding. Because Asterisk is a B2BUA it can transcode, so it would
 choose g722 where the other party is doing g711.

 For known parties, maybe one could change the SIP configuration on the fly
 using the Asterisk realtime engine, or modify the settings of the phone
 with an http request. Generally, an Asterisk configuration option like
 prioritize_matching_codecs would be needed, but I don't think this is
 very useful. In this case there should also be all sound files available in
 g722. Even if you have them, some channels might still be silent as
 sometimes users choose to get MOH, for example, from the phone itself.
 Phones usually store sound files in a single format assuming that somebody
 else is able to transcode if necessary.

 Please correct me, if my description is incorrect.

 jg


You are correct. Your idea of the prioritize_matching_codecs option is what
I am looking for. Yes Asterisk can transcode, but why transcode when you
don't need to. If the phone is advertising both formats it should support
them. If the phone only supports local MOH in one format then the phone
should only advertise that format.

If Answer and Playback are called first then the format would have already
been sent back in the 200 OK and Asterisk would transcode when Dial is
called. If Dial is called first, change the format for the 200 OK and use
it for the rest of the call. I haven't looked into what happens with
transfers.

The idea comes from the following setup. I have 450 users on a FreePBX /
Asterisk server with a Sangoma transcoding card. However I am limited in
the number of sessions. I also have a number of smaller 10-50 user
deployments without transcoding cards.

Remote users have phones with g729
Local users have phones with g722,ulaw,g729
SIP Trunks with ulaw,g729
PRIs with ulaw

Remote to local should use g729
Local to local should use g722
Remote to SIP trunk should use g729
Local to SIP trunk should use ulaw
Local to PRI should use ulaw
Remote to PRI would transcode g729 to ulaw

If I set these codecs on the devices depending on which side initiates that
call transcoding occurs more often than I would like. I could reverse the
codec order, however a lower bandwidth codec is chosen in cases where I
would prefer a higher bandwidth codec.

I looked at this a year ago on Asterisk 1.8 and ended up using ulaw for
everything but remote phones. The remote phones end up transcoding g729 to
ulaw for most calls.

Ryan
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Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

[Ryan Wagoner]

On Sun, Dec 15, 2013 at 9:32 AM, jg webaccou...@jgoettgens.de wrote:

 Is it possible to let the Sangoma card work only on the most demanding
 codecs? This requires some analysis to estimate the benefits. Another
 question is whether the user phones are provisioned or not. If provisioned,
 then you are the maker of rules.


Most users have both a desk Polycom phone and a soft phone on their mobile
device or laptop. I don't have control over how the soft phones are
provisioned on mobile devices. I've found a workaround that prevents
transcoding for outbound calls.

remote phone
allow=g729

local phone
allow=ulawg729

trunk
allow=ulawg729

In FreePBX extensions_custom.conf I've added the following. This tries to
force the outbound channel to match the inbound channel's format.

[macro-dialout-trunk-predial-hook]
exten =
s,1,Set(_SIP_CODEC_OUTBOUND=${CHANNEL(audionativeformat):1:$[${LEN(${CHANNEL(audionativeformat)})}-2]})

Remote to local g729 pass through
Local to remote g729 transcoding
Local to trunk ulaw pass through
Remote to trunk g729 pass through (addressed by the
dialout-trunk-predial-hook)
Trunk to local ulaw pass through
Trunk to remote g729 transcoding

Alternatively I could set trunk allow=g729,ulaw, which would prevent
transcoding for all inbound calls. Outbound from the local phone would use
the hook to change to ulaw.

I still don't have a way to enable the higher quality g722 codec for
internal use without making a transcoding mess. Maybe Asterisk 12 with
pjsip will have a better solution.

Ryan
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[asterisk-users] Why doesn't Asterisk try to prevent transcoding

[Ryan Wagoner]

Let's say I have two devices configured and the follow call scenarios occur.

[100]
disallow=all
allow=g722ulaw

Polycom phone with g722,ulaw,alaw,g729

[101]
disallow=all
allow=ulaw

Polycom phone with g722,ulaw,alaw,g729

101 dials 100 - ulaw to ulaw is chosen
100 dials 101 - g722 to ulaw is chosen

Ideally when 100 dials 101 ulaw would be chosen since it is the common
format. Looking into this deeper

Device 100 sends INVITE to Asterisk offering g722,ulaw,alaw,g729
Asterisk sends INVITE to device 101 offering ulaw
Device 101 sends 200 OK to Asterisk offering ulaw
Asterisk sends 200 OK to device 100 offering g722,ulaw

I can prevent transcoding by adding SIP_CODEC_INBOUND=ulaw to the dialplan
for extension 101. This causes Asterisk to send 200 OK to device 100
offering ulaw. Am I missing why Asterisk wouldn't just offer the highest
priority codec they have in common to prevent transcoding?

Ryan
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Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

[Ryan Wagoner]

On Sat, Dec 14, 2013 at 10:31 PM, Ryan Wagoner rswago...@gmail.com wrote:

 Let's say I have two devices configured and the follow call scenarios
 occur.

 [100]
 disallow=all
 allow=g722ulaw

 Polycom phone with g722,ulaw,alaw,g729

 [101]
 disallow=all
 allow=ulaw

 Polycom phone with g722,ulaw,alaw,g729

 101 dials 100 - ulaw to ulaw is chosen
 100 dials 101 - g722 to ulaw is chosen

 Ideally when 100 dials 101 ulaw would be chosen since it is the common
 format. Looking into this deeper

 Device 100 sends INVITE to Asterisk offering g722,ulaw,alaw,g729
 Asterisk sends INVITE to device 101 offering ulaw
 Device 101 sends 200 OK to Asterisk offering ulaw
 Asterisk sends 200 OK to device 100 offering g722,ulaw

 I can prevent transcoding by adding SIP_CODEC_INBOUND=ulaw to the dialplan
 for extension 101. This causes Asterisk to send 200 OK to device 100
 offering ulaw. Am I missing why Asterisk wouldn't just offer the highest
 priority codec they have in common to prevent transcoding?

 Ryan


I should have mentioned I'm using Asterisk 11.2-cert2. The core debug from
the above shows

[2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7911 sip_new:
*** Our native formats are (g722)
[2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7912 sip_new:
*** Joint capabilities are (ulaw|g722)
[2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7913 sip_new:
*** Our capabilities are (ulaw|g722)
[2013-12-14 22:51:59] DEBUG[25200][C-004d]: chan_sip.c:7914 sip_new:
*** AST_CODEC_CHOOSE formats are g722

[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7911 sip_new:
*** Our native formats are (ulaw)
[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7912 sip_new:
*** Joint capabilities are (nothing)
[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7913 sip_new:
*** Our capabilities are (ulaw)
[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7914 sip_new:
*** AST_CODEC_CHOOSE formats are ulaw
[2013-12-14 22:51:59] DEBUG[27830][C-004d]: chan_sip.c:7916 sip_new:
*** Our preferred formats from the incoming channel are (g722)

I'm looking at the code now. I am hoping to write a patch, if I can wrap my
head around the code, to determine join capabilities between the joint
capabilities of each channel. If this exists then set both channels this
codec.

Ryan
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Re: [asterisk-users] Asterisk on Windows

[Ryan Wagoner]

On Wed, Dec 4, 2013 at 10:19 AM, CDR vene...@gmail.com wrote:

 Digium is 100% lost in the map. If they would come up with a Paid
 version of Asterisk, one that would use the .NET framework in Windows,
 something simple to install, they could go public on the product.
 Linux has a very steep learning curve. A Windows application that
 would do exactly the same would be a home run. Note: I am a Linux
 expert user, but it took me years to get here. And still, moving from
 regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. The .NET
 framework and Windows server 2012 are miles away in terms of
 friendliness and on equal footing on performance. I don´t mean another
 slow cygwin port, I man a native Asterisk for windows. In fact, I
 would invest on the project if somebody wants to do it.


Windows and Linux should be able to coexist. I have had great success
setting up a VMware ESXi server with Windows VMs for AD and Exchange and
Linux VMs for Asterisk and Web / FTP. Asterisk with Exchange UM for
voicemail is a winning combination and works seamlessly. It is essentially
a private cloud of the customer. Why not use the OS that works for the task
at hand?

Ryan
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Re: [asterisk-users] Trouble with upgrading - RBS T1

[Ryan Wagoner]

I have a system with two Sangoma A104D cards running Asterisk
1.8.11-cert10, Dahdi 2.5.0.1, LibPRI 1.4.12, and Wanpipe 3.5.23. The PRI
spans are configured with esf,b8zs. Everything has been working great,
which is why I haven't updated it further. You might try an older Dahdi
version just to see. Although this might be tricky depending on the OS
version.

Ryan


On Mon, Dec 9, 2013 at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote:

  Upgrading an ancient customer installation... was running 1.4.23.1
 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running
 fine for 5+ years.  Customer getting anxious about hardware failure, so we
 built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma
 A104D.  The single active span is an RBS T1 B8ZS/ESF/EM Wink.

 I tried to move one span over one night which was working fine on the old
 box.  Once plugged in there were no alarms, Sangoma wanpipemon utility
 showed connected.  I tried calling in on a DID number, and in the 'full'
 log, with debug and verbose set to 100:

 [Dec  5 00:51:37] VERBOSE[5283] sig_analog.c: -- Starting simple
 switch on 'DAHDI/9-1'
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.17E+04, Et=1.45E+06,
 s/n=  0.01
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.76E+03, Et=1.10E+06,
 s/n=  0.00
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.06E+04, Et=1.39E+06,
 s/n=  0.01
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=4.68E+03, Et=1.40E+06,
 s/n=  0.00
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00,
 s/n=  -nan
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00,
 s/n=  -nan
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00,
 s/n=  -nan
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00,
 s/n=  -nan
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=0.00E+00, Et=0.00E+00,
 s/n=  -nan
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.30E+10, Et=2.11E+12,
 s/n=  0.01
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: analog_exception 9
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: Exception on 15, channel 9
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: __analog_handle_event 9
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: Got event
 UNKNOWN/OTHER(131127) on channel 9 (index 0)
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: DTMF Down '7'
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: Begin DTMF digit: 0x37 '7' on
 DAHDI/9-1
 [Dec  5 00:51:38] DEBUG[5283] chan_dahdi.c: Begin DTMF digit: 0x37 '7' on
 DAHDI/9-1
 [Dec  5 00:51:38] DTMF[5283] channel.c: DTMF begin '7' received on
 DAHDI/9-1
 [Dec  5 00:51:38] DTMF[5283] channel.c: DTMF begin ignored '7' on DAHDI/9-1
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.02E+10, Et=4.01E+12,
 s/n=  0.01
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.88E+10, Et=3.89E+12,
 s/n=  0.00
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=4.78E+10, Et=1.17E+12,
 s/n=  0.04
 [Dec  5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=5.10E+03, Et=6.26E+06,
 s/n=  0.00
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: analog_exception 9
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: Exception on 15, channel 9
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: __analog_handle_event 9
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: Got event
 UNKNOWN/OTHER(262199) on channel 9 (index 0)
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: Detected digit '7'
 [Dec  5 00:51:38] DEBUG[5283] sig_analog.c: End DTMF digit: 0x37 '7' on
 DAHDI/9-1
 [Dec  5 00:51:38] DEBUG[5283] chan_dahdi.c: End DTMF digit: 0x37 '7' on
 DAHDI/9-1
 [Dec  5 00:51:38] DTMF[5283] channel.c: DTMF end '7' received on
 DAHDI/9-1, duration 0 ms
 [Dec  5 00:51:38] DTMF[5283] channel.c: DTMF end accepted without begin
 '7' on DAHDI/9-1
 [Dec  5 00:51:38] DTMF[5283] channel.c: DTMF end passthrough '7' on
 DAHDI/9-1
 [Dec  5 00:51:38] DEBUG[5283] chan_dahdi.c: Enabled echo cancellation on
 channel 9
 [Dec  5 00:51:38] VERBOSE[5283] sig_analog.c: -- Unknown extension '7'
 in context 'from-pstn' requested
 ...

 At this point I hear 'invalid extension' and get hung up on, but if you
 grep out all the DTMF events from this call, you get:

 root@astsouth:/var/log/asterisk# grep 'DTMF end' /tmp/foo | grep received
 [Dec  5 00:51:38] DTMF[5283] channel.c: DTMF end '7' received on
 DAHDI/9-1, duration 0 ms
 [Dec  5 00:51:41] DTMF[5283] channel.c: DTMF end '1' received on
 DAHDI/9-1, duration 80 ms
 [Dec  5 00:51:41] DTMF[5283] channel.c: DTMF end '5' received on
 DAHDI/9-1, duration 80 ms
 [Dec  5 00:51:41] DTMF[5283] channel.c: DTMF end '7' received on
 DAHDI/9-1, duration 80 ms
 [Dec  5 00:51:41] DTMF[5283] channel.c: DTMF end '6' received on
 DAHDI/9-1, duration 80 ms
 [Dec  5 00:51:41] DTMF[5283] channel.c: DTMF end '0' received on
 DAHDI/9-1, duration 80 ms
 [Dec  5 00:51:41] DTMF[5283] channel.c: DTMF end '0' received on
 DAHDI/9-1, duration 80 ms

 And '715-7600' is the 

Re: [asterisk-users] SIP Presence across two servers

[Ryan Wagoner]

I haven't tried it, but the res_corosync module states it will sync device
state across servers.

https://wiki.asterisk.org/wiki/display/AST/Corosync


On Thu, Nov 14, 2013 at 3:54 AM, Leandro Dardini ldard...@gmail.com wrote:

 Aligning presence over multiple servers is not simple and require some
 changes on the dialplan and some custom code to transmit the state from one
 server to the other.

 The BLF on the phone is displayed using the hint of an extension. To be
 able to manually manage the hint of an extension, you need to first link
 the internal hint to the Custom hint. In the extensions.conf just add:

 exten = _.,hint,Custom:${EXTEN}

 I was unable to create the same entry in the AEL language or in the
 realtime extensions table... if any was able, I will appreciate.

 If a phone want to know the status for the 100-TEST sip account, it will
 poll the hint for 100-TEST and in the end, it will check the status for
 Custom:100-TEST.

 Now you need an application to capture the change in status of every
 extension on server A and send it to server B, so the Custom:100-TEST will
 have the same value on both servers.

 I solved this problem creating a small pair of php application, using
 Asterisk Manager Interface to continuously listen to events. If I see a
 phone dialing out, I change its Custom state to IN_USE... if he hangups, I
 change the state back to AVAILABLE ... if it is ringing, I change the state
 in RINGING and so on. You need to take into account multiple calls can be
 made by the same phone and so it is not really so straightforward. When the
 php AMI application identify a change in the state for a phone, it notifies
 the same application running on the other server about the change, so both
 asterisk are taken aligned.

 Let me know if you need additional details.

 Leandro



 2013/11/13 Lincoln King-Cliby linc...@controlworks.com

 Hi All,



 We’ve been running Asterisk for years in our offices but just recently
 replaced an Asterisk Appliance* in our smaller office with an actual
 server, upgraded the server in hardware in our HQ location and upgrading
 both ends to 11.5.0 with Gareth’s patch for Cisco phones.

 99.99% of our endpoints are Cisco 7961Gs.



 Each office is more-or-less standalone for ease of management and fault
 tolerance but we have a unified dialplan and SIP “trunking” from site to
 site via our VPN.



 Everything presence related works wonderfully for local users, but I’m
 hoping there’s a way we could get presence for the people “at the other end
 of the pipe” fairly transparently.

 We have a lot of cross-office collaboration, and our office
 manager/receptionist (who has the battleship of a 7961G+7914+7914 BLF)
 would love to “at a glance” know if the remote folks are available for a
 call or not.



 I’m sure this has been covered, but my Googlefu us turning up a ton of
 redundant, old, and deprecated information so I’ve resorted to asking here.

 From what I have found it sounds like it may be “easier” with IAX2 but my
 experiments with IAX2 haven’t yielded wonderful results and management
 prefers “SIP everywhere”



 If anyone has any pointers I’d greatly appreciate it – thanks in advance!



 Lincoln



 *- One of the worst IT decisions I’ve made for better or worse. Looked
 good on paper; in practice not a good idea for anything beyond a very
 simple SOHO.

 --

 Lincoln King-Cliby, CTS, DMC-D, CCMP-S

 Commercial Market Director

 Sr. Systems Architect | Crestron Certified Master Programmer (Silver)

 V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com

 Crestron Services Provider



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Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

[Ryan Wagoner]

On Sat, Aug 4, 2012 at 1:22 PM, Shahid H shah...@gmail.com wrote:

 Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to
 do 200 calls recordings.

 Once the call hangup/completed it will then move recording file to SATA
 HDD.

 What do you think of this?




You want some form of raid for redundancy. I usually go with two 15K SAS
drives in raid 1 or four 7.2k SATA drives in raid 10. Performance between
the two should be similar. With drives being as cheap as they are skip raid
5.

Ryan
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Re: [asterisk-users] T.38 gateway patch against Asterisk 1.8.11.0

[Ryan Wagoner]

On Fri, Mar 30, 2012 at 1:16 PM, Bryant Zimmerman brya...@zktech.comwrote:

 What does this patch fix? Why is it not in Jarr?

 Thanks

 Bryant


It looks like the patch is a backport of the t.38 gateway functionality in
Asterisk 1.10.

Ryan
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Re: [asterisk-users] Stuck DAHDI Lines

[Ryan Wagoner]

2012/2/9 Antonio Modesto mode...@isimples.com.br

 **
 Hi,

 Sometimes some of my dahdi channels become stuck, It is very strange,
 here is the output of the core show channels command:

 pabx*CLI core show channels
 Channel  Location State
 Application(Data)
 Local/104@ramais-cc0 104@ramais:1 Up  Transferred
 Call(DAHDI/13-1)
 Local/104@ramais-cc0 ~~s~~@dial_dahdi:15  Up  Dial(DAHDI/10/
 91208788,120,T
 DAHDI/10-1   (None)   Up  AppDial((Outgoing
 Line))
 DAHDI/13-1   s@from_celular:1 Up  Transferred
 Call(Local/104@ram


 When I issue a hangup request DAHDI/10-1 in this case, All the other
 channels are cleaned, I don't know what can be causing it.


 Regards.


I had this happen every few weeks with Asterisk 1.6.1.18, DAHDI 2.3, and
libpri 1.4.10. We do use local channels extensively. When it happened
running core show channels would hang the asterisk console without only a
partial output of the command. I would have to ctrl+c and reopen to issue
another command. Additionally some inbound calls wouldn't complete if they
happened to hit the stuck channel.

I have since moved to Asterisk 1.8.7, DAHDI 2.5, and libpri 1.4.12. I'm
currently up to 16 weeks, 2 days of uptime and 914,745 calls processed
without a stuck channel.

Ryan
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Re: [asterisk-users] Hang up phone after declined attended transfer

[Ryan Wagoner]

On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote:

 We have a customer who has asked us to change this behavior, but I haven't
 been able to find a way to do it.  Server is Asterisk 1.6 and the phones
 are SPA 303 and 504.

 Receptionist gets an outside call, starts an attended transfer
 The office person being called answers by pressing the speaker button
 Declines to take the call
 Receptionist goes back to the original outside call by pressing the line
 button
 The office phone goes to hold instead of hanging up

 If the receptionist hangs presses the hookswitch instead of the line
 button, then it does hang up the call to the internal office phone, however
 that phone then goes into reorder tone.

 --
 Carlos Alvarez
 TelEvolve
 602-889-3003


When the receptionist presses the hookswitch it should hang up the remote
internal phone. Playing the reorder tone is due to a setting on the SPA
phone. I had to change this for a client that used the SPA phones and I'm
drawing a blank as to which setting.

Ryan
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Re: [asterisk-users] Hang up phone after declined attended transfer

[Ryan Wagoner]

On Tue, Jan 10, 2012 at 1:57 PM, Ryan Wagoner rswago...@gmail.com wrote:

 On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote:

 We have a customer who has asked us to change this behavior, but I
 haven't been able to find a way to do it.  Server is Asterisk 1.6 and the
 phones are SPA 303 and 504.

 Receptionist gets an outside call, starts an attended transfer
 The office person being called answers by pressing the speaker button
 Declines to take the call
 Receptionist goes back to the original outside call by pressing the line
 button
 The office phone goes to hold instead of hanging up

 If the receptionist hangs presses the hookswitch instead of the line
 button, then it does hang up the call to the internal office phone, however
 that phone then goes into reorder tone.

 --
 Carlos Alvarez
 TelEvolve
 602-889-3003


 When the receptionist presses the hookswitch it should hang up the remote
 internal phone. Playing the reorder tone is due to a setting on the SPA
 phone. I had to change this for a client that used the SPA phones and I'm
 drawing a blank as to which setting.

 Ryan


I did a quick search and found the setting. Go to the Regional tab and find
the Reorder Delay. Change that to 255, which will disable the order tone
and cause the phone to hangup.

Ryan
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Re: [asterisk-users] Best non polycom SIP conference room phone

[Ryan Wagoner]

On Sun, Jan 8, 2012 at 12:03 PM, brya...@zktech.com wrote:

 Thank you for your responses. No where did I say I hate polycom phones. I
 personally do not like their approach to sip as a company. Their audio
 quality  is top notch but for me the rest leaves me wanting. Has anyone
 used the newer snom conference room phone?


If the Snom conference phone is anything like their deskphone speakerphone
I would stay away. We purchased Snom 360s for the large number of BLF and
VPN capability. However I quickly had complaints about the speakerphone.
Additionally the user interface was laggy. I've tried changing settings and
they still sound like a non duplex speakerphone. I only have a few Snom
phones left and everything else is Polycom. You can't beat their sound
quality and the user interface is responsive. If you keep an eye out on the
clearance deals at telephonydepot.com you can sometimes grab a Polycom
speakerphone for a great price.

Ryan
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Re: [asterisk-users] Blind transfers being cancelled by asterisk hanging up on remote caller

[Ryan Wagoner]

On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg l...@solvent-llc.com wrote:

 Doug:
 for what it's worth I am having the exact same nightmare.  Not sure exactly
 when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I
 am
 running 1.8.9rc1).  I also have Polycom (335, 550, 650)  and blind
 transfers
 are broken.  All legs of the call are dropped when the xfer is executed.  A
 calls B, B xfer to C and (C) blips for a split second like its ringing but
 then all calls go dead.  I tried to debug myself using some sip tracing but
 I didn't get very far.  I even tried mucking around with a few settings in
 my Polycom provisioning I thought might be related e.g.

  voIpProt.SIP.allowTransferOnProceeding
  voIpProt.SIP.connectionReuse.useAlias
  voIpProt.SIP.useContactInReferTo
  voIpProt.SIP.conference.parallelRefer
  voIpProt.SIP.strictLineSeize
  voIpProt.SIP.strictUserValidation
  voIpProt.SIP.strictReplacesHeader
  voIpProt.SIP.useContactInReferTo

 and also upgraded to the new 3.3.4 firmware which is out yesterday,  didn't
 change a thing.
 stuck here for now,  Attended xfers seem to work.I am not sure this is
 a
 Polycom-specific issue because I was seeing this bad behavior even using
 some Softphones I set up for testing.

 my next recourse is to try rolling back to 1.8.8.0 or earlier and if that
 fixes it then I will open a JIRA ticket with more details.

 Luke


 --
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas
 Mortensen
 Sent: Thursday, January 05, 2012 3:04 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Blind transfers being cancelled by asterisk 
 hanging up on remote caller

 Hello all,

 I have a system running AsteriskNOW with asterisk
 asterisk-1.8.8.1-1_centos5
 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so
 that
 blindpreferred=1 (all transfers default as blind transfers). If a customer
 calls in  we answer  transfer, everything works fine. But if we call out
 to a customer  then transfer to another internal extension, that extension
 quickly rings  then the call is immediately gone  hung up. We are using
 Polycom firmware 3.3.3.

 In troubleshooting this  analyzing the asterisk logs ( asterisk SIP
 debug), I am seeing a few interesting items. Any help would be appreciated.

 [...]

 Thanks,
 -
 Doug Mortensen


I can't reproduce this on a test system with Asterisk 1.8.8.1 using a
Polycom 335 and 550 running firmware 3.2.6. I called an external number
using Vitelity then blind transferred to the other phone. I am interested
as I have a production system with Polycom 335 phones running 1.8.7.0 that
works.

Ryan
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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[Ryan Wagoner]

On Fri, Jan 6, 2012 at 10:01 AM, Benny Amorsen benny+use...@amorsen.dkwrote:

 David Backeberg dbackeb...@gmail.com writes:

  Thanks for clearing that up. I was getting all excited that I could
  flash the PAP2T; I've always used regular voice tones over SIP with
  the PAP2Ts.

 SPA-2102 supports T.38. If you ignore the WAN-port, it is practically a
 PAP2T. The only time you cannot ignore the WAN-port is when doing
 provisioning.


I ignore the LAN (yellow) port and only use the WAN (blue) port. The LAN
port has a DHCP server enabled by default. The WAN port is setup with a
DHCP client. The good thing about using the WAN port is if the settings are
cleared you won't have an unknown DHCP server on your network.

I plug a phone in and dial the below. Then you can configure the rest from
the web interface.

 to enter admin ivr
7932 enter 1 to enable web interface
110 will read the WAN address

Ryan
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Re: [asterisk-users] 1.6 and 1.8

[Ryan Wagoner]

On Thu, Dec 29, 2011 at 12:05 AM, Bruce B bruceb...@gmail.com wrote:

 I have been running 1.8.7 with a few fixes back ported from the 1.8.8
 release candidate for the last 2.5 months. The system processes around
 4,000 calls per day over PRIs for 250 Polycom phones.

 Previously I was running 1.6.1.18 with a bunch of back ports for fixes
 and features. Overall it was stable but every few months I had an issue
 where a channel would get hung. When this happened core show channels would
 crash the console and I would eventually have to restart Asterisk.

 Ryan


 What od you mean by, been running 1.8.7 with a few fixes back ported
 from the 1.8.8 release candidate. So, this is a version 1.8.7 release that
 you are using or a 1.8.8 or is this a mix of both that you come up with?
 Can you please be specific with fixes?

 Thanks


It was a mix I came up with as I was hitting a few bugs in 1.8.7 and 1.8.8
wasn't released. At this point I would just go for 1.8.8. The issue was
mainly 17541 which was filling my logs and basically made Asterisk unusable.

https://issues.asterisk.org/jira/browse/ASTERISK-17541
https://issues.asterisk.org/jira/browse/ASTERISK-18570
https://issues.asterisk.org/jira/browse/ASTERISK-18101

I had tested 1.8.4 before and was hit by a bunch of dtmf issues that were
fixed in 1.8.5. When 1.8.7 came out it looked fairly stable so I switched
from 1.6.1.18. I was running the 1.6.1 branch as I needed TCP SIP support.
Right now I have been testing 1.8.8 which looks to be a good release. The
1.8 series has come a long way in a few releases as far as fixing major
bugs.

Ryan
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Re: [asterisk-users] 1.6 and 1.8

[Ryan Wagoner]

On Thu, Dec 29, 2011 at 12:18 PM, Bruce B bruceb...@gmail.com wrote:

 Log are being filled with g729 transcoding error in 1.8.7x now :-(
 I don't dare to test 1.8.8x as it might have something else broken.
 Unfortunately I can no longer trust the release candidates. Thanks for the
 input.


What are you using for transcoding? I'm running 1.8.7 with a Sangoma
transcoding card. I would give 1.8.8 a try as they fixed the transcoding
issue in 1.8.7 or at least try the patch I mentioned before.

Ryan
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Re: [asterisk-users] 1.6 and 1.8

[Ryan Wagoner]

On Wed, Dec 28, 2011 at 4:33 PM, Danny Nicholas da...@debsinc.com wrote:

 I understand the end of life issue.  What I fail to understand is that if
 1.8 is the Cadillac of Asterisk, why did they make 10.0 and why does 1.8
 have so many bugs (just what I read here, not from my actual experience)?


Once released a version will only have bug and security fixes. New features
go into trunk to be included in the next version. Asterisk has long term
support releases like 1.4 and 1.8 and standard releases like 1.6 and 10.
This model is no different than other software like Ubuntu.

Even though a series only has bug and security fixes I have found
regressions occur between point releases. Just make sure to test thoroughly
before putting a system in production. I tend to stick with a version until
I need the features in a newer version or back porting a security fix
becomes overly involved.

I have been running 1.8.7 with a few fixes back ported from the 1.8.8
release candidate for the last 2.5 months. The system processes around
4,000 calls per day over PRIs for 250 Polycom phones.

Previously I was running 1.6.1.18 with a bunch of back ports for fixes and
features. Overall it was stable but every few months I had an issue where a
channel would get hung. When this happened core show channels would crash
the console and I would eventually have to restart Asterisk.

Ryan
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Re: [asterisk-users] Codec warnings after upgrade to 1.8

[Ryan Wagoner]

On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling ewiel...@nyigc.com wrote:

 I'm getting various codec related warnings after upgrading to 1.8.  Did I
 miss something in the UPGRADE file?  Does Asterisk no longer transcode 8-)?

 WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel
 DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native
 formats 0x4 (ulaw)

 And

 WARNING[11120]: channel.c:4909 ast_write: Codec mismatch on channel
 SIP/interglobe-sip-01e6 setting write format to g722 from ulaw native
 formats 0x4 (ulaw)


When I upgraded to 1.8.7 I had a bunch of those warnings. I had backported
the below patch which was included in a 1.8.8 release candidate. Since
1.8.8 has been released I would just upgrade to that.

https://issues.asterisk.org/jira/browse/ASTERISK-17541

Ryan
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Re: [asterisk-users] CDR mysql with asterisk 1.4

[Ryan Wagoner]

On Mon, Nov 21, 2011 at 7:13 AM, salaheddine elharit
salah.elharit...@gmail.com wrote:
 hello list



 i have asterisk 1.4 installed and i want to use CDR mysql  during the
 installation i didn’t check the cdr mysql with  make menuselect



 my question : i want to check this option now  after the installtion and
 configuration of all options but he asks me to do. /configure before to use
 make menuselect



 i want to know if there any problem if i do. / configure and make menuselect
 to install cdr because this server is very important for me and i can’t stop
 it


How did you initially install Asterisk? When compiling from source
./configure is the first step before you can run make. It shouldn't
prompt to run ./configure for make menuselect if you are just changing
some options from a previously compile and install.

If you were able to run make menuselect without configure you might be
able to load the module while Asterisk is running. You would copy the
cdr_mysql.so to the lib directory and run module load cdr_mysql.
However I would still plan this for after hours in case of an issue.

Ryan

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Re: [asterisk-users] Polycom Attended Transfer

[Ryan Wagoner]

On Wed, Nov 16, 2011 at 1:49 PM, eherr email.eherr9...@gmail.com wrote:
 When you perform an attended transfer, the extension of the person
 transferring is displayed to the co-worker.



 Can I override the caller ID to display the caller’s callerID during a blind
 transfer?



 Thanks,

 --E


The point of an attended transfer is to announce the calling party.
When you hit transfer on the Polycom you have the option to select
Blind on the screen. A blind transfer will use the caller id of the
incoming call, not the person making the transfer.

Ryan

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Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

[Ryan Wagoner]

On Sat, Oct 8, 2011 at 10:41 AM, Luke Hamburg l...@solvent-llc.com wrote:
 Interesting.  I just signed up with Gafachi (haven't even tested the service
 yet) but I planned to make use of their T38 support since they are listed at
 voip-info as being one of the ITSP's that _do_ support T38.  Have you tried
 contacting Gafachi with these results about their broken implementation?  I
 would hope/expect them to try to fix this, instead of trying to force
 Asterisk to violate RFCs.

It sounds like that Gafachi's T38 implementation
is horribly, horribly broken I'm not tied to them
at all, so if their stuff is broken, I'll go
somewhere else.


I signed up with Gafachi a few weeks ago to use them for T38 as well.
I haven't had any luck getting it to work. I have been mainly trying
to use Asterisk in T38 pass through mode and have tested with a
Linksys SPA2102 and Zoiper. Gafachi basically told me they have many
customers utilizing their T38 implementation and that it works. When
asked for a list of compatible devices they said there were too many
combinations and it was up to me to find a working solution.

I am still looking a PAYG service provider that has a working T38
implementation. It seems like these are impossible to find.

Ryan

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Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

[Ryan Wagoner]

On Sat, Oct 8, 2011 at 3:51 PM, James Sharp ja...@fivecats.org wrote:
 On 10/08/2011 02:38 PM, Ryan Wagoner wrote:

 I signed up with Gafachi a few weeks ago to use them for T38 as well.
 I haven't had any luck getting it to work. I have been mainly trying
 to use Asterisk in T38 pass through mode and have tested with a
 Linksys SPA2102 and Zoiper. Gafachi basically told me they have many
 customers utilizing their T38 implementation and that it works. When
 asked for a list of compatible devices they said there were too many
 combinations and it was up to me to find a working solution.

 I wonder how many of these customers are just getting fallback to G711 when
 the T38 stack falls over.  Heck, I thought I was getting T38 until I
 realized that I had SendFAX running with the audio fallback option. Turned
 that off, and fax fails 100% of the time.

 I am still looking a PAYG service provider that has a working T38
 implementation. It seems like these are impossible to find.

 I found t38faxing.com.  I was going to try them until I saw that their
 opening credit is $10.  More than I want to spend to try for just home
 faxing.


I tried to sign-up with them a week ago, but received an error
message. I went to their contact page and saw the grnvoip.com email.
It turns out grnvoip and t38faxing are both owned by ez call service.
I signed up for grnvoip.com, but was unable to get the t.38 faxing to
work. Additionally ez call service's administration panel is not laid
out the best and doesn't let you change the static IPs that are
allowed to send calls to them.

I have tested T38 faxing and pass through with Asterisk 1.8 and
combinations of the Linksys SPA2102 ATA, Zoiper, and Asterisk. The
faxes are sent and received successfully. Analyzing the packet traces
with Wireshark shows they were sent with T38. I just need to find a
provider that has a working T38 implementation.

Ryan

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Re: [asterisk-users] OT - Polycom - Which provisioning protocol to choose ?

[Ryan Wagoner]

On Wed, Jun 29, 2011 at 8:34 AM, Olivier oza_4...@yahoo.fr wrote:


 2011/6/29 Ruben Rögels ruben.roeg...@jumping-frog.org

 Personally I would use HTTP too.

 Simple reason: You are much more flexible with it and a in most
 scnearios you have a webserver running anyway.

 I build some PHP-Script to provision SNOM VoIP phones for mass
 deployment and it works like a charm.

 What is specific to Polycom phones is that they upload several files (log
 files, config files) which not easy to handle for casual TFTP server.


 Regards,
 Ruben

You can turn on WebDav support and the Polycom phones can upload the
logs to HTTP. I'm using HTTP with a 250 phones and a php script to
configure them off my FreePBX database. Works great!

Ryan

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Re: [asterisk-users] VMX Locator

[Ryan Wagoner]

On Thu, Jun 23, 2011 at 7:45 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
 Hello All,

 I've been doing some looking into VMX Locator(part of FreePBX from what I
 see). One of my sales guys came from a company that was running FreePBX and
 we are running straight asterisk installed using custom built RPM's.
 Currently in the voicemail app the only key press that does anything is *,
 which kicks the person out into their own voicemail at the moment.

 However, VMX Locator gives options for pressing 0, 1 and 2 and have
 different stuff happen based on those. My question is has anyone actually
 tried or gotten this to work in Asterisk itself? I've been looking it up but
 no luck so far. Thanks.
 --

You can install FreePBX on a VM, etc and see the dialplan it generates
for vmx. It looks like they are emulating the first part of the
Asterisk voicemail system to give the menu choices.

Ryan

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Re: [asterisk-users] Connected Line ID

[Ryan Wagoner]

On Mon, Jun 20, 2011 at 5:39 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
 Oke,

 But is there a patch from version 1.6.2.12?

 Greeting,

 Arjan

 -Oorspronkelijk bericht-
 Van: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle
 Verzonden: 20-06-2011 11:36
 Aan: Asterisk Users Mailing List - Non-Commercial Discussion
 Onderwerp: Re: [asterisk-users] Connected Line ID

 Arjan Kroon | Mobillion wrote:
 And if I can see it, this patch is already included in version 1.6.2.12.  Or 
 am I wrong?

 That I can't answer.  I'm still using 1.4.x and am experimenting with
 1.8.x.  I recall reading that it wasn't supported directly until 1.8
 without patches.

 Doug


I am using 1.8 now, but I had updated the patch for SIPCalledRPID()
for 1.6.2 and was using it successfully.

http://pastebin.com/K1mmGU1c

Ryan

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Re: [asterisk-users] Cisco registration problem with 1.8.3.3

[Ryan Wagoner]

On Sun, May 29, 2011 at 3:18 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
 And f/w POS3-07-4-00

That is strange that Asterisk is not sending anything back in response
to the register. Have you looked at the Asterisk console or logs to
see why it is rejecting the register. You might have to enable debug
mode

core set debug 5
sip set debug on

Also if you want to see debug output on the screen check that the
following is uncommented in /etc/asterisk/logger.conf

console = notice,warning,error,debug

Is it possible for you to try a later firmware version? Although 7.4
looks to be a good version according to others notes.

http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx

Ryan

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Re: [asterisk-users] Cisco registration problem with 1.8.3.3

[Ryan Wagoner]

On Mon, May 30, 2011 at 2:45 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
 Console is showing the following. Looks like it doesn't like the format of the
 REGISTER message???

 --- SIP read from UDP:192.168.1.114:5060 ---
 REGISTER sip:192.168.1.41 SIP/2.0
 Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK2e11eaa2
 From: sip:702@192.168.1.41;user=phone
 To: sip:702@192.168.1.41;user=phone
 Call-ID: 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114
 CSeq: 101 REGISTER
 User-Agent: CSCO/7
 Contact: sip:702@192.168.1.114:5060
 Content-Length: 0
 Expires: 120


 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7539 find_call: = Looking for
 Call ID: 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 (Checking From)
 --From tag  --To-tag
 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7543 find_call: REGISTER request
 has no from tag, dropping callid:
 00078599-2e4e0002-23aa7a4e-0b32ceef@192.168.1.114 from:
 sip:702@192.168.1.41;user=phone
 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:24110 handle_request_do: Invalid
 SIP message - rejected , no callid, len 337

The log states find_call: REGISTER request has no from tag, dropping
callid. If you look at the From: line, it should end with
;tag=SOMEVALUE. Looking at sip.conf you could set pedantic=no and the
phone should register. The best solution would be to upgrade the phone
firmware. I know 8.12 works.

Ryan

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Re: [asterisk-users] Cisco registration problem with 1.8.3.3

[Ryan Wagoner]

On Mon, May 30, 2011 at 5:18 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
 Many thanks for that.

 I tried pedantic=no (adding it directly to the [702] section in
 sip_additional.conf: I'm using the freepbx frontend and it doesn't seem to
 have a way to enter that through the gui), but it didn't fix it: same console
 log.

The setting is a global setting. With FreePBX you want to add
pedantic=no to /etc/asterisk/sip_general_custom.conf You can verify
from the Asterisk console with sip show settings. You should see
Pedantic SIP support: No under Global Signalling Settings

Ryan

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Re: [asterisk-users] Cisco registration problem with 1.8.3.3

[Ryan Wagoner]

On Sat, May 28, 2011 at 4:08 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
 I am having a problem registering my cisco phones which is exactly like that
 described in

 http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html

 except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00

 The symptoms are:

 o 7960 lines show [X]
 o Outbound calls can be made from the phone, including call pickup of inbound
 calls, but not to it.
 o Trace shows REGISTER packets sent from phone but no response from Asterisk

 Is there any way this regressed code could be picked up in a 1833 build or
 have I got another problem?

I'm able to register a 7940 against Asterisk 1.8.4.1. You might try
out that version as it has the fix for registering Cisco phones.
However I thought the bug was introduced in 1.8.4 and not 1.8.3.3.

I know in the past when I had issues registering Cisco phones I had to
make sure the nat settings matched. If you set nat=yes in the sip.conf
you must set nat_enable: 1 in SIPDefault.cnf for the phone. What I
noticed was when nat=yes is set in Asterisk it ignores the rport and
always sends the reply on the port used for the request. Cisco will
ignore this reply and not register.

Ryan

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Re: [asterisk-users] Cisco registration problem with 1.8.3.3

[Ryan Wagoner]

On Sat, May 28, 2011 at 5:18 PM, Ian S. Worthington
ianworthing...@usa.net wrote:
 I too had heard that 1833 did NOT have the 184 problem, which makes me
 suspicious that it's not that.

 I don't think its a NAT problem.  Neither a sip trace not tcpdump show any
 response at all to the incoming REGISTER.

 The phone is on the local lan.  I have nat=no and nat_enable: 0


You are running tcpdump on the Asterisk server? Are you capturing all
traffic or only certain ports? What firmware are you running on the
phone? I am using PS03-8-12-00. It wouldn't hurt to try with nat
enabled, see below. I setup all my phones this way as it saves having
to reconfigure when users take them home.

sip.conf
nat=yes

SIPDefault.cnf
nat_enable: 1
nat_address: 
nat_received_processing: 1

Ryan

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Re: [asterisk-users] how to know how many calls are on hold

[Ryan Wagoner]

On Tue, May 17, 2011 at 10:16 AM, virendra bhati virbh...@gmail.com wrote:
 hi list,

 please help me how to know how many calls are on hold.


If they are SIP channels you can use: sip show inuse The last column
are calls on hold.

Ryan

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Re: [asterisk-users] Polycom IP335 3.3.1 Call Waiting

[Ryan Wagoner]

On Thu, May 19, 2011 at 1:24 PM, Ryan Wagoner rswago...@gmail.com wrote:
 I updated my phones to the UCS 3.3.1 firmware a few months back. The
 scenario is I place a call and receive an incoming call. With 3.3.1
 the screen will show call 1/2 and I have to press the down arrow to
 see the caller name / number. Has anybody else noticed this with
 3.3.1? I had thought with 3.2.4 it would automatically show call
 waiting name and number without pressing any keys. It could be
 possible I missed a setting, but I didn't see anything in the admin
 guide.

 Ryan

For those wondering it appears to be a bug in 3.2.5 and later
versions. I downgraded to 3.2.4 and the caller id for the incoming
call waiting call will show for 10 seconds as described in the Polycom
user guide. This only effects those with IP33x model phones.

Ryan

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[asterisk-users] Polycom IP335 3.3.1 Call Waiting

[Ryan Wagoner]

I updated my phones to the UCS 3.3.1 firmware a few months back. The
scenario is I place a call and receive an incoming call. With 3.3.1
the screen will show call 1/2 and I have to press the down arrow to
see the caller name / number. Has anybody else noticed this with
3.3.1? I had thought with 3.2.4 it would automatically show call
waiting name and number without pressing any keys. It could be
possible I missed a setting, but I didn't see anything in the admin
guide.

Ryan

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Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription

[Ryan Wagoner]

On Fri, May 13, 2011 at 2:58 PM, Skyler skchopper...@gmail.com wrote:
 Hi all,



  Anyone know how to make asterisk properly reply to  options keep-alive? Or
 just force a 200 OK somehow?



  I recently took over a server and they have ~80 pap2 devices that send nat
 keep-alive and * always replies with 481 No subscription. It’s more of an
 annoyance, I know but I like to keep my pcap’s clean.


Which version of Asterisk? 1.8 should have this built-in. I made a
patch for 1.6.2 which you can download at http://pastebin.com/Ls3m8t15

Ryan

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Re: [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?

[Ryan Wagoner]

On Fri, May 6, 2011 at 2:52 PM, Andrew Latham lath...@gmail.com wrote:
 On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian va...@arminco.com wrote:
 Has anyone used this board as an Asterisk server?
 http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y

 I'm mostly interested about the possible compatibility issues this board may
 have with the AEX800 card.

 Yes that is a great system and the built-in IPMI is a livesaver...  if
 you are using a full size harddrive you need to apply some protection
 to the card in the case (the superserver 1U).  They are close but not
 touching...

 --
 ~~~ Andrew lathama Latham lath...@gmail.com ~~~

I have the X7SPE-HF-D525, 1U SC503-200B case and SSD for my firewall.
Just keep in mind the 1U case with no fans is like an oven. In a 75F
room the system temp was 132F and the CPU was 163F. This is within
operating limits of the Atom platform. However I'm not sure I would
want a hard drive and telco card in there as well. I ended up putting
a 40mm rated for 7cfm of airflow fan in the case. The temps dropped
dramatically to 120F system and 131F.

Ryan

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Re: [asterisk-users] Asterisk, SIP Firewalls

[Ryan Wagoner]

On Wed, Apr 27, 2011 at 1:16 PM, Myles Wakeham my...@techsol.org wrote:
 It kinda scares me though.  I know that SIP is an attractive attack-vector,
 and that there are scripts out there that target SIP devices.  I know I
 could run Fail2Ban on the server, which is fine (we're doing that anyway
 now), but before I go down this path, I wanted to get general feedback if we
 are using our Asterisk system using 'best practices' or whether it should
 never be sitting behind a Firewall, despite the fact that it is working
 pretty close to perfect as it is right now.  I just want to find a way to
 reduce the latency.

I have placed Asterisk outside the firewall / nat router to avoid the
translation. I usually will setup the server with dual NICs. One has
the public IP and another has the internal private IP. Set the default
gateway to the public IP gateway. Then just configure iptables to
firewall the server interfaces accordingly. This configuration allows
Asterisk to sit directly on the Internet while keeping your internal
phones from going out your nat router and back to Asterisk. Basically
the best of both worlds.

Ryan

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Re: [asterisk-users] Warm Transfer in Asterisk

[Ryan Wagoner]

On Sat, Apr 23, 2011 at 8:56 AM, Jeff Johnson
jjohn...@neturallyspeaking.com wrote:
 Is there a way do what is sometimes called a 3rd party transfer in
 Asterisk.  That is; Call A comes in and is answered B.  B then places A on
 hold and calls C.  After C answers, BC chat for a moment, then B brings A
 on line.  After making intro’s B then drops off call.


Yes it is called an attended transfer. You can use the atxfr feature
code or most phones will have transfer capability built in. On Polycom
phones the transfer button defaults to attended transfer. There is a
separate blind transfer button as well.

Ryan

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Re: [asterisk-users] 1.8.4-rc2: ReceiveFAX fails

[Ryan Wagoner]

On Fri, Apr 15, 2011 at 7:00 PM, sean darcy seandar...@gmail.com wrote:
 Using spandsp-0.0.6-pre18, the Jan 22 release.


You might try using spandsp-0.0.6-pre17. That version works great for
me with 1.8.4-rc2. When I tried pre18 it failed to receive any faxes.

Ryan

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Re: [asterisk-users] Asterisk PRI back-to-back connect

[Ryan Wagoner]

On Tue, Mar 22, 2011 at 12:53 PM, satish patel satish...@hotmail.com wrote:
 Hey Guys!

 We have two Asterisk with A102D Sangoma cards now i want to connect them
 back-to-back over PRI line via Cross-cable so what would be the
 configuration specially timing source and all? anybody did it before like
 this ?

 I want to make sure everything before putting in production.. (saving my
 downtime)

 -S


If is no different then setting up the card to connect with a telco.
One Asterisk box will be the net and the other is cpe. You can use
whatever protocol national, 5ess, etc you like. Any reason not to join
the boxes via SIP?

Ryan

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Re: [asterisk-users] Need to buy the Digium card, to confirm

[Ryan Wagoner]

On Sat, Feb 26, 2011 at 5:33 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Hi All;

 My server and its slots written in it the following so I need to know which 
 card to order it (I need a card supporting 2 E1s):

 PCIE_G2_X4
 PCIE_G2_X8

 Actually I do not know what is meaning by G2.

 OK I tried to buy directly from the below link but I found it is mentioned 
 that it is x1 and not x4 or x8 so how can I get x4 or x8?

 The link:

 http://store.digium.com/productview.php?product_code=TE220B

 Description for the product:
 Two (2) span digital T1/E1/J1/PRI PCI-Express x1 card

 So please advise what do to?
 Regards
 Bilal

You can place 1x card in a 4x or 8x slot. The same goes for placing a
4x card in an 8x slot.

Ryan

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Re: [asterisk-users] Paging with Polycom 3.3.x

[Ryan Wagoner]

On Thu, Feb 24, 2011 at 1:41 PM, Mike l...@net-wall.com wrote:
 Hi,



 My phones stopped auto-answering when being paged, since I moved on to
 Polycom firmware 3.3.0 (3.3.1 is the same, I tried).  That is with Asterisk
 1.6.2.16.



 I looked at the wiki but nothing I try there works, even if I cut and paste
 the same setup.



 Any one has any idea of what I should change from my old 3.2.3 setup?  My
 older phone (501) still using 3.1.6 still auto-answer correctly.


Polycom changed some of the config file options as outlined in the UC
Software upgrade guide. I am using the following for paging.

voIpProt.SIP
  voIpProt.SIP.alertInfo
voIpProt.SIP.alertInfo.3.class=ringAutoAnswer
voIpProt.SIP.alertInfo.3.value=Ring Answer
  /voIpProt.SIP.alertInfo
/voIpProt.SIP 
  /voIpProt

Ryan

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Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?

[Ryan Wagoner]

On Sun, Feb 20, 2011 at 9:11 AM, Ken D'Ambrosio k...@jots.org wrote:
 On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote:
 On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote:

 Hi, all.  I've finally made the jump from 1.4 to 1.8.  I've installed
 everything (I think), my Sangoma card initializes right... but there's
 no dahdi command -- not from the base, nor as a subset of the core
 commands.  I've got my channels configured in my chan_dahdi.conf file.
 What am I missing, here?

 What version of dahdi do you have installed? I would try using the
 latest version 2.4.0. It is important to compile and install in the
 correct order. I usually do dahdi, libpri, asterisk, and then wanpipe.

 I'm running the latest of everything, except my kernel -- I went with
 2.6.32.27 as being a well-maintained long-term kernel.  (2.6.37 gave me
 grief -- too new, I guess.)  I'm running -- if it makes a difference -- on
 an Ubuntu 8.04-4 system.  I've re-installed everything, in the order you
 gave, to, alas, the exact same result: everything seems to initialize,
 install, etc., correctly, but no dahdi feature in Asterisk.  Is there a
 module I need to load?  Or... something?  I'd hate to have to revert to
 1.4 after all this work.

 Thanks!

 -Ken

If you have autoload=yes in modules.conf it should load automatically.
Have you checked log, usually /etc/asterisk/full to see if you are
getting any error messages relating to dahdi?

Ryan

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Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?

[Ryan Wagoner]

On Sun, Feb 20, 2011 at 9:44 AM, Ryan Wagoner rswago...@gmail.com wrote:
 On Sun, Feb 20, 2011 at 9:11 AM, Ken D'Ambrosio k...@jots.org wrote:
 On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote:
 On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote:

 Hi, all.  I've finally made the jump from 1.4 to 1.8.  I've installed
 everything (I think), my Sangoma card initializes right... but there's
 no dahdi command -- not from the base, nor as a subset of the core
 commands.  I've got my channels configured in my chan_dahdi.conf file.
 What am I missing, here?

 What version of dahdi do you have installed? I would try using the
 latest version 2.4.0. It is important to compile and install in the
 correct order. I usually do dahdi, libpri, asterisk, and then wanpipe.

 I'm running the latest of everything, except my kernel -- I went with
 2.6.32.27 as being a well-maintained long-term kernel.  (2.6.37 gave me
 grief -- too new, I guess.)  I'm running -- if it makes a difference -- on
 an Ubuntu 8.04-4 system.  I've re-installed everything, in the order you
 gave, to, alas, the exact same result: everything seems to initialize,
 install, etc., correctly, but no dahdi feature in Asterisk.  Is there a
 module I need to load?  Or... something?  I'd hate to have to revert to
 1.4 after all this work.

 Thanks!

 -Ken

 If you have autoload=yes in modules.conf it should load automatically.
 Have you checked log, usually /etc/asterisk/full to see if you are
 getting any error messages relating to dahdi?

 Ryan


You might also do a rm -rf /usr/lib/asterisk/modules/*.so and make
install Asterisk again. You could have some modules left around from a
previous version conflicting with things.

Ryan

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Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?

[Ryan Wagoner]

On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote:
 Hi, all.  I've finally made the jump from 1.4 to 1.8.  I've installed
 everything (I think), my Sangoma card initializes right... but there's no
 dahdi command -- not from the base, nor as a subset of the core
 commands.  I've got my channels configured in my chan_dahdi.conf file.
 What am I missing, here?

 Thanks...

 -Ken


What version of dahdi do you have installed? I would try using the
latest version 2.4.0. It is important to compile and install in the
correct order. I usually do dahdi, libpri, asterisk, and then wanpipe.

Ryan

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Re: [asterisk-users] Polycom IP335

[Ryan Wagoner]

On Thu, Feb 17, 2011 at 12:02 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
 No parameters were rejected. Maybe my perception of backlight off is
 incorrect. When it is off I expect it so be similar to a Cisco 7961. So no
 light whatsoever and very hard to read in dim light. Yet in the Idle state
 the screen of the IP670, to me, still looks like it is still lit and I can
 clearly read anything that's on the screen. Made pics of backlight off in
 idle state and on. Am I missing something?

 http://www.xs4all.nl/~pjl/tmp/IP670_backlight_off.jpg
 http://www.xs4all.nl/~pjl/tmp/IP670_backlight_on.jpg

 Regards.
 Patrick


The color screen must be different or it is a firmware bug. Was it any
different on 3.2.x vs 3.3.x? On my IP550 you can still read the screen
with the backlight off, however the screen background is dark. At
night with the backlight off the display doesn't light up the room and
is hard to read. The backlight has a whitish color when it is on.

Ryan

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Re: [asterisk-users] Polycom IP335

[Ryan Wagoner]

On Wed, Feb 16, 2011 at 2:51 PM, ERIC HERRON e...@lanline.com wrote:
 I am posting here since you guys are my last hope.



 I am trying to configure a Polycom Soundpoint IP 335 with MWI.

 Is there any way to eliminate the scrolling messages and Msgs softkey?

 I am trying to get it where it’s just the light that indicates the new
 messages.

 I don’t know if Asterisk has to send a different notification or what have
 you.

 Thanks,

 --Eric

I've had that same request a few times. I've looked through the
Polycom manual, even the new UC software 3.3.1, and never found the
setting for it. It is either all or nothing for MWI. The scrolling
messages is the part I get complaints about. People would rather have
the clock shown on the screen.

Ryan

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Re: [asterisk-users] Polycom IP335

[Ryan Wagoner]

On Wed, Feb 16, 2011 at 3:05 PM, ERIC HERRON e...@lanline.com wrote:
 I have it on the 430s.



 I think it’s a firmware issue but I am having trouble replicating it on the
 430



 I could have sworn I had it on one phone before I rebooted it but memory
 might be influenced by hopes.



What setting were you using to configure it that way. I've was running
3.2.3 and am now using 3.3.1 on the IP335s and never had luck
disabling the scrolling message.

Ryan

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Re: [asterisk-users] Polycom IP335

[Ryan Wagoner]

On Wed, Feb 16, 2011 at 5:49 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
 I share your pain. I have an IP335 and IP670 here. Have not configured the
 IP335 yet but using the latest Admin Guide (3.3.1) did configure the IP670
 running the latest bootrom (4.3.0) and firmware (3.3.1). Problems on the
 IP670:

 1) it seems impossible to turn off the backlight

 2) it seems impossible to disable that stupid periodic MWI sound.
   Whoever at Polycom thought that that was a good idea should meet a
   seriously big clue-by-4.

 To me it seems like their 3.3.x branch could use a few bugfixes...

 Have you tried an older or newer release?

 Regards,
 Patrick

Backlight works fine on a IP550 with 3.3.1 . I have mine set to off
when idle. I like that the 3.3.x series doesn't required the default
sip.cfg and phone1.cfg files. The structure of the XML seems cleaner
and more consistent.

  up
up.backlight up.backlight.idleIntensity=0 up.backlight.onIntensity=3
/up.backlight
  /up

The only bug I've seen with 3.3.1 is on the IP335. After dialing when
it connects the caller name and number jump 1 pixel higher, which
looks weird as it is close to the line. One 3.2.3 it didn't move up
and looked centered. However the scrolling caller id for incoming
calls make this minor annoyance worth the upgrade.

Ryan

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Re: [asterisk-users] Polycom IP335

[Ryan Wagoner]

On Wed, Feb 16, 2011 at 8:38 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
 On 02/17/2011 12:10 AM, Ryan Wagoner wrote:

   up
     up.backlight up.backlight.idleIntensity=0
 up.backlight.onIntensity=3
     /up.backlight
   /up

 Here's what I have:

 up
 up.idleTimeout=10
 up.backlight.idleIntensity=0
 up.backlight.onIntensity=3
 /

 That's obviously using a different way (is syntax the proper word?). Don't
 know if that could make a difference. The config does work except for this
 setting and the MWI chirp.

Your config looks fine to me. For the 3.3.x series they changed how
the xml was grouped. For a setting like x.y.z it used to just be x
x.y.z=value / now it is xx.y x.y.z=value/x.y/x. From what
I have noticed the phone only cares about the x.y.z=value and not
which section it is under. My 3.2.x config file worked except for
alert info, ringer, and feature settings, which was outlined in
Simplified_Configuration_Improvements_in_UC_Software3_3_0_TB60519.pdf

Either way check the log the phone uploads on the provisioning server.
It will tell you which parameters were rejected. You can also find the
number of parameters accepted in rejected in the phone's menu.

Ryan

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Re: [asterisk-users] Moving asterisk from one network to another.

[Ryan Wagoner]

On Fri, Dec 24, 2010 at 7:40 AM, Jim Dickenson dicken...@cfmc.com wrote:
 If you set bindaddr in any conf file you will need to change the IP address
 there.
 --
 Jim Dickenson
 mailto:dicken...@cfmc.com
 CfMC
 http://www.cfmc.com/

You will also need to change externip and localnet if those are set in sip.conf.

Ryan

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Re: [asterisk-users] Why does sip show peers show my router/gateway address as the client IP address?

[Ryan Wagoner]

On Sat, Dec 11, 2010 at 3:06 AM, Bruce B bruceb...@gmail.com wrote:
 Hi Everyone,
 I am using pfSense to do firewall and NAT on an Asterisk server. I have
 ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP
 192.168.5.5. However, when a user from outside using Linksys WRP400 ata
 connects to the Asterisk server and registers I see them as 192.168.1.1 in
 the sip show peers command. In face, all many different of the Linksys
 WRP400 show the same. It seems that pfsense does something to the packets
 that when they reach Asterisk it thinks they are sent from the Gateway
 rather than the actual endpoint hence the calls are not reaching the other
 side but registration is made.
 Any experience with this?
 Thanks

Do you have the siproxd package installed on pfsense? It is suspossed
to handle registrations from multiple phones behind NAT. In your case
since the phones are external I would probably remove it if installed.
I haven't needed siproxd.

Also on Asterisk set externip to your static IP in sip.conf. Or if you
don't have a static IP set externhost. You also need to configure
localnet.

Ryan

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Re: [asterisk-users] Why does sip show peers show my router/gateway address as the client IP address?

[Ryan Wagoner]

On Sat, Dec 11, 2010 at 11:45 AM, Bruce B bruceb...@gmail.com wrote:
 Thanks for the feedback Ryan.
 Siproxd is not installed. I think Siproxd like you said just does the
 reverse meaning if phones are part of pfSense subnet then it connects to
 outside world. But in my case they are coming into Asterisk which is on
 pfSense subnet. I do have a static IP and it's set like:
 externip=34.34.34.34
 localnet=192.168.5.0/255.255.255.0
 Do you use pfSense for this same situation? Can you do a sip show peers and
 let me know if you actually see the outside public IP addresses for the
 clients? Also how is your outbound NAT setup? AON?
 Thanks


Yep I am using pfSense 1.2.3 with a static IP. I have port forwarded
UDP SIP and the UDP RTP port range to the private IP of the Asterisk
box. I have enabled manual outbound nat and configured the static port
option. If you use the automatic outbound nat it will randomize the
ports, which you don't want. My sip.conf looks like yours with the
externip and localnet set. When I do sip show peers I see the external
IP.

Ryan

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Re: [asterisk-users] Why does sip show peers show my router/gateway address as the client IP address?

[Ryan Wagoner]

On Sat, Dec 11, 2010 at 1:04 PM, Bruce B bruceb...@gmail.com wrote:
 Thanks for the confirmation. Do you have both LAN and WAN as outbound AON
 like this:
 WAN any * * * * * YES
 LAN  any * * * * * YES
 ???
 I am stumped as to why pfSense behaves like this in this instance.
 Thanks again.

You only want one outbound NAT if you only have WAN and LAN interfaces. Mine is

WAN 192.168.1.0/24 * * * * * YES

Replace 192.168.1.0/24 with your internal network range.

Ryan

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Re: [asterisk-users] Asterisk (1.8-beta2) and SIP IPv4/IPv6 dual-stack possibilities

[Ryan Wagoner]

On Wed, Aug 4, 2010 at 10:44 AM, Wouter Schoot wou...@schoot.org wrote:
 Dear list,

 I'm trying to get Asterisk to work dual-stack on Linux and I'm left with
 a question.

 Imagine that a user (on the road) connects to Asterisk from various
 places. Many of them probably don't have IPv6 support yet. However, his
 house and office do have IPv6 connectivity. I would like to make sure
 that whenever IPv6 is available, the connection will be made over IPv6,
 but offer IPv4 as a fallback option.

 The pitfall, in my opinion, is to create one sip.conf entry for that
 user which supports the voicecalls over IPv4 and IPv6. However, settings
 like nat=, directmedia= and/or canreinvite= seem to be addressfamily
 unrelated. I want to configure it in a way that when I connect using
 IPv6, no NAT options should be set and the mediapath (almost) always
 should be directly between the peers and not over the Asterisk server
 (so, nat=no and canreinvite=yes).

 But, when a user comes via IPv4, changes are that he's on NAT. When that
 happens obviously the connections should traverse the NAT using options
 like nat=yes and canreinvite=no.

 There's little to no documentation available as far as my google-skills
 go. There's some in sip.conf, and I couldn't find anything on the website.

 Does anyone have some pointers for me, either for the configuration of
 the sip.conf entry or for more documentation on this?

 Best regards,

 Wouter Schoot


I'm interested in this as well. I tried binding Asterisk to both IPv4
and IPv6 addresses, but Asterisk keeps printing the following warnings

WARNING[3542]: chan_sip.c:3183 ast_sip_ouraddrfor: Address remapping
activated in sip.conf but we're using IPv6, which doesn't need it.
Please remove localnet and/or externaddr settings.

I need localnet and externaddr for IPv4 clients behind NAT. I also
want IPv6 support for clients that support it. It seems that it is not
possible to run Asterisk in a dual stack configuration and support
clients behind NAT.

Ryan

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Re: [asterisk-users] Polycom Park by EFK

[Ryan Wagoner]

On Fri, Dec 3, 2010 at 8:02 PM, Andrew Joakimsen joakim...@gmail.com wrote:
 Has anyone gotten one-touch call parking to work on Polycom phones via
 the Enhanced Feature Keys feature working? I've looked at various
 examples, they appear correct, but the phones (501, 3.1.x firmware)
 show the Park button while in a call but this does not actually cause
 the call to be parked. Doing a SIP debug, I don't see that anything is
 transmitted as a result of pressing the call park key. My
 understanding of the below configuration is it should cause the DTMF
 sequence #70 to be sent across the SIP channel -- but it isn't.


I have mine configured to park using the softkey settings. The action
does a phone side blind transfer to 70.

  softkey
softkey.1.label=Park Call

softkey.1.action=$FTransfer$$FDialpad7$$FDialpad0$$FDialpadPound$$Cp3$$Chu$
softkey.1.enable=1
softkey.1.use.active=1
softkey.1.use.hold=1
softkey.1.precede=0
  /

Ryan

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[asterisk-users] 1.8 Console Welcome Message

[Ryan Wagoner]

With previous Asterisk versions when running asterisk -r a welcome
message is displayed with the version. I just upgraded to 1.8 and
noticed it is not appearing. All I get is Verbosity is at least 3 and
the console prompt. I looked at main/asterisk.c and still see the
welcome message code. Any idea why it is not being shown?

Ryan

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Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces

[Ryan Wagoner]

The Loop Back Plug on the link you provided is correct. You take a
few inches of CAT5 and remove the outer jacket. Loop the wires into
the RJ-45 connector like the diagram shows and then crimp.

Ryan

On Tue, Oct 5, 2010 at 3:02 PM, Danny Dias ing.diasda...@gmail.com wrote:
 Hello my friend Ingmar,
 I would like to know the cable you used? how was the connection? i'm using
 this one:
 http://wiki.sangoma.com/Pinouts#A108 Loop Back
 Is this ok? what should i do my friend, my problems are understand the
 fisicall connection :(
 Best Regards!!!

 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl

 Hi DD,



 We usually use loopback cables and use the open source SIP test tool
 “SIPp” to initiate SIP calls that are sent from one group of 4 ports to
 another group of 4 ports.



 Met vriendelijke groet,

 Ingmar Steen

 Teleknowledge



 Van: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Dias
 Verzonden: vrijdag 24 september 2010 11:05
 Aan: Asterisk Users Mailing List - Non-Commercial Discussion
 Onderwerp: [asterisk-users] How to test BIG traffic through
 DAHDI/WANPIPEinterfaces



 Hello Community,



 I need to test or simulate many calls through dahdi/wanpipe, i have a
 Sangoma A108D, and i need to test the stability of the
 card/drivers/firmwares with a test environment, do you think is possible?



 What should i do? using some loopback cable maybe?



 Thanks in advance



 DD

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[asterisk-users] Asterisk 1.6.2.12 Download

[Ryan Wagoner]

Anybody else notice that the 1.6.2.12 download has a version and
changelog for 1.6.2.12-rc1?

http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.12.tar.gz

Ryan

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Re: [asterisk-users] 'dirty' upgrade of 1.4

[Ryan Wagoner]

When you run make, it compiles the binaries in the src directory. Once
it is done compiling stop asterisk. Running make install will copy the
compiled binaries into their respective folders on your system. Then
just start asterisk. If you need to revert, stop asterisk, run make
install in the old src directory, then start asterisk.

Ryan

On Mon, Jul 26, 2010 at 9:45 AM, Andrew Thomas a...@datavox.co.uk wrote:
 Hi Danny,

 I understand (and welcome) the separate src directories.  This would
 allow me to 'revert' should I feel the need (assuming I can just
 re-compile over each one).  I just need to know if I can re-compile over
 the existing first.

 Thanks for your reply.



 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas
 Sent: 26 July 2010 14:15
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4


From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
Subject: [asterisk-users] 'dirty' upgrade of 1.4

Apologies if this has been asked before.

Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1?

Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the
 source files for 1.4.34 over the top of the existing 1.4.24.1 files.

Also, will I need to stop * to perform this routine - or can I just
 'upgrade' and then do a * 'restart'?

 Question 1 - unless you are un-tarring to a specific directory, you
 would have /usr/local/src/asterisk-1.4.24.1 and
 /usr/local/src/asterisk-1.4.34 segregated source trees.

 Question 2 - you don't have to stop asterisk, but you should (best
 practice?) since installing a new release usually involves
 removing/replacing the .so files in /usr/lib/asterisk/modules.



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[asterisk-users] Exchange UM Play on Phone

[Ryan Wagoner]

I haven't been successful in getting this to work. The issue looks to
be that Asterisk is wanting peer authentication for the invite request
as it sends back 401 Unauthorized.  I am using FreePBX 2.7 and have
tested both Asterisk 1.6.1.18 and 1.6.2.9. My trunk settings are

type=peer
transport=tcp
qualify=yes
insecure=port,invite
host=10.10.1.31
context=from-internal

Here is snippets of the SIP debug output. I added in the debug Peer
has insecure flags to see what was happening.

INVITE sip:2...@voip.mydomain.net;user=phone SIP/2.0
FROM: sip:2...@exch.testdev.local;user=phone;epid=079E8F8013;tag=849256682
TO: sip:2...@voip.mydomain.net;user=phone
...
Sending to 10.10.1.31 : 19219 (no NAT)
Using INVITE request as basis request - 5c97e0f0-5456-4b82-a7f4-e7f2adeba338
Found peer '2001' for '2001' from 10.10.1.31:19219
Peer has insecure flags no

SIP/2.0 401 Unauthorized

Due to Exchange making the call from / to the same valid extension
Asterisk is wanting authentication for the 2001. I thought by using
host and insecure in the trunk settings if the from address matched
the host it would use that as the peer. Alternatively I couldn't find
the option to tell Exchange to make the call from a different
extension. In looking at an anonymous call Asterisk doesn't have a
peer for the from number so it looks in from-sip-external.

INVITE sip:1112223...@voip.mydomain.net SIP/2.0
From: 111222 sip:1112223...@voip.remotedomain.com;tag=as1c8404f3
To: sip:2223334...@voip.mydomain.net
...
Sending to xxx.xxx.xxx.xxx : 5060 (no NAT)
Using INVITE request as basis request -
29989544375bf8a162da163d1d9df...@voip.remotedomain.com
No matching peer for '111222' from 'xxx.xxx.xxx.xxx:5060'
Looking for 222333 in from-sip-external (domain voip.mydomain.net)

Thanks,
Ryan

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Re: [asterisk-users] Exchange UM Play on Phone

[Ryan Wagoner]

On Sat, Jul 24, 2010 at 12:07 PM, Ryan Wagoner rswago...@gmail.com wrote:
 I haven't been successful in getting this to work. The issue looks to
 be that Asterisk is wanting peer authentication for the invite request
 as it sends back 401 Unauthorized.  I am using FreePBX 2.7 and have
 tested both Asterisk 1.6.1.18 and 1.6.2.9. My trunk settings are

 type=peer
 transport=tcp
 qualify=yes
 insecure=port,invite
 host=10.10.1.31
 context=from-internal

 Here is snippets of the SIP debug output. I added in the debug Peer
 has insecure flags to see what was happening.

 INVITE sip:2...@voip.mydomain.net;user=phone SIP/2.0
 FROM: sip:2...@exch.testdev.local;user=phone;epid=079E8F8013;tag=849256682
 TO: sip:2...@voip.mydomain.net;user=phone
 ...
 Sending to 10.10.1.31 : 19219 (no NAT)
 Using INVITE request as basis request - 5c97e0f0-5456-4b82-a7f4-e7f2adeba338
 Found peer '2001' for '2001' from 10.10.1.31:19219
 Peer has insecure flags no

 SIP/2.0 401 Unauthorized

 Due to Exchange making the call from / to the same valid extension
 Asterisk is wanting authentication for the 2001. I thought by using
 host and insecure in the trunk settings if the from address matched
 the host it would use that as the peer. Alternatively I couldn't find
 the option to tell Exchange to make the call from a different
 extension. In looking at an anonymous call Asterisk doesn't have a
 peer for the from number so it looks in from-sip-external.

 INVITE sip:1112223...@voip.mydomain.net SIP/2.0
 From: 111222 sip:1112223...@voip.remotedomain.com;tag=as1c8404f3
 To: sip:2223334...@voip.mydomain.net
 ...
 Sending to xxx.xxx.xxx.xxx : 5060 (no NAT)
 Using INVITE request as basis request -
 29989544375bf8a162da163d1d9df...@voip.remotedomain.com
 No matching peer for '111222' from 'xxx.xxx.xxx.xxx:5060'
 Looking for 222333 in from-sip-external (domain voip.mydomain.net)

 Thanks,
 Ryan


Looks like I just answered my own question. You can't have a device
that matches the user extension. With it configured like this the
invite from won't match a SIP peer and it will default to IP lookup.

Using INVITE request as basis request - 413feda8-186c-4fe1-a82d-eb074be527e1
Found peer 'exchange-vm' for '2001' from 10.10.1.31:63436
Peer has insecure flags port,invite
Looking for 2001 in from-internal (domain voip.mydomain.net)

Ryan

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Re: [asterisk-users] Integration with Toshiba Strata DK424

[Ryan Wagoner]

On Sat, Jul 24, 2010 at 12:44 PM, Joel Maslak jmas...@antelope.net wrote:
 I'm posting here in case anyone else runs into this and needs some help.
 I'll probably update the voip-info Wiki pages on Toshiba integration in a
 bit.  Asterisk 1.6 makes things a bit easier than what is on that page.

 I'm integrating an Asterisk server with a Toshiba Strata system at my
 office.  Right now, it is driving some VoIP phones (Cisco ATAs with analog
 phones plugged into them) at a neighboring building (I work for a great
 company which actually temporarily moved us into another building when the
 air conditioning broke in ours), to provide phone service during this mini
 disaster - with a Wifi connection to the HQ building and VoIP, we had full
 services within a day or so, rather than weeks if we had waited for the
 telephone company.  We're now moving the IT department over to the system
 (Polycom IP450s) permanently, as it has worked so well with the Cisco ATAs.
 Next we plan on replacing the voicemail/auto-attendant box on the Toshiba
 with Asterisk.  Following that, the next step will be to eliminate the
 incoming POTS lines into the Toshiba and replace them with an ISDN PRI line
 going into the Asterisk box.  Finally, we expect a slow but study migration
 of users over to VoIP.

 The connection between Asterisk and the Toshiba is a set of analog phone
 lines (the Toshiba provides dialtone).  Later, I'll need additional lines
 where the Asterisk box generates dialtone (when we move to the ISDN) as
 well.

 I needed a way of having a Toshiba extension ring the appropriate phone on
 the Asterisk box.  It turns out that the best way to do this is to tell the
 Toshiba that the Asterisk box is a voicemail server.  You need to set some
 stuff up on the Toshiba to do this.  I set up the box in VM Group 2 (group 1
 is the legacy VM), and as hunt group 851/901 (users dial 851, but Toshiba
 knows it as 901).  Your configuration might be a bit different.  Here's what
 I did on the Toshiba:

 - All analog ports for the Asterisk box need to be set up in Toshiba Program
 31 with LEDs 06 (VM Group 2), LED 15 (Toshiba Strata VM Integration - send
 A/D tones), LED 16 (Receive VM ID code), LED 17 (End to End DTMF passing),
 and LED 18 (Privacy Override Blocking).  The hunt group needs to be set up
 with Toshiba program *40.

 This means that any Toshiba extension forwarded (either after no answer as a
 VM box is configured or via standard forward-all-calls) to the Asterisk box
 (x851) will cause the call to send 91xxx (xxx is the called extension, the
 one doing the forwarding) to the Asterisk box, after the Asterisk box
 answers via DTMF.  It sends 92xxx if you hit the message light on a
 Toshiba phone when it is lit.  For some unknown reason, I've also seen #
 or ## sent before or after the extension 91xxx/92xxx code - I couldn't
 tell you what that means, I just strip them in the Asterisk dialplan.  The
 Asterisk box can turn the message waiting indicator on/off by sending #63xxx
 (xxx = extension with the light) or #64xxx.

 If x851 is called directly, no DTMF is sent in the first few seconds.

 When the Toshiba system disconnects one of these analog lines (the
 Toshiba/PTSN user hung up), a DTMF D is sent.  When an outbound call is
 made from Asterisk to a Toshiba extension, the Toshiba sends a DTMF A when
 the call is answered.  The D tone in particular is important, because the
 Toshiba's analog extension cards don't do any other form of disconnect
 supervision.

 To detect the D tone (and semi-mute the A tone), I set up two features
 in features.conf.

 toshibahangup = D,self,Hangup
 toshibaanswer = A,peer/callee,Noop

 One thing I discovered was that the toshibahangup feature, when enabled,
 would only work for typical calls that were bridged.  They would not work
 for things like voicemail.  The voip-info wiki suggests using a meetme
 conference that listens for D, but I think using the feature is cleaner -
 when it works.

 So, to make the call bridged, I send it across a dhadi local span.  My
 DAHDI configuration - system.conf:

 fxsks=1-8

 dynamic=loc,1:0,31,0
 dynamic=loc,1:1,31,0

 bchan=9-23,25-39
 dchan=24

 bchan=40-54,56-70
 dchan=55

 Channels 1-8 are the analog lines to the Toshiba.  I then set up two E1
 spans (they aren't physical spans, but all stay within the system).
 Basically, channel 9 ends up connected to channel 40.

 Now, when a call comes in, if digits are provided early, I dial out via one
 virtual E1 span into the other one.  I listen for the disconnect/answer
 tones on the side facing the Toshiba.  The other side connects to the
 extension, which immediately answer()'s and then dial()'s the destination
 extension.  This makes Asterisk actually bridge the call between the Toshiba
 and the fake E1, letting it hear the DTMF and process it.  Now the D tones
 work and the A tones are mostly muted.  Outbound calls work a similar way,
 just in reverse.

 If nothing is dialed by the Toshiba on an 

Re: [asterisk-users] Exchange UM Play on Phone

[Ryan Wagoner]

On Sat, Jul 24, 2010 at 12:30 PM, Ryan Wagoner rswago...@gmail.com wrote:
 On Sat, Jul 24, 2010 at 12:07 PM, Ryan Wagoner rswago...@gmail.com wrote:
 I haven't been successful in getting this to work. The issue looks to
 be that Asterisk is wanting peer authentication for the invite request
 as it sends back 401 Unauthorized.  I am using FreePBX 2.7 and have
 tested both Asterisk 1.6.1.18 and 1.6.2.9. My trunk settings are

 type=peer
 transport=tcp
 qualify=yes
 insecure=port,invite
 host=10.10.1.31
 context=from-internal

 Here is snippets of the SIP debug output. I added in the debug Peer
 has insecure flags to see what was happening.

 INVITE sip:2...@voip.mydomain.net;user=phone SIP/2.0
 FROM: 
 sip:2...@exch.testdev.local;user=phone;epid=079E8F8013;tag=849256682
 TO: sip:2...@voip.mydomain.net;user=phone
 ...
 Sending to 10.10.1.31 : 19219 (no NAT)
 Using INVITE request as basis request - 5c97e0f0-5456-4b82-a7f4-e7f2adeba338
 Found peer '2001' for '2001' from 10.10.1.31:19219
 Peer has insecure flags no

 SIP/2.0 401 Unauthorized

 Due to Exchange making the call from / to the same valid extension
 Asterisk is wanting authentication for the 2001. I thought by using
 host and insecure in the trunk settings if the from address matched
 the host it would use that as the peer. Alternatively I couldn't find
 the option to tell Exchange to make the call from a different
 extension. In looking at an anonymous call Asterisk doesn't have a
 peer for the from number so it looks in from-sip-external.

 INVITE sip:1112223...@voip.mydomain.net SIP/2.0
 From: 111222 sip:1112223...@voip.remotedomain.com;tag=as1c8404f3
 To: sip:2223334...@voip.mydomain.net
 ...
 Sending to xxx.xxx.xxx.xxx : 5060 (no NAT)
 Using INVITE request as basis request -
 29989544375bf8a162da163d1d9df...@voip.remotedomain.com
 No matching peer for '111222' from 'xxx.xxx.xxx.xxx:5060'
 Looking for 222333 in from-sip-external (domain voip.mydomain.net)

 Thanks,
 Ryan


 Looks like I just answered my own question. You can't have a device
 that matches the user extension. With it configured like this the
 invite from won't match a SIP peer and it will default to IP lookup.

 Using INVITE request as basis request - 413feda8-186c-4fe1-a82d-eb074be527e1
 Found peer 'exchange-vm' for '2001' from 10.10.1.31:63436
 Peer has insecure flags port,invite
 Looking for 2001 in from-internal (domain voip.mydomain.net)

 Ryan


There has got to be a better solution to this involving the invite
from field peer domain. It looks like find_peer just matches on the
name and ignores the domain. If domain support is enabled shouldn't we
only find SIP peers if the from domain on the invite matches one in
the list? The sip invites I have looked at from Polycom and Linksys
devices put use...@registrationserver for the from. Or am I missing
something that this would break?

Ryan

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Re: [asterisk-users] Exchange UM Play on Phone

[Ryan Wagoner]

On Sat, Jul 24, 2010 at 9:25 PM, Ryan Wagoner rswago...@gmail.com wrote:
 On Sat, Jul 24, 2010 at 12:30 PM, Ryan Wagoner rswago...@gmail.com wrote:
 On Sat, Jul 24, 2010 at 12:07 PM, Ryan Wagoner rswago...@gmail.com wrote:
 I haven't been successful in getting this to work. The issue looks to
 be that Asterisk is wanting peer authentication for the invite request
 as it sends back 401 Unauthorized.  I am using FreePBX 2.7 and have
 tested both Asterisk 1.6.1.18 and 1.6.2.9. My trunk settings are

 type=peer
 transport=tcp
 qualify=yes
 insecure=port,invite
 host=10.10.1.31
 context=from-internal

 Here is snippets of the SIP debug output. I added in the debug Peer
 has insecure flags to see what was happening.

 INVITE sip:2...@voip.mydomain.net;user=phone SIP/2.0
 FROM: 
 sip:2...@exch.testdev.local;user=phone;epid=079E8F8013;tag=849256682
 TO: sip:2...@voip.mydomain.net;user=phone
 ...
 Sending to 10.10.1.31 : 19219 (no NAT)
 Using INVITE request as basis request - 5c97e0f0-5456-4b82-a7f4-e7f2adeba338
 Found peer '2001' for '2001' from 10.10.1.31:19219
 Peer has insecure flags no

 SIP/2.0 401 Unauthorized

 Due to Exchange making the call from / to the same valid extension
 Asterisk is wanting authentication for the 2001. I thought by using
 host and insecure in the trunk settings if the from address matched
 the host it would use that as the peer. Alternatively I couldn't find
 the option to tell Exchange to make the call from a different
 extension. In looking at an anonymous call Asterisk doesn't have a
 peer for the from number so it looks in from-sip-external.

 INVITE sip:1112223...@voip.mydomain.net SIP/2.0
 From: 111222 sip:1112223...@voip.remotedomain.com;tag=as1c8404f3
 To: sip:2223334...@voip.mydomain.net
 ...
 Sending to xxx.xxx.xxx.xxx : 5060 (no NAT)
 Using INVITE request as basis request -
 29989544375bf8a162da163d1d9df...@voip.remotedomain.com
 No matching peer for '111222' from 'xxx.xxx.xxx.xxx:5060'
 Looking for 222333 in from-sip-external (domain voip.mydomain.net)

 Thanks,
 Ryan


 Looks like I just answered my own question. You can't have a device
 that matches the user extension. With it configured like this the
 invite from won't match a SIP peer and it will default to IP lookup.

 Using INVITE request as basis request - 413feda8-186c-4fe1-a82d-eb074be527e1
 Found peer 'exchange-vm' for '2001' from 10.10.1.31:63436
 Peer has insecure flags port,invite
 Looking for 2001 in from-internal (domain voip.mydomain.net)

 Ryan


 There has got to be a better solution to this involving the invite
 from field peer domain. It looks like find_peer just matches on the
 name and ignores the domain. If domain support is enabled shouldn't we
 only find SIP peers if the from domain on the invite matches one in
 the list? The sip invites I have looked at from Polycom and Linksys
 devices put use...@registrationserver for the from. Or am I missing
 something that this would break?

 Ryan


I have developed a patch that checks the invite from field domain
against the domain list when domain support is enabled.

https://issues.asterisk.org/view.php?id=17700

Ryan

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Re: [asterisk-users] [NAT] * + private IP + locked-down firewalls?

[Ryan Wagoner]

On Fri, Jul 9, 2010 at 4:28 AM, Gilles codecompl...@free.fr wrote:
 On Mon, 05 Jul 2010 12:45:34 +0200, Gilles codecompl...@free.fr
 wrote:
Provided the user doesn't have access to the firewall (eg. corporate
or hotel), and the firewall doesn't allow dynamic port opening through
UPnP or NAT-PMP...

 For those interested, I was tipped through private e-mail about using
 OpenVPN to open a steady tunnel between the client and Asterisk, and
 have the SIP client send packets through that tunnel instead of trying
 to connect directly.




I have around 50 Snom 370s configured this way. They work great for
remote workers. However the Snom speakerphone is terrible compared to
Aastra and Polycom. If there is any background noise it will cut in
and out the other party.

Ryan

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Re: [asterisk-users] Update the LCD with the callee's name after dialing

[Ryan Wagoner]

On Tue, Jul 6, 2010 at 10:19 AM,  unsero...@aol.com wrote:
 The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
 compile but need to be tested to verify that they work. I have the
 1.6.2.9 version in production and plan to put the 1.6.1.20 version in
 sometime this weekend.

 In you are just using Asterisk in the dialplan you can set the called
 remote party id with something like below. Otherwise check out the
 previous FreePBX 2.7 patch.

 exten =



 100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)
 })

 Ryan

 If you installed Asterisk from source you just need to patch and
 recompile / install.

 cd asterisk-version
 patch -p1  ../asterisk-verson-called-
 rpid.patch
 make install

 Otherwise if your using trixbox, etc you would probably want to grab
 their SRPMS, add the patch to the spec file, and rebuild them. However
 that is outside of the scope of this mailing list.

 Ryan

 Which version of Asterisk? The patches were made against the latest
 releases. If you are running an earlier version you might need to
 manually patch your install.

 Ryan

 --

 Version 1.6.1.20

 But it was my individual problem. Installing from scratch solved the
 patching issue.

 Now the application SIPCalledRPID is active and gets executed but i still
 don't get the name of the called person

 on my display. Maybe this is client dependent? I am using 3CX Softphone.
 Or
 is somethins else missing?


 The client needs to support the Remote-Party-ID SIP header. If you
 want to verify the header is being added run tcpdump and analyze it
 with Wireshark. I know that Polycom phones have support for this. I
 just put a modified version of the Asterisk 1.6.1 patch into
 production for 25 Polycom phones, soon to be 150 phones. I changed the
 return -1 to return 0 so that the call continues even if they
 SIPCalledRPID args are invalid.

 Ryan

 --
 Just to make sure that we are talking about the same issue.

 What I want is that when two users are registered at the same peer that

 when user A calls user B user A gets the name of user B displayed on his
 client.

 Is this what you are trying to fix with the patch?

 Because from my understanding as an absolute newbie to SIP and Asterisk, the
 header

 should already contain the let's call it displayname and look something
 like

 INVITE sip:2...@192.168.1.10:5060 SIP/2.0
 Via: SIP/2.0/TCP
 192.168.1.149:3822;branch=z9hG4bK-d8754z-9f01b74a4b708b04-1---d8754z-;rport
 Max-Forwards: 70
 Contact:
 sip:1...@192.168.1.149:3823;rinstance=8f3067c0aac0abc4;transport=TCP
 To: Callee Name sip:2...@192.168.1.10:5060
 From: Caller Name sip:1...@192.168.1.10:5060;tag=cf41cd30

 according to SIP rfc 3261 http://tools.ietf.org/html/rfc3261


Yes that is what the patch addresses. The phones will only display the
name of the called extension if Remote-Party-ID or P-Asserted-Identity
is set.

Ryan

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Re: [asterisk-users] Update the LCD with the callee's name after dialing

[Ryan Wagoner]

On Thu, Jul 1, 2010 at 3:26 PM,  unsero...@aol.com wrote:




 -Original Message-
 From: Ryan Wagoner rswago...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thu, Jul 1, 2010 6:19 pm
 Subject: Re: [asterisk-users] Update the LCD with the callee's name after
 dialing

 On Thu, Jul 1, 2010 at 11:52 AM,  unsero...@aol.com wrote:
 Thanks a lot.

 Applying the patch gives me a

 Hunk #5 failed at 9881



 -Original Message-
 From: Ryan Wagoner rswago...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thu, Jul 1, 2010 5:37 pm
 Subject: Re: [asterisk-users] Update the LCD with the callee's name after
 dialing

 On Thu, Jul 1, 2010 at 11:29 AM,  unsero...@aol.com wrote:
 Sounds great.

 Could you please give me a hint how to install the patch?
 Sorry for my stupid question but I'm a newbie to Asterisk.

 Thanks.



 -Original Message-
 From: Ryan Wagoner rswago...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thu, Jul 1, 2010 5:06 pm
 Subject: Re: [asterisk-users] Update the LCD with the callee's name after
 dialing

 On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
 Ryan Wagoner wrote:

 together one for 1.4 that compiles. I'll post both to the list
 hopefully later today.



 Thank you!

 Doug

 --


 The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
 compile but need to be tested to verify that they work. I have the
 1.6.2.9 version in production and plan to put the 1.6.1.20 version in
 sometime this weekend.

 In you are just using Asterisk in the dialplan you can set the called
 remote party id with something like below. Otherwise check out the
 previous FreePBX 2.7 patch.

 exten =


 100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})

 Ryan

 If you installed Asterisk from source you just need to patch and
 recompile / install.

 cd asterisk-version
 patch -p1  ../asterisk-verson-called-
 rpid.patch
 make install

 Otherwise if your using trixbox, etc you would probably want to grab
 their SRPMS, add the patch to the spec file, and rebuild them. However
 that is outside of the scope of this mailing list.

 Ryan

 Which version of Asterisk? The patches were made against the latest
 releases. If you are running an earlier version you might need to
 manually patch your install.

 Ryan

 --

 Version 1.6.1.20

 But it was my individual problem. Installing from scratch solved the
 patching issue.

 Now the application SIPCalledRPID is active and gets executed but i still
 don't get the name of the called person

 on my display. Maybe this is client dependent? I am using 3CX Softphone. Or
 is somethins else missing?


The client needs to support the Remote-Party-ID SIP header. If you
want to verify the header is being added run tcpdump and analyze it
with Wireshark. I know that Polycom phones have support for this. I
just put a modified version of the Asterisk 1.6.1 patch into
production for 25 Polycom phones, soon to be 150 phones. I changed the
return -1 to return 0 so that the call continues even if they
SIPCalledRPID args are invalid.

Ryan

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Re: [asterisk-users] Update the LCD with the callee's name after dialing

[Ryan Wagoner]

On Thu, Jul 1, 2010 at 5:55 AM, Doug Lytle supp...@drdos.info wrote:
 CunningPike wrote:
 On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnellmattdarn...@gmail.com  wrote:


 We use the patch in https://issues.asterisk.org/view.php?id=6643. Works 
 great.



 There is a much newer patch for 1.4 that can be found at:

 https://issues.asterisk.org/view.php?id=8824

 But, it won't apply cleanly on the latest 1.4 series.  It's like 4
 versions back.  Once I get into work, I'll post the version I'm running
 it on.

 Doug



This is the version that went into trunk for 1.8. It should send the
remote party id without dialplan changes. I had looked into using it
with 1.6.1 and 1.6.2. However due to the number of changes since the
patch was merged I was worried that I would introduce bugs. The
previous patch is simple, but does require a one line dial plan
change.

On the previous patch I posted for 1.6.2 I also have a 1.6.1 version.
It compiles but hasn't been tested. Let me see if I can quickly put
together one for 1.4 that compiles. I'll post both to the list
hopefully later today.

Ryan

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Re: [asterisk-users] Update the LCD with the callee's name after dialing

[Ryan Wagoner]

On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
 Ryan Wagoner wrote:

 together one for 1.4 that compiles. I'll post both to the list
 hopefully later today.



 Thank you!

 Doug

 --


The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to be tested to verify that they work. I have the
1.6.2.9 version in production and plan to put the 1.6.1.20 version in
sometime this weekend.

In you are just using Asterisk in the dialplan you can set the called
remote party id with something like below. Otherwise check out the
previous FreePBX 2.7 patch.

exten = 
100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})

Ryan


asterisk-1.6.1.20-called-rpid.patch
Description: Binary data


asterisk-1.4.33.1-called-rpid.patch
Description: Binary data
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Re: [asterisk-users] Update the LCD with the callee's name after dialing

[Ryan Wagoner]

On Thu, Jul 1, 2010 at 11:29 AM,  unsero...@aol.com wrote:
 Sounds great.

 Could you please give me a hint how to install the patch?
 Sorry for my stupid question but I'm a newbie to Asterisk.

 Thanks.



 -Original Message-
 From: Ryan Wagoner rswago...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thu, Jul 1, 2010 5:06 pm
 Subject: Re: [asterisk-users] Update the LCD with the callee's name after
 dialing

 On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
 Ryan Wagoner wrote:

 together one for 1.4 that compiles. I'll post both to the list
 hopefully later today.



 Thank you!

 Doug

 --


 The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
 compile but need to be tested to verify that they work. I have the
 1.6.2.9 version in production and plan to put the 1.6.1.20 version in
 sometime this weekend.

 In you are just using Asterisk in the dialplan you can set the called
 remote party id with something like below. Otherwise check out the
 previous FreePBX 2.7 patch.

 exten =
 100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})

 Ryan

If you installed Asterisk from source you just need to patch and
recompile / install.

cd asterisk-version
patch -p1  ../asterisk-verson-called-rpid.patch
make install

Otherwise if your using trixbox, etc you would probably want to grab
their SRPMS, add the patch to the spec file, and rebuild them. However
that is outside of the scope of this mailing list.

Ryan

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Re: [asterisk-users] Update the LCD with the callee's name after dialing

[Ryan Wagoner]

On Thu, Jul 1, 2010 at 11:52 AM,  unsero...@aol.com wrote:
 Thanks a lot.

 Applying the patch gives me a

 Hunk #5 failed at 9881



 -Original Message-
 From: Ryan Wagoner rswago...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thu, Jul 1, 2010 5:37 pm
 Subject: Re: [asterisk-users] Update the LCD with the callee's name after
 dialing

 On Thu, Jul 1, 2010 at 11:29 AM,  unsero...@aol.com wrote:
 Sounds great.

 Could you please give me a hint how to install the patch?
 Sorry for my stupid question but I'm a newbie to Asterisk.

 Thanks.



 -Original Message-
 From: Ryan Wagoner rswago...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thu, Jul 1, 2010 5:06 pm
 Subject: Re: [asterisk-users] Update the LCD with the callee's name after
 dialing

 On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote:
 Ryan Wagoner wrote:

 together one for 1.4 that compiles. I'll post both to the list
 hopefully later today.



 Thank you!

 Doug

 --


 The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
 compile but need to be tested to verify that they work. I have the
 1.6.2.9 version in production and plan to put the 1.6.1.20 version in
 sometime this weekend.

 In you are just using Asterisk in the dialplan you can set the called
 remote party id with something like below. Otherwise check out the
 previous FreePBX 2.7 patch.

 exten =

 100,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})

 Ryan

 If you installed Asterisk from source you just need to patch and
 recompile / install.

 cd asterisk-version
 patch -p1  ../asterisk-verson-called-
 rpid.patch
 make install

 Otherwise if your using trixbox, etc you would probably want to grab
 their SRPMS, add the patch to the spec file, and rebuild them. However
 that is outside of the scope of this mailing list.

 Ryan


Which version of Asterisk? The patches were made against the latest
releases. If you are running an earlier version you might need to
manually patch your install.

Ryan

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Re: [asterisk-users] Update the LCD with the callee's name after dialing

[Ryan Wagoner]

On Wed, Jun 30, 2010 at 6:10 PM, CunningPike cunningp...@gmail.com wrote:
 On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote:
 Thank you Andrew,

 I will check it out.  We are currently running 1.4.

 -Matt

 On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote:
 Remote Party ID in trunk, it works  There are hacks for other versions.


 We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great.

 CP


Until Asterisk 1.8 is released this looks like the easiest way to get
remote party id working. I have modified the patch to work with
Asterisk 1.6.2.9. I have also attached a patch against FreePBX 2.7 to
add the necessary changes to the dialplan. I have verified this works
on a Polycom 550.

Ryan


asterisk-1.6.2.9-called-rpid.patch
Description: Binary data


freepbx-2.7.0.8-core-called-rpid.patch
Description: Binary data
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Re: [asterisk-users] Codec negotiation

[Ryan Wagoner]

On Tue, Jun 29, 2010 at 6:42 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
 Hi!

 Because the codec is already chosen before the call is made, and you
 told it that g722 is permitted.

 There are all sorts of discussions in play about codec negotiation,
 but at this point in time, if you want different behaviour you'll need to
 go and code it yourself

 Look at the list archive - there is a codec negotiation patch around:

 http://lists.digium.com/pipermail/asterisk-users/2010-
 February/244835.html

 The OP might also want to consider to use different lines to the same
 PBX, one for normal narrowband, and another one for g722.

 Philipp


 --

Thanks! I'm going to try setting the _SIP_CODEC variable for outbound
calls to force ulaw. This should solve the issue. Having two lines
would work but I can't sell this to a customer. There has got to be a
better way to have Asterisk handle this. With Asterisk in the middle
of the RTP stream it knows what both parties support. If it turns out
Asterisk is transcoding it could check for a common codec and
renegotiate one endpoint.

Ryan

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Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

[Ryan Wagoner]

On Wed, Jun 23, 2010 at 12:57 PM, James Lamanna jlama...@gmail.com wrote:
 On Tue, Jun 22, 2010 at 8:57 PM, Andres and...@telesip.net wrote:

 completely as well.

 Below I've posted a patch that responds with a 200 OK to these
 keep-alive requests, and I believe
 also solves the temporary loss of registration problem, though more
 testing in different environments
 for those who experience this problem would be greatly appreciated.

 The patch is against 1.4.32.


 A workaround we have used for a long time is to simply change the config
 on the Linksys phones to send an empty packet as a keep-alive.  There is
 obviously no response from asterisk but it keeps the NAT bindings alive
 and well on every router we have tested.

 Hi Andres,
 I have noticed that on Linksys phones that have a short REGISTER time,
 the lack of
 NAT keep alive responses can cause the phone to no longer be able to register.
 That's why I've spent a lot of effort to hopefully make these
 keep-alives supported.


 Andres
 http://www.neuroredes.com
 -- James

 -- James

 --

Is anybody running 1.6.2 with Linksys phones that would be willing to
help test the patch on https://issues.asterisk.org/view.php?id=17379

Ryan

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[asterisk-users] Codec negotiation

[Ryan Wagoner]

I have Polycom phones that support the g722 codec. Adding allow=g722
to the [general] section of sip.conf works great and I can make calls
between the phones using g722. However Asterisk is negotiating g722
for calls going out my voip provider and transcoding these to ulaw. In
sip.conf for the provider I have deny=all and allow=ulaw. This can
cause potential audio degrading and wastes cpu cycles. If Asterisk
knows the trunk only supports ulaw why would it offer g722 to the
phone.

Ryan

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Re: [asterisk-users] ISP down internal phones become unavailable

[Ryan Wagoner]

On Tue, Jun 22, 2010 at 9:33 AM, Mr Shunz mrsh...@gmail.com wrote:
 Hi all,

 I have a PRI, and when the Internet connection goes out so do my
 phones.  I suspect it is some type of DNS issue.  I do have a SIP
 trunk, and it appears that if I lose DNS to the SIP trunk, the entire
 PBX is offline.  I have no actual proof of any of this, and have not
 done any extensive testing to prove or disprove this.

 well, we have various asterisk installations, ranging from 1.4.25
 to (upgraded today) 1.4.33 (we don't use 1.6.X yet) and two
 of them show this behaviour...
 one is upgraded to 1.4.33, the other is 1.4.30, they have similar 
 configuration
 to all the other machines (which work flawlessy even when connection
 is down), and the phones are the same brand/model we use everywhere,
 with almost the same configuration.

 I'm not sure about a DNS issue because all our customers have local
 DNS/cache servers and we configure all the phones (and sip trunks
 on asterisks) with ip addresses and not FQDNs just to be sure...

 what we see is when the trunk goes down, i.e.
 'Registration for ...@yy.yy.yy.yy timed out, trying again (Attempt #ZZ)'
 we have also 'Peer XXX is now UNREACHABLE (internal phones), even if they
 are pingable/accessibile on the LAN...

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 dani...@santi.vr.it ..o () ascii ribbon campaign
 Linux User #415108  ooo /\  www.asciiribbon.org
 


It is interesting that you are seeing this on different machines with
the same Asterisk version. There must be something different in the
configuration or DNS. However Asterisk should gracefully handle no DNS
or a SIP provider issue without affecting the phones. I haven't been
able to troubleshoot this much since I can't just take the Internet
connection down.

Ryan

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Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

[Ryan Wagoner]

On Tue, Jun 22, 2010 at 6:26 PM, James Lamanna jlama...@gmail.com wrote:
 On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote:
 James Lamanna schrieb:
 If you've used Linksys phones against recent Asterisk 1.4.x you may
 have noticed
 that they may drop registration for a quick bit and then go back to being ok
 if your phone is behind NAT.
 If you turn Asterisk's sip debug information on, you'll probably find
 errors like these in your logs:

 NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce
 received from '11 sip:999...@208.90.186.10'

 I believe I have determined that this is caused by a bug in the
 Linksys firmware that is related to the NAT Keep-Alive packets.
 Because recent Asterisk 1.4.x's do not establish a SIP dialog for
 NOTIFY requests, the 489 Bad Event
 replies were going back to the wrong address if your phone was behind NAT.
 This lack of reply would cause the next REGISTER message to use the
 same nonce as the previous REGISTER,
 resulting in the stale nonce errors and temporarily dropping
 registration. I've also seen the lack of response to
 the NAT keep-alive cause the phone to stop being able to register
 completely as well.

 Below I've posted a patch that responds with a 200 OK to these
 keep-alive requests, and I believe
 also solves the temporary loss of registration problem, though more
 testing in different environments
 for those who experience this problem would be greatly appreciated.

 The patch is against 1.4.32.

 -- James

 Hello,

 you also just could set the NAT KEEP ALIVE MESSAGE on Ext 1 from $NOTIFY
 to $OPTIONS and make this extension in your default context:
 exten = s,1,hangup

 and you also would get a 200 ok for the keep alive package.

 IMHO a stale nonce would only occur when a user tries to register faster
 than 3600s cause of the register timeout used in asterisk. Maybe you
 should also try to set a higher register timeout on your phones. but i
 dont have an 1.4 system running, only around 2k of linksys phones on a
 1.2.40 and 300 on 1.6.1.18 and i dont see this problem there.

 I'm not sure how this works.
 The OPTIONS message fails chan_sip.c:parse_request() so the OPTIONS
 message never gets processed.
 The options message I receive from a Linksys942 6.1.3(a) looks like this:

 --- SIP read from xxx.xxx.xxx.xxx:8037 ---
 OPTIONS
 -

 -- James

 --

I had the same result when using $OPTIONS on a SPA941 phone with
firmware 5.1.8. I am running Asterisk 1.6.2.9 that has the keep-alive
support, however I still see Asterisk sending a 489 Bad Event. I just
reopened the issue and provided the necessary debug log at
https://issues.asterisk.org/bug_view_page.php?bug_id=17379

Ryan

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Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

[Ryan Wagoner]

On Tue, Jun 22, 2010 at 8:30 PM, James Lamanna jlama...@gmail.com wrote:
 On Tue, Jun 22, 2010 at 4:31 PM, Ryan Wagoner rswago...@gmail.com wrote:
 On Tue, Jun 22, 2010 at 6:26 PM, James Lamanna jlama...@gmail.com wrote:
 On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote:
 James Lamanna schrieb:
 If you've used Linksys phones against recent Asterisk 1.4.x you may
 have noticed
 that they may drop registration for a quick bit and then go back to being 
 ok
 if your phone is behind NAT.
 If you turn Asterisk's sip debug information on, you'll probably find
 errors like these in your logs:

 NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce
 received from '11 sip:999...@208.90.186.10'

 I believe I have determined that this is caused by a bug in the
 Linksys firmware that is related to the NAT Keep-Alive packets.
 Because recent Asterisk 1.4.x's do not establish a SIP dialog for
 NOTIFY requests, the 489 Bad Event
 replies were going back to the wrong address if your phone was behind NAT.
 This lack of reply would cause the next REGISTER message to use the
 same nonce as the previous REGISTER,
 resulting in the stale nonce errors and temporarily dropping
 registration. I've also seen the lack of response to
 the NAT keep-alive cause the phone to stop being able to register
 completely as well.

 Below I've posted a patch that responds with a 200 OK to these
 keep-alive requests, and I believe
 also solves the temporary loss of registration problem, though more
 testing in different environments
 for those who experience this problem would be greatly appreciated.

 The patch is against 1.4.32.

 -- James

 Hello,

 you also just could set the NAT KEEP ALIVE MESSAGE on Ext 1 from $NOTIFY
 to $OPTIONS and make this extension in your default context:
 exten = s,1,hangup

 and you also would get a 200 ok for the keep alive package.

 IMHO a stale nonce would only occur when a user tries to register faster
 than 3600s cause of the register timeout used in asterisk. Maybe you
 should also try to set a higher register timeout on your phones. but i
 dont have an 1.4 system running, only around 2k of linksys phones on a
 1.2.40 and 300 on 1.6.1.18 and i dont see this problem there.

 I'm not sure how this works.
 The OPTIONS message fails chan_sip.c:parse_request() so the OPTIONS
 message never gets processed.
 The options message I receive from a Linksys942 6.1.3(a) looks like this:

 --- SIP read from xxx.xxx.xxx.xxx:8037 ---
 OPTIONS
 -

 -- James

 --

 I had the same result when using $OPTIONS on a SPA941 phone with
 firmware 5.1.8. I am running Asterisk 1.6.2.9 that has the keep-alive
 support, however I still see Asterisk sending a 489 Bad Event. I just
 reopened the issue and provided the necessary debug log at
 https://issues.asterisk.org/bug_view_page.php?bug_id=17379

 Ryan,
 This is most likely because the packet never makes it to 
 handle_request_notify.
 I haven't looked at the code for 1.6.2.9 yet, but in 1.4.32 without my
 patch, the
 NOTIFY request would never make it out of find_call() and return early with a
 489 Bad Event response.

 Were you getting any response at 1.6.2.9 with the OPTIONS message?

 -- James

 --

The out of dialog support was the trick for 1.6.2.9 since it has
support for sending a keep-alive. I have attached a modified version
of your patch that worked for me. Do you mind if I attach the modified
version of the patch to my issue report?

Ryan


asterisk-1.6.2.9-keep-alive.patch
Description: Binary data
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[asterisk-users] ISP down internal phones become unavailable

[Ryan Wagoner]

I saw the following lines in the log this morning. From my router logs
I see that the connection went down as my ISP was doing maintenance
for a few minutes last night. I can understand the external
registrations timing out, but why do the phones become unreachable.
They are on the internal lan within the same subnet as the Asterisk
server. Internal DHCP and DNS was functional. If I had a PRI card in
this system as well that would mean I couldn't make phone calls
because the Internet is down.

Ryan

[Jun 21 01:51:26] NOTICE[13657]: chan_sip.c:11569 sip_reg_timeout:
-- Registration for '...@newyork.voip.ms' timed out, trying again
(Attempt #1)
[Jun 21 01:51:46] NOTICE[13657]: chan_sip.c:22943 sip_poke_noanswer:
Peer '1850' is now UNREACHABLE!  Last qualify: 15
[Jun 21 01:51:46] NOTICE[13657]: chan_sip.c:11569 sip_reg_timeout:
-- Registration for '...@sip.flowroute.com' timed out, trying
again (Attempt #1)
[Jun 21 01:52:07] NOTICE[13657]: chan_sip.c:22943 sip_poke_noanswer:
Peer '1800' is now UNREACHABLE!  Last qualify: 7
[Jun 21 01:52:07] NOTICE[13657]: chan_sip.c:22943 sip_poke_noanswer:
Peer '1801' is now UNREACHABLE!  Last qualify: 11
[Jun 21 01:52:07] NOTICE[13657]: chan_sip.c:11569 sip_reg_timeout:
-- Registration for '...@newyork.voip.ms' timed out, trying again
(Attempt #2)
  == Extension Changed 2028[ext-local] new state Unavailable for
Notify User 1850
[Jun 21 01:52:17] NOTICE[13657]: chan_sip.c:18314
handle_response_peerpoke: Peer '1800' is now Reachable. (10ms /
2000ms)
  == Extension Changed 2028[ext-local] new state Idle for Notify User 1850
[Jun 21 01:52:17] NOTICE[13657]: chan_sip.c:18314
handle_response_peerpoke: Peer '1801' is now Reachable. (14ms /
2000ms)
[Jun 21 01:52:22] NOTICE[13657]: chan_sip.c:18314
handle_response_peerpoke: Peer '1850' is now Reachable. (16ms /
2000ms)

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Re: [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)

[Ryan Wagoner]

On Sat, Jun 19, 2010 at 12:00 PM, James Lamanna jlama...@gmail.com wrote:
 On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote:
 James Lamanna schrieb:
 It appears as though the 489 Bad Event response to the NAT keep alive
 event responds to the local address, instead of responding to the
 NATted address.
 This causes Linksys phones to go amber (no registration) after a short
 amount of time after placing calls.
 Turning the Linksys NAT keep alive off is a workound, but non-ideal in
 may situations.

 Apparently the asterisk devs don't even think this is a bug:
 https://issues.asterisk.org/view.php?id=17532

 Has anyone dealt with this at all?

 Thanks.

 -- James

 Hello james,

 in the SPA config webpage on EXT 1 in the Nat Keep Alive MSG you should
 set $OPTIONS instead of $NOTIFY.

 then in your asterisk extension default context just set this:

 exten = s,1,Hangup

 then the phone will send a options packet and you will get a 200 OK
 instead of 489 Bad event.

 this should help.

 best regards

 Thanks Steve,
 I'll give that a try.
 I think I'll also look into why responses to NOTIFYs don't do the
 right thing in terms of NAT either.


 steve

 -- James


I have created an issue report on this a few weeks on with Asterisk
1.6.2.8-rc1. This was happening on a client site, which I didn't have
a chance to stop back by, so they closed the issue.

https://issues.asterisk.org/bug_view_page.php?bug_id=17379

It looked to me like Asterisk was rejecting the NOTIFY message due to
no callid, which is in the message. I couldn't figure out what was
going and there is code in 1.6.2.x to return a 200 OK to a NOTIFY
message.

Ryan

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Re: [asterisk-users] How to stop intruder from registering sip?

[Ryan Wagoner]

On Sun, Jun 13, 2010 at 4:06 PM, sean darcy seandar...@gmail.com wrote:
 On 06/13/2010 01:59 PM, Dave Platt wrote:
 If you leave your asterisk box open to the world with passwords like 
 you deserve to be hacked..

 Well, without making a moral judgment, I will agree that you are *going*
 to be hacked if you do this!

 The O.P. seems to have made two (fairly common) mistakes:

 -  Used a secret so obvious that it could be guessed... and
     even if not, so short that it could have been determined by
     a very simple brute-force attack.

 -  Used the user's extension number as the SIP user ID... and
     thus making it easy to figure out which user IDs on which a
     password attack could be carried out.

 Doing a brute-force SIP-registration attack against all
 possible 3- and 4-digit extensions, using a handful of
 obvious secret strings ( through , 1234, 4321,
 same number as the extension) wouldn't take an attacker
 very long at all.  Nor would trying to call all of these
 numbers once to figure out which extensions exist, then doing
 a brute-force password attack against those which exist.  I
 have no doubt that there are numerous crackers out on the
 net doing just these sorts of attacks on a regular basis.

 The cure for these problems is, obviously, don't do that:

 (1) SIP user IDs should not be based on the extension number,
      and preferably should not be based on the owner's name
      or user login.  Make 'em hard to guess or brute-force!

 (2) Make the secrets equally hard to guess or brute-force.
      No short strings of numbers, no dictionary words or
      simple leet-speak transforms of them, etc.

 One of your best tools is a program or script to generate
 random sequences of letters and digits and other legal-
 in-SIP-names characters.  Try something like

     dd if=/dev/urandom bs=512 count=1 | base64

 and then copy some 10- or 12-character substrings out of this
 mass of gibberish and use 'em for SIP secrets.  With this many
 bits of randomness in the secrets, they'll be effectively
 invulnerable to guessing or brute force attacks.

 Are your travelling people using softphones? If they are VPN would be a good
 idea..

 A very good idea, and not just for security reasons.  Running SIP over
 a VPN tunnel can be a very effective remedy for all sorts
 of firewall- and NAT-related problems.

 I've found that running OpenVPN between my various SIP clients,
 and my Asterisk server, produces far better results than depending
 on STUN or on SIP-aware routers and firewalls.


 Thanks for not suggesting I ponder my sins!

 As I mentioned, I'm not inclined to mess with the secrets, too much
 hassle for users. That's why I'm considering deny/permit.

 Does that solve my problem?

 But I'm struck with your notion of having sip user ids different from
 extensions. That would not require any user effort, or messing with each
 phone. But...

 We use a combo of aastra 9133i and 57i's. Don't the user id and the
 extension HAVE to be the same? I had thought the aastra's used the
 extension as the SIP id to register.

 sean


The deny/permit will work only for phones within your internal
network. It will not allow any remote phones to connect so how do you
plan on getting your remote users up and running?

How are secrets too much hassle? You set the password once and forget
it. With the Aastra phones you could setup phone provisioning files to
automate the process.

Ryan

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Re: [asterisk-users] Caller ID questions

[Ryan Wagoner]

On Sat, May 22, 2010 at 11:28 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
 On Sat, 22 May 2010, GlenM wrote:

 Hello Folks;

 I have a dilemma:

 I have a client with Asterisk 1.4x and he needs to have a record of all
 incoming calls - caller ID and date/time is sufficient. Since I am not
 an Asterisk wizard, I am doing it this way.

 I set a cron job to tailf the last 10 lines of the Master.csv file and
 package those nicely in an email. However, I can see some inefficiencies
 in this. Main one is what if there are more than 10 incoming calls
 between cron runs?

 So, questions:

 1. has anyone done this?
 2. is there a better way?
 3. if so, can you 'skool' me ?

 AIUI, Asterisk opens for append the Master.csv file, (fopen (... a))
 which creates the file if it doesn't existis.. writes a line to it then
 closes it for each CDR recorded, so ...

 You can rename the Master.csv file then email the file then delete it...

 Pseudocode:

 Once every 10 miuntes from cron:

   if Master.csv does not exist, then exit      // No calls

   rename Master.csv work.csv
   sleep 1
   process and email work.csv to whoever
   delete work.csv
   exit

 The sleep may not be needed, but it won't do any harm in the event that
 you rename the file after asterisk opens it but before it writes the line
 into and closed it.

 And instead of deleting the work.csv you could append it to some other
 file for a permanent log...

 Gordon

 --

I use asterisk-addons with mysql to store cdr data. I process this
data and insert it into the companies call database link to users, you
could just email it. I basically added a column to mysql and mark each
row as processed.

Ryan

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Re: [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?

[Ryan Wagoner]

On Thu, May 20, 2010 at 11:41 AM, Olivier oza_4...@yahoo.fr wrote:
 Hi,

 I'm evaluating what could keep me from upgrading production systems to
 1.6.2.
 As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
 issue with BLF-pickup which kept me from going further.

 Have you met other issues I should include include in my checklist ?

 Regards

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I have been using 1.6.1.18 after having transfer issues with early
1.6.2 releases and phones becoming unreachable from the below bug.

https://issues.asterisk.org/view.php?id=16936

Luckily the fix for this bug is in 1.6.2.8-rc1. I'll probably just
stay on 1.6.1.18 until 1.8 comes out as it has been the best of the
1.6.x series. The first 1.6.1 releases had major issues with DTMF
detection. Then there was the TCP SIP issues with Exchange UM. I've
found 1.6.1.18 to work all around with only a minor DTMF issue with
Exchange UM that I was able to patch.

Ryan

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[asterisk-users] T.38 Fax With Flowroute SIP Provider

[Ryan Wagoner]

Does anybody have T.38 faxing working with Flowroute? I am running
Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully
receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in
sip.conf. When I receive a fax it tries to negotiate T.38 and
Flowroute sends back a Bad Request response saying I have a SIP syntax
error.

Flowroute support is recommending that I try again after removing
externip and localnet from sip.conf. They state that their service
will recognize the private IP and rewrite the SIP packets. However
this is going to cause issues for my remote SIP phones.

Thanks,
Ryan

DEBUG[32389] app_fax.c: Negotiating T.38 for receive on SIP/flowroute-

INVITE sip:+num...@xx.xx.xx.xx:5060 SIP/2.0
...
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 2048302926 2048302927 IN IP4 xx.xx.xx.xx
s=Asterisk PBX 1.6.2.7-rc3
c=IN IP4 xx.xx.xx.xx
t=0 0
m=image 4575 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPFEC

SIP/2.0 400 Bad Request
...
CSeq: 102 INVITE
Error-Info: sip:+num...@xx.xx.xx.xx;cause=[line 023] SIP syntax error
Content-Length: 0

WARNING[32389] app_fax.c: Transmission error

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Re: [asterisk-users] T.38 Fax With Flowroute SIP Provider

[Ryan Wagoner]

I wasn't sure how the lines were counted. Here is the debug output
from Asterisk where it is building the invite packet. I looked at the
a=T38 lines and nothing is standing out to me.

Ryan

[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  0 [ 47]: INVITE
sip:+num...@x.x.x.x:5060 SIP/2.0
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  1 [ 63]: Via:
SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2837f4cf;rport
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  2 [ 54]: Route:
sip:x.x.x.x;lr,sip:x.x.x.x;lr
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  3 [ 16]: Max-Forwards: 70
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  4 [ 59]: From:
sip:+num...@x.x.x.x:5060;tag=as7d21d6f3
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  5 [ 53]: To:
sip:+num...@x.x.x.x:5060;tag=gK0d4c48f7
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  6 [ 39]: Contact:
sip:num...@x.x.x.x
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  7 [ 39]: Call-ID:
302861516_123483...@x.x.x.x
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  8 [ 16]: CSeq: 102 INVITE
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  9 [ 36]:
User-Agent: Asterisk PBX 1.6.2.7-rc3
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 10 [ 72]: Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 11 [ 26]:
Supported: replaces, timer
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 12 [ 52]:
X-asterisk-Info: SIP re-invite (External RTP bridge)
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 13 [ 29]:
Content-Type: application/sdp
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 14 [ 19]: Content-Length: 293
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 15 [  0]:
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  0 [  3]: v=0
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  1 [ 48]: o=root
2048302926 2048302927 IN IP4 x.x.x.x
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  2 [ 26]:
s=Asterisk PBX 1.6.2.7-rc3
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  3 [ 21]: c=IN IP4 x.x.x.x
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  4 [  5]: t=0 0
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  5 [ 22]: m=image
4575 udptl t38
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  6 [ 17]: a=T38FaxVersion:0
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  7 [ 21]:
a=T38MaxBitRate:14400
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  8 [ 22]:
a=T38FaxFillBitRemoval
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body  9 [ 37]:
a=T38FaxRateManagement:transferredTCF
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body 10 [ 24]:
a=T38FaxMaxDatagram:1400
[May  6 13:29:05] DEBUG[32389] chan_sip.c:Body 11 [ 23]:
a=T38FaxUdpEC:t38UDPFEC


On Thu, May 6, 2010 at 6:54 PM, Kevin P. Fleming kpflem...@digium.com wrote:
 On 05/06/2010 05:46 PM, Ryan Wagoner wrote:
 Does anybody have T.38 faxing working with Flowroute? I am running
 Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully
 receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in
 sip.conf. When I receive a fax it tries to negotiate T.38 and
 Flowroute sends back a Bad Request response saying I have a SIP syntax
 error.

 Flowroute support is recommending that I try again after removing
 externip and localnet from sip.conf. They state that their service
 will recognize the private IP and rewrite the SIP packets. However
 this is going to cause issues for my remote SIP phones.

 Thanks,
 Ryan

 DEBUG[32389] app_fax.c: Negotiating T.38 for receive on 
 SIP/flowroute-

 INVITE sip:+num...@xx.xx.xx.xx:5060 SIP/2.0
 ...
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.2.7-rc3
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 X-asterisk-Info: SIP re-invite (External RTP bridge)
 Content-Type: application/sdp
 Content-Length: 293

 v=0
 o=root 2048302926 2048302927 IN IP4 xx.xx.xx.xx
 s=Asterisk PBX 1.6.2.7-rc3
 c=IN IP4 xx.xx.xx.xx
 t=0 0
 m=image 4575 udptl t38
 a=T38FaxVersion:0
 a=T38MaxBitRate:14400
 a=T38FaxFillBitRemoval
 a=T38FaxRateManagement:transferredTCF
 a=T38FaxMaxDatagram:1400
 a=T38FaxUdpEC:t38UDPFEC

 SIP/2.0 400 Bad Request
 ...
 CSeq: 102 INVITE
 Error-Info: sip:+num...@xx.xx.xx.xx;cause=[line 023] SIP syntax error
 Content-Length: 0

 Which line is 'line 23' of the T.38 re-INVITE?

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

 --
 _
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Re: [asterisk-users] T.38 Fax With Flowroute SIP Provider

[Ryan Wagoner]

On Thu, May 6, 2010 at 7:11 PM, Warren Selby wcse...@selbytech.com wrote:
 On Thu, May 6, 2010 at 5:54 PM, Kevin P. Fleming kpflem...@digium.com
 wrote:

 On 05/06/2010 05:46 PM, Ryan Wagoner wrote:
  Does anybody have T.38 faxing working with Flowroute? I am running
  Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully
  receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in
  sip.conf. When I receive a fax it tries to negotiate T.38 and
  Flowroute sends back a Bad Request response saying I have a SIP syntax
  error.
 
  Flowroute support is recommending that I try again after removing
  externip and localnet from sip.conf. They state that their service
  will recognize the private IP and rewrite the SIP packets. However
  this is going to cause issues for my remote SIP phones.
 

 Last I checked with Flowroute, they weren't yet supporting T.38.  Has this
 changed in the last month or so?

 --
 Thanks,
 --Warren Selby
 http://www.selbytech.com


I found some websites mentioning they supported it. Plus when I
receive a fax with t38 turned off I get the following in the log

WARNING[27824] chan_sip.c: Unsupported SDP media type in offer: image
19738 udptl t38

Ryan

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