Re: [asterisk-users] Removing "Parsing /etc/asterisk/manager.conf" from CLI

[Adrian A]

There is an OpenSER proxy in front of Asterisk which handles the clients.
The script is called by OpenSER whenever a client sends a SUBSCRIBE request
for MWI. It uses php to connect to Asterisk like so:
fsockopen("$mhost","5038", $errno, $errstr, 5) and gets the user's voicemail
counts.

I'm not sure how I would maintain this as a persistent connection that would
live if I restart Asterisk. I'd have to detect that somehow.

Adrian

On Thu, Apr 10, 2008 at 12:14 AM, Stefan Reuter <[EMAIL PROTECTED]>
wrote:

> Adrian A wrote:
> > Is there any way of removing this line from showing on the console? I
> > have a script that logs in every few seconds to manager (...)
>
> Maybe a better solution is to rethink your architecture. The Manager API
> is well suited for long running connections, so there is no need to
> reconnect every few seconds.
>
> =Stefan
>
> --
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> Neusser Str. 110
> 50760 Koeln
> Germany
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Re: [asterisk-users] Removing "Parsing /etc/asterisk/manager.conf" from CLI

[Adrian A]

Anything more than 'core set verbose 1' produces this message, however
verbose 1 does not display much of anything.

On Thu, Apr 10, 2008 at 1:53 AM, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:

> On Wed, Apr 09, 2008 at 11:55:13PM -0700, Adrian A wrote:
> > Hello,
> >
> > Is there any way of removing this line from showing on the console? I
> have a
> > script that logs in every few seconds to manager and it makes the CLI
> output
> > very hard to follow because of the "  == Parsing
> > '/etc/asterisk/manager.conf': Found". (Yes, Found! manager.conf was
> there 3
> > seconds ago, guess what it's still there.)
>
> What verbosity level do you use?
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
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[asterisk-users] Removing "Parsing /etc/asterisk/manager.conf" from CLI

[Adrian A]

Hello,

Is there any way of removing this line from showing on the console? I have a
script that logs in every few seconds to manager and it makes the CLI output
very hard to follow because of the "  == Parsing
'/etc/asterisk/manager.conf': Found". (Yes, Found! manager.conf was there 3
seconds ago, guess what it's still there.)

There is a very old feature request about this at
http://bugs.digium.com/view.php?id=3085 but I cannot see the resolution.
Mantis shows "APPLICATION ERROR #801" at the end of the page...

Adrian
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Re: [asterisk-users] RTCP not being sent when on hold

[Adrian A]

The RTP codec 126 is a bogus RTP packet sent by Bria to maintain the NAT
binding.

I've identified the issue as this:

Bria has an inactivity timer that is based on RTCP. Basically, if during the
call there is RTCP, Bria uses it to make sure the call is still alive.
Asterisk does send RTCP when call is active, but it stops when call is put
on hold by Bria. The default timeout for Bria is 30 seconds, thus it
disconnects the call because it has not received any RTP or RTCP during this
time.

I am not sure at this point which is correct implementation. Should the
client not rely on RTP/RTCP when it's on hold or should Asterisk send some
sort of keep alive RTP/RTCP when it knows one of the clients is on hold?


On Wed, Apr 9, 2008 at 7:15 AM, Steve Langstaff <[EMAIL PROTECTED]>
wrote:

>  It would be interesting to see a wireshark trace of the SIP and RTP
> traffic during call setup and hold, to see:
> a) what codec 126 has been negotiated as and
> b) who is sourcing the unknown RTP datagram.
>
>  --
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Adrian A
> *Sent:* 09 April 2008 00:55
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] RTCP not being sent when on hold
>
> Hello,
>
> When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I
> place the call on hold, the call is dropped after 30 seconds.
> It looks like there is no RTCP/RTP sent to the client from Asterisk while
> on hold (music on hold playing to caller) thus client disconnects the call.
> During this time, I get the following messages in the CLI:
>
> NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0'
>
> In sip.conf I have rtpkeepalive=15 but that does not seem to help.
>
> Does anyone know what I can do to fix this, other than increase the
> timeout on Bria?
>
> Thanks,
> Adrian
>
>
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[asterisk-users] RTCP not being sent when on hold

[Adrian A]

Hello,

When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I
place the call on hold, the call is dropped after 30 seconds.
It looks like there is no RTCP/RTP sent to the client from Asterisk while on
hold (music on hold playing to caller) thus client disconnects the call.
During this time, I get the following messages in the CLI:

NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0'

In sip.conf I have rtpkeepalive=15 but that does not seem to help.

Does anyone know what I can do to fix this, other than increase the timeout
on Bria?

Thanks,
Adrian
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[asterisk-users] Disable SIP notify for peers

[Adrian A]

Hello,

I am using OpenSER together with Asterisk.
I have the users registered to OpenSER and have added peer definitions for
each user so that the NOTIFY for MWI is sent to user when voicemail is left
in their respective mailbox. That works great so far in terms of voicemail
integration. On the OpenSER I have a script being executed for when
message-summary SUBSCRIBE's are received which uses the manager interface
for Asterisk to retrieve message counts and send them using sipsak.

The one thing which I would like to change is that when I do a 'reload' or
restart Asterisk, a NOTIFY is sent to each peer. When I have around 200 of
these, Asterisk tries to send 200 NOTIFY messages at once which seems to
sometimes lock it up and it also probably overloads the network
unnecessarily. Is there any way to disable these NOTIFY's? I only want it to
send them to the user when a voicemail is left.

Thanks,
Adrian
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[asterisk-users] Voicemail marking messages as Old

[Adrian A]

It seems to me that simply listening to a new voicemail message will move
the message to the Old folder, without any other user interaction. I'm
working on a voicemail callback queue script and I have wrongly assumed that
messages remain in INBOX unless the user actually saves or deletes them.
I'm running an older 1.2 version of Asterisk.
Is anyone able to confirm the same behavior in newer versions? Is there a
way for Asterisk voicemail to behave like regular voicemail where a message
remains "New" until the caller does something to it (other than simply
listening to it) ?

Thanks.
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[Asterisk-Users] eyeBeam 1.5

[Adrian A]

I thought some of you may be interested.  CounterPath has released a new version of the SIP softphone eyeBeam on Friday as a free upgrade to users of 1.1.  Interface looks mostly the same but has more features such as QoS and MS Outlook integration. 
http://support.counterpath.com/viewtopic.php?t=6182&sid=1b74dd310d55c40862cbe37ed640047c
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[Asterisk-Users] Vmail.cgi and #include

[Adrian A]

Is there a way to have the web voicemail script parse and execute #include statements from voicemail.conf?  I prefer to keep my default context entries in a different file for easier parsing by other scripts, but vmail.cgi
 cannot seem to handle that, even with %s/voicemail.conf/myfile.ext/g.Thanks.
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Re: [Asterisk-Users] Asterisk with Vonage

[Adrian A]

Works great with these settings:[vonage]type=peersecret=passwordusername=phone numberhost=sphone.vopr.vonage.netport=5061dtmfmode=rfc2833fromuser=phone number
fromdomain=sphone.vopr.vonage.netcanreinvite=nocontext=vonage_incominginsecure=veryOn 3/29/06, 
Steve Jones <[EMAIL PROTECTED]> wrote:













I know Vonage doesn't officially have a "bring
your own device" type program, but they do offer a softphone.  Has anyone
gotten Asterisk to connect directly to Vonage?  This would be a great help!!







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Re: [Asterisk-Users] SIP/Video client for PocketPC that works with Asterisk?

[Adrian A]

CounterPath PocketPC softphone..On 3/9/06, Joe <[EMAIL PROTECTED]> wrote:
Anyone know of a SIP client that does video and runs on the PocketPC?I've tried Microsoft Portrait but have had issues.Thanks!___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] Asterisk native sounds now available!

[Adrian A]

If I understand this correctly, this sounds package is a subset of the Asterisk sounds package.  Can I just copy the native sounds (eg. ulaw) in the existing sounds directory and Asterisk will automatically use them instead of the default gsm ones?  How does Asterisk pick which one to play, does it know about the .ulaw extension?
>Doug,It looks like you have installed asterisk-sounds.  asterisk-sounds is
not included in the Asterisk Native Sounds Package.  That is a separatecollection of prompts arranged by John Todd and contributed to thecommunity.  I have already talked with him about that.Other people have brought this up too.  Basically, I'll consider
re-doing (and paying for) the sounds in asterisk-sounds based on thedonations I receive for what is provided so far in the Native AsteriskSounds package.--Kristian Kielhofner___
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[Asterisk-Users] SetCallerID and CDR

[Adrian A]

Hi,I am forcing caller ID to be sent to our VoIP provider using the SetCallerID app:exten => _91.,1,SetCallerPres(allowed)exten => _91.,2,SetCallerID("Company Name" <5>)
exten => _91.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])Ever since I started doing this however, the CDR gets overwritten with this new value for the originating caller.  I can no longer see who is the extension on my system that made the call.  Is there any way to record the caller ID of the original caller?
Thanks.
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[Asterisk-Users] Disable music on hold per user

[Adrian A]

Hi all,I've been having complaints from some users that when they are in a conference call and then need to take another call, music on hold is played into the conference.  Is there a way to disable music on hold for cases such as these?
Thanks,Adrian
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[Asterisk-Users] safe_asterisk not working?

[Adrian A]

I've been experiencing some crashes in Asterisk in the past few weeks.  I haven't been able to find out why as gdb shows it's in a different function every time.  But, in the meantime, I've been using safe_asterisk hoping that it would simply restart Asterisk by itself.  It doesn't seem to do that.  Whenever Asterisk crashed, the list of processes doesn't show asterisk or safe_asterisk running anymore. I do not get an e-mail notification and the core dumped is in the standard format 
e.g. core.18875 not core.`hostname`-`date -Iseconds`.  Does anyone know what I could do to troubleshoot this since I can't really force a crash to see what the script is doing?  I have the following variables set in safe_asterisk:
#!/bin/shCLIARGS="$*"    # Grab any args passed to safe_asteriskTTY=9   # TTY (if you want one) for Asterisk to run onCONSOLE=no  # Whether or not you want a console
NOTIFY=[EMAIL PROTECTED]   # Who to notify about crashesDUMPDROP=/tmpI also noticed the line "elif [ $EXITSTATUS -gt 128 ];"inside the script yet my cores show:
"Core was generated by `asterisk -vvvg'.Program terminated with signal 11, Segmentation fault."so perhaps that is why the restart is not triggered?
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Re: [Asterisk-Users] Re: setting up asterisk to handle incoming SIP URI

[Adrian A]

You do need to have a _sip._udp SRV record in DNS for somedomain.com pointing to your Asterisk host.On 1/11/06, Henry Junior
 <[EMAIL PROTECTED]> wrote:I would like to setup my Asterisk server to process an incoming SIP
URI and redirect all requests to a specific context.Example:(1) using a sip phone I'd like to be able to call: sip:somedomain.com*or* 
sip:[EMAIL PROTECTED](2) i'd like my asterisk server to answer the call and route it tothe context=in-from-sipclient which would play thru some DP actionsCan anyone give me the lowdown on what I need to do to set this up in
extensions.conf and sip.conf?I have my softphones working fine but the asterisk server doesn'tappear to handle SIP URIs.Is it necessary to also modify a DNS record to make this work??Thanks,
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Re: [Asterisk-Users] SIP - SIP bridge dropping calls?

[Adrian A]

I was able to get a full debug report (packet dump and asterisk debug) for one of these dropped calls and it does seem to be the provider that is at fault.  I can see that they stop sending RTP packets to Asterisk when this happens and after a while they send a BYE.  I will keep investigating though.
On 12/22/05, David C. Nicosia <[EMAIL PROTECTED]> wrote:













In addition to having this with my SIP phones, I have also
experienced it with SCCP.

 

It started when I updated to the 1.2 release of asterisk. At
the time I updated I also switched VoIP providers and thought it was them.

 

Did you file this as a bug or find a solution to it? Thanks!







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[Asterisk-Users] Re: Weird rtpmap issue

[Adrian A]

Upon further investigation I found that this only happens when the client sends out Speex only.  For example:Client sends:a=fmtp:101 0-15a=rtpmap:97 speex/8000a=rtpmap:101 telephone-event/8000a=sendrecv
Server replies:a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:97 iLBC/8000a=rtpmap:97 speex/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16Can anyone else confirm this?  Just leave speex as the only enabled codec in your client (I use Eyebeam) and try an echo test or anything to Asterisk.
On 12/21/05, Adrian A <[EMAIL PROTECTED]> wrote:
I have the following in sip.conf:disallow=all    ; First disallow all codecsallow=ulaw  ; Allow codecs in order of preferenceallow=alawallow=gsmallow=ilbcallow=speex
All SIP messages that come out of Asterisk have the following in the SDP:m=audio 19092 RTP/AVP 0 8 3 97 97 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:97 iLBC/8000
a=rtpmap:97 speex/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16Does anyone have any idea why speex and iLBC have the same rtpmap number???  This is an Asterisk 1.2.1 installation.


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[Asterisk-Users] Weird rtpmap issue

[Adrian A]

I have the following in sip.conf:disallow=all    ; First disallow all codecsallow=ulaw  ; Allow codecs in order of preferenceallow=alawallow=gsmallow=ilbcallow=speex
All SIP messages that come out of Asterisk have the following in the SDP:m=audio 19092 RTP/AVP 0 8 3 97 97 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:97 iLBC/8000
a=rtpmap:97 speex/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16Does anyone have any idea why speex and iLBC have the same rtpmap number???  This is an Asterisk 1.2.1 installation.
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[Asterisk-Users] SIP - SIP bridge dropping calls?

[Adrian A]

Asterisk 1.2.1 installation.  It seems that calls are being dropped for no valid reason, completely random, in the middle of the call.  I first thought that it was either the network or the VoIP provider dropping packets and confusing Asterisk into hanging up the call.  
However I happened to be running rtp debug at one time this happened and here's the log:Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <>:12006 (type 0, seq 27353, ts 8114560, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <>:36420 (type 0, seq 54972, ts 11318620, len 160)Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <>:12006 (type 0, seq 27354, ts 8114720, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <>:36420 (type 0, seq 54973, ts 11318780, len 160)Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <>:12006 (type 0, seq 27355, ts 8114880, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <>:36420 (type 0, seq 54974, ts 11318940, len 160)Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <>:12006 (type 0, seq 27356, ts 8115040, len 160)
Dec 19 19:38:29 VERBOSE[7114] logger.c: Got RTP packet from <>:36420 (type 0, seq 54975, ts 11319100, len 160)Dec 19 19:38:29 VERBOSE[7114] logger.c: Sent RTP packet to <>:12006 (type 0, seq 27357, ts 8115200, len 160)
Dec 19 19:38:29 DEBUG[7114] channel.c: Didn't get a frame from channel: SIP/298-6427Dec 19 19:38:29 DEBUG[7114] channel.c: Bridge stops bridging channels SIP/298-6427 and SIP/provider-81d5Dec 19 19:38:29 DEBUG[7114] chan_sip.c: update_call_counter(1x) - decrement call limit counter
Dec 19 19:38:29 DEBUG[7114] app_dial.c: Exiting with DIALSTATUS=ANSWER.Dec 19 19:38:29 VERBOSE[7114] logger.c:   == Spawn extension (default, 91, 1) exited non-zero on 'SIP/298-6427'Asterisk complains that it did not get a frame from the channel...but everything seems to be OK...unless there is another type of frame that it's expecting.  The channel that Asterisk thinks it "did not receive a frame" from is then left with dead air.
All messages that I've seen on the list regarding the "Didn't get a frame from channel" error seem to refer to failure in call setup (ie. call dropped as soon as it is answered), however in this case it happens *during* the call, while the two parties are speaking (so not a silence timeout issue).  
Again, this is completely random and it's hard to get a packet dump of the whole thing..Does anyone have any idea why this is happening before I file this as a bug? 

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[Asterisk-Users] asterisk.conf question

[Adrian A]

Hi,
Does anyone know what exactly the option transmit_silence_during_record
in asterisk.conf does?  Is this useful for voicemail recording? 
Providers such as Broadvoice and Vonage hang up the channel after 30
seconds because they think the other party is not there when someone
records a voicemail.  I've been using rtpkeepalive=15 in sip.conf
to overcome this problem but I am wondering if this asterisk.conf
option better serves this purpose?
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[Asterisk-Users] Send SIP NOTIFY frequency

[Adrian A]

Am I missing a parameter somewhere or does a frequency (eg 5 mins) for
sending NOTIFY when there is voicemail does not exist?  The
clients are registered on SER so if they are offline when the voicemail
comes in, they miss the notify message with no chance of getting it
again except if I reload Asterisk.
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[Asterisk-Users] Asterisk with Vonage problems

[Adrian A]

Does anyone currently use Vonage with Asterisk?  I've tried to set
it up but it looks like Asterisk (at least the version that I have)
does not handle well the SIP call dialog, sending a BYE with the wrong
tag.  As a result, when I hang up, Vonage sends back a 400 Bad
Request and the call on the PSTN side does not hang up.  
I know that Vonage does a lot of nasty stuff which impacts UA's but
Xten Eyebeam handles it correctly at least.  I have tried
pedantic=yes but no difference.
Here is the sip.conf and the BYE dialog with numbers replaced:
 
[vonage]
type=friend
secret=
username=
host=sphone.vopr.vonage.net
dtmfmode=rfc2833
port=5061
fromuser=
fromdomain=sphone.vopr.vonage.net
canreinvite=no
context=context
insecure=very


BYE sip:(PSTN Number)@216.115.20.171:5060 SIP/2.0
Via: SIP/2.0/UDP (Asterisk IP):5070;branch=z9hG4bK1dc3ea2d
Route: 216.115.20.171:5060>
From: "Adrian" sphone.vopr.vonage.net>;tag=as74d54cec
To: sphone.vopr.vonage.net:5061>;tag=2067764114
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 104 BYE
Proxy-Authorization: Digest username="(Vonage No)",
realm="216.115.25.198", algorithm=MD5, uri="sip:216.115.25.198",
nonce="18861432149", response="5de1aaac0fa9db87sdfb074a1fe324b",
opaque=""
Content-Length: 0


---
  == Spawn extension (default, 8(PSTN Number), 3) exited non-zero on 'SIP/370-29aa'
Destroying call '[EMAIL PROTECTED]'

<-- SIP read from 216.115.25.198:5061: 
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP (Asterisk IP):5070;branch=z9hG4bK1dc3ea2d
From: "Adrian" sphone.vopr.vonage.net>;tag=as74d54cec
To: sphone.vopr.vonage.net:5061>;tag=2067764114
Call-ID: [EMAIL PROTECTED]
CSeq: 104 BYE
Max-Forwards: 15
Content-Length: 0

The issue I think is that Asterisk uses the "To" tag from the 183
Session Progress instead of the tag from the 200 OK that Vonage sends.
If anyone uses Vonage with Asterisk and it works fine for you (ie.
landline hangs up when you hang up), can you please let me know which
version you're using? (I'm using CVS HEAD from a couple of months ago
and would like to know if an upgrade may fix the issue.)

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[Asterisk-Users] Option 1 in IVR menu

[Adrian A]

Hi all,
I'm trying to setup a simple IVR menu in a context in extensions.conf.  So far, I have:
extension s for playing back the menu
# to repeat it
* for directory
0 for operator
1 which goes to another context:  exten => 1,1,GoTo(option_1,s,1)

Here is what I have in extensions.conf:

[incoming]

; main greeintg
exten => s, 1, Ringing
exten => s, 2, Wait(10)
exten => s, 3, NoOp()
exten => s, 4, Answer
exten => s, 5, Playback(silence/1)
exten => s, 6, Background(a2)
exten => s, 7, Background(b)
exten => s, 8, WaitExten(20)
exten => s, 9, Hangup

; repeats the message
exten => #,1,Goto(s,6)

;Operator
exten => 0,1,Macro(stdexten,302)

; Dial extension
exten => 1,1,Goto(option_1,s,1)  ; this one is delayed
exten => 2,1,Goto(option_1,s,1)  ; this one is executed immediately

;Directory
exten => *,1,Directory(default)

exten => t,1,Goto(s7)  
exten => i,1,Playback(invalid) 


The problem is that when user presses 1, there's a delay before the
action Goto is executed (I'm assuming delay is caused by some
timeout).  All other options work fine, if I use 2 or any other
number instead
of 1 for that action, it also works fine - the Goto gets executed
immediately.
I do get the standard "NOTICE[11563]: rtp.c:281 process_rfc3389: Comfort
noise support incomplete in Asterisk (RFC 3389).  Please turn off on
client if possible." when user presses a key.
So what is so special about sending DTMF key 1 and how can I get around it?

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Re: [Asterisk-Users] Call forwarding

[Adrian A]

Turns out it should actually be:
exten => s,1,AGI(agi-script.agi|arg1|arg2)

On 7/26/05, Adrian A <[EMAIL PROTECTED]> wrote:
> Thanks, that actually helps a lot.
> One problem I have (kind of unrelated) is with the AGI script
> requiring two arguments.  You have:
> exten => s,1,AGI(forward-get.agi,internal,${MACRO_EXTEN})
> On my Asterisk installation that somehow passes the two arguments
> internal and ${MACRO_EXTEN} as one argument to the bash script causing
> the blank check for ${exten} to exit the script.  I have even tried
> other suggestions such as:
> exten => s,1,AGI(forward-get.agi|internal&${MACRO_EXTEN}) or
> exten => s,1,AGI,forward-get.agi,internal ${MACRO_EXTEN}
> I'm running a recent version of CVS HEAD.
> Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686
> running Linux on 2005-07-07 18:42:16
> 
> 
> 
> On 7/25/05, Cullin J. Wible <[EMAIL PROTECTED]> wrote:
> > 1) You could use asterisk realtime and a mysql database.
> >
> > 2) You could use an asterisk database and allow users to set call forwarding
> > by calling an extension.
> >
> > 3) You could write some scripts to use an external database (what we did)
> > and either allow users to update their forwarding options via a web page or
> > telephone.
> >
> > I have attached some simple shell AGI-scripts and parts of our dial-plan so
> > you can see how it all works. We authenticate against the mysql voicemail
> > database and then our standard extension macro checks the database, possibly
> > adding another channel to the dial command.
> >
> > I hope this helps.
> >
> > Cullin
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
> > Sent: Monday, July 25, 2005 4:15 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] Call forwarding
> >
> > Is there an easy way to allow the users to go to a webpage or dial an
> > extension and enter a phone number that their extension can be
> > forwarded to?
> > I'm using SER+Asterisk so doing this in sip.conf for example would not
> > work since all users are registered to SER.  Currently in
> > extensions.conf I have:
> > exten => s,2,Dial(SIP/[EMAIL PROTECTED],20)
> > Is there a way to check that the user at ${ARG1} has setup forwarding
> > and retrieve the forwarding destination?
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
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Re: [Asterisk-Users] Call forwarding

[Adrian A]

Thanks, that actually helps a lot.  
One problem I have (kind of unrelated) is with the AGI script
requiring two arguments.  You have:
exten => s,1,AGI(forward-get.agi,internal,${MACRO_EXTEN})
On my Asterisk installation that somehow passes the two arguments
internal and ${MACRO_EXTEN} as one argument to the bash script causing
the blank check for ${exten} to exit the script.  I have even tried
other suggestions such as:
exten => s,1,AGI(forward-get.agi|internal&${MACRO_EXTEN}) or
exten => s,1,AGI,forward-get.agi,internal ${MACRO_EXTEN}
I'm running a recent version of CVS HEAD.
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686
running Linux on 2005-07-07 18:42:16



On 7/25/05, Cullin J. Wible <[EMAIL PROTECTED]> wrote:
> 1) You could use asterisk realtime and a mysql database.
> 
> 2) You could use an asterisk database and allow users to set call forwarding
> by calling an extension.
> 
> 3) You could write some scripts to use an external database (what we did)
> and either allow users to update their forwarding options via a web page or
> telephone.
> 
> I have attached some simple shell AGI-scripts and parts of our dial-plan so
> you can see how it all works. We authenticate against the mysql voicemail
> database and then our standard extension macro checks the database, possibly
> adding another channel to the dial command.
> 
> I hope this helps.
> 
> Cullin
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
> Sent: Monday, July 25, 2005 4:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Call forwarding
> 
> Is there an easy way to allow the users to go to a webpage or dial an
> extension and enter a phone number that their extension can be
> forwarded to?
> I'm using SER+Asterisk so doing this in sip.conf for example would not
> work since all users are registered to SER.  Currently in
> extensions.conf I have:
> exten => s,2,Dial(SIP/[EMAIL PROTECTED],20)
> Is there a way to check that the user at ${ARG1} has setup forwarding
> and retrieve the forwarding destination?
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> 
>
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[Asterisk-Users] Call forwarding

[Adrian A]

Is there an easy way to allow the users to go to a webpage or dial an
extension and enter a phone number that their extension can be
forwarded to?
I'm using SER+Asterisk so doing this in sip.conf for example would not
work since all users are registered to SER.  Currently in
extensions.conf I have:
exten => s,2,Dial(SIP/[EMAIL PROTECTED],20)
Is there a way to check that the user at ${ARG1} has setup forwarding
and retrieve the forwarding destination?
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Re: [Asterisk-Users] Soft Phone

[Adrian A]

X-Lite is available for Linux as well.

On 7/25/05, Alex Ongena <[EMAIL PROTECTED]> wrote:
> Any recommendation for Linux environments (without WINE) ?
> Thanks
> Alex
> 
> On Mon, 2005-07-25 at 10:04 -0400, Kanuri, Seshu (Company IT) wrote:
> > Firefly Third Party version beats all others.
> >
> >
> > __
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Dave
> > Morrow
> > Sent: Friday, July 22, 2005 4:12 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] Soft Phone
> >
> >
> >
> > Can anyone recommend a good soft phone that's easy to configure under
> > Asterisk and works well on a typical Windows XP system?
> >
> > David A. Morrow
> > Technical Systems Lead
> > Autodata Solutions Company
> > [EMAIL PROTECTED]
> > http://www.autodata.net
> > Tel: (519) 951-6079
> > Fax: (519) 451-6615
> >
> > < Poor planning on your part does not necessarily constitute an
> > emergency on my part! >
> >
> > This message has originated from Autodata Solutions. The attached
> > material is the Confidential and Proprietary Information of Autodata
> > Solutions. This email and any files transmitted with it are
> > confidential and intended solely for the use of the individual or
> > entity to whom they are addressed. If you have received this email in
> > error please delete this message and notify the Autodata system
> > administrator at [EMAIL PROTECTED]
> > 
> >
> >
> >
> > __
> > NOTICE: If received in error, please destroy and notify sender.
> > Sender does not waive confidentiality or privilege, and use is
> > prohibited.
> >
> >
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> 
> --
> NEW: aXs GUARD hands-on Trainings v.7.0
> more info at http://www.axsguard.com/indextraining.htm
> 
> aXs GUARD has completed security and anti-virus checks on this e-mail
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[Asterisk-Users] Re: [Serusers] NAT considerations...

[Adrian A]

You will also need your SIP clients that are behind the same NAT to
support ICE (Interactive Connectivty Establishment) if you want calls
between them.  Xten Eyebeam and Snom phones are the only ones I'm
aware of that support it.

On 7/5/05, Ricardo Martinez <[EMAIL PROTECTED]> wrote:
> And even worst.
> There are some kind of NAT that STUN does not work.
> You can check the mailing list i think some people call it "crap nat".
> Regards,
> 
> Ricardo Martinez.-
> 
> > -Mensaje original-
> > De: Andres [mailto:[EMAIL PROTECTED]
> > Enviado el: Martes, 05 de Julio de 2005 17:17
> > Para: Giovanni Balasso
> > CC: [EMAIL PROTECTED]
> > Asunto: Re: [Serusers] NAT considerations...
> >
> >
> > Giovanni Balasso wrote:
> >
> > >Just some thoughts based on my experience...
> > >After months trying to make everything work using
> > rtpproxy-mediaproxy with
> > >almost everything accomplished but video, I tried to switch
> > to stun solution.
> > >All my problems are gone now, I have audio, video, presence
> > and instant
> > >messages working like a charm. And most important media
> > server doesn't flow
> > >thru my server so network load remains very low. I have been
> > testing for some
> > >days now and I'm quite happy since I still have to stumble
> > on major problems.
> > >Now some considerations... On a poll onsip.org STUN usage is
> > very low and
> > >rtpproxy-mediaproxy rule as NAT trasversal solution. Why
> > don't people use
> > >stun? Has it some major drawbacks I still haven't found?
> > What are main
> > >advantages of rtpproxy-mediaproxy solutions?
> > >I'm really curious to know serusers opinions about this issue.
> > >
> > >thank you all for your two cents ;)
> > >
> > >
> > >
> > STUN does not work if your NAT is Symmetric.  For example all
> > Linux NATs
> > or routers with Linux OS like the Linksys ones.  Unless you have full
> > control on what type of NAT your customer will deploy, it
> > will be very
> > hard to stick to an all STUN solution.
> >
> > --
> >
> > Andres
> > Network Admin
> > http://www.telesip.net
> >
> >
> > ___
> > Serusers mailing list
> > [EMAIL PROTECTED]
> > http://mail.iptel.org/mailman/listinfo/serusers
> >
> 
> ___
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>
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Re: [Asterisk-Users] Gizmo: Skype done right?

[Adrian A]

I have a Gizmo account working perfectly in my Xten Eyebeam, so there
should be no problem using it for Asterisk.  You already have the
username (1747...etc) and your password, the proxy is
proxy01.sipphone.com (or you can sniff packets to see where SIP
messages are being sent to).

On 6/30/05, Robert Webb <[EMAIL PROTECTED]> wrote:
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of hank
> > Sent: Thursday, June 30, 2005 6:49 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Gizmo: Skype done right?
> >
> > they claim to have a windows download but I can't get the program.
> > also they give no instructions on how to get it connected to asterisk
> 
> Which brings us to the question... Why is this being said to be good for
> Asterisk?? I did download it and load it on my computer. But there are
> NO options for connecting to anything or anyone else but a "Gizmo"
> account.
> 
> So just how is this good for the open source VoIP community and
> Asterisk??
> 
> Robert
> 
> 
> 
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Re: [Asterisk-Users] Sipura 3000 help

[Adrian A]

Right, turns out I am an idiot and I do have Asterisk running on 5070
instead of 5061.  It's all working.
Now, if I could find out why calls coming from PSTN have horrible
voice quality

On 6/16/05, Luki <[EMAIL PROTECTED]> wrote:
> > I can see on tcpdump traces that the Invite packets
> > do go to through to the asterisk machine on port 5061,
> > but it's not picking them up.  sip debug does not show
> > any packets either.
> 
> That would imply that the Sipura config is fine, but your Asterisk
> setup is not listening at the right interface, IP or port. If the
> extension was invalid or there was an authentication issue, Asterisk
> should send a reply message not just ignore it...
> 
> --Luki
>
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[Asterisk-Users] Sipura 3000 help

[Adrian A]

Anyone know what I need to do to get the FXO port on the SPA 3000 to
forward calls to Asterisk?  My Asterisk is running on port 5061 and I
set the dial plan on the device to forward to [EMAIL PROTECTED]:5061 but
Asterisk is not picking it up.  I can see on tcpdump traces that the
Invite packets do go to through to the asterisk machine on port 5061,
but it's not picking them up.  sip debug does not show any packets
either.
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[Asterisk-Users] Inbound provider in Canada

[Adrian A]

Does anyone have any recommendations for a SIP/IAX provider I can use
for inbound callls?  The plan is to have a 1800 number people can call
and reach my Asterisk.  The only provider I'm familiar with is Vonage,
but they don't really like the idea of Asterisk according to their
terms of service and they would probably notice if there are for
example 3 calls coming in at the same time.  They also provide 604
area code which I'm looking for though.
I'm thinking of just getting a analog gateway and connecting the local
telco provider, but that wouldn't really give me any cost savings.
Basically I need a service provider that will allow me to have some
number of incoming calls through the same number at the same time and
covers BC, Canada.  Let me know if I'm even asking the right question,
this is a little confusing to me.
Thanks,
Adrian
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[Asterisk-Users] FXO Gateway recommendation

[Adrian A]

>From your experience, would you recommend purchasing 8 Sipura 3000 1
port FXO gateways or 1 Audiocodes 8 port FXO gateway?
The way I see it, the advantage of going to the Sipura solution is
that it is more scalable (ie. I would only need maybe 5 in the
beginning and then add one by one as the needs grow) and seems to be
cheaper: ~$800 for 8 Sipura's versus $1300 for 1 Audiocodes.
The disadvantage is of course having to manage multiple devices and
extra space/power requirements.
Any thoughts?
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Re: [Asterisk-Users] a simple call to my girlfriend

[Adrian A]

X-Lite is available for Linux.

http://www.xten.com/index.php?menu=products&smenu=download


On 6/2/05, Hendrik Wouters <[EMAIL PROTECTED]> wrote:
> Hi,
> 
> Some background:
> 
> I would like to call my girlfriend over the internet. We are both behind a nat
> router and I want to avoid portmapping.
> I've heard that you can call someone behind a firewall (nat router) with the
> IAX protocol, but I'm not sure.
> 
> The questions:
> 
> Do I have to set up my own PBX asterisk server or are there any other (free)
> servers where I can register on and connect to?
> 
> Which is the best (linux) client to call someone with IAX?
> 
> Thanks in advance
> 
> Greetings Hendrik
> 
> P.S. I don't want to use skype (not open standard, it still doens't work well
> in Linux and eats al the time of my old laptop CPU).
> 
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Re: [Asterisk-Users] FXO Gateways

[Adrian A]

I am in Canada and telephone company is Telus. I did also try with a
Vonage ATA plugged into the Mediatrix but Caller ID still does not go
through.  The caller's number is nowhere to be found in the SIP
message sent from Mediatrix to Asterisk.

On 5/18/05, Calin Serbanescu <[EMAIL PROTECTED]> wrote:
> which country are you in and what is your provider ?
> 
> On Wed, 2005-05-18 at 12:25 -0700, Adrian A wrote:
> > Does anyone have any experience with the Audiocodes MP-108 FXO
> > gateway?  I'm looking to get one for incoming PSTN lines.
> >
> > In particular, does it pass caller ID information to Asterisk?
> >
> > I currently have a Mediatrix 1204 but Caller ID does not work, even
> > though the specs say it does.  All it sends are the names of the ports
> > set up internally on the gateway (ie. "pstnline1" etc) when a call
> > comes in.
> >
> > Thanks.
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Re: [Asterisk-Users] Polycom Instant Messaging

[Adrian A]

MSN Messenger does not support SIP, Windows Messenger does.  There's a
difference between the two.

On 5/18/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> Since the Polycom Instant Messaging features uses MSN Messenger, I
> doubt it will work with Asterisk.
> 
> C F wrote:
> 
> > Asterisk can with the sendtext cmd which is available in CVS-HEAD.
> >
> > On 5/18/05, Chris Coulthurst <[EMAIL PROTECTED]> wrote:
> >
> >>
> >>
> >>
> >>Can anyone explain the Polycom Text Messaging features built in to the IP
> >>500/600?   Can Asterisk (or something else) talk to it?  I've seen vague
> >>references to MSN Messenger, and somehow that's mentally disturbing…
> 
> 
> --
> Always do right. This will gratify some people and astonish the rest.
> Mark Twain
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[Asterisk-Users] FXO Gateways

[Adrian A]

Does anyone have any experience with the Audiocodes MP-108 FXO
gateway?  I'm looking to get one for incoming PSTN lines.

In particular, does it pass caller ID information to Asterisk?  

I currently have a Mediatrix 1204 but Caller ID does not work, even
though the specs say it does.  All it sends are the names of the ports
set up internally on the gateway (ie. "pstnline1" etc) when a call
comes in.

Thanks.
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Re: [Asterisk-Users] Polycom Instant Messaging

[Adrian A]

It is based on SIP so it can communicate with other SIP devices which
have instant messages capabilities such as Xten's Eyebeam or Windows
Messenger (5.1 not MSN).  I do not think it will work with Asterisk
since Asterisk does not support IM as far as I know.  It would work
with SER just fine though.

On 5/18/05, Chris Coulthurst <[EMAIL PROTECTED]> wrote:
>  
>  
> 
> Can anyone explain the Polycom Text Messaging features built in to the IP
> 500/600?   Can Asterisk (or something else) talk to it?  I've seen vague
> references to MSN Messenger, and somehow that's mentally disturbing… 
> 
>   
> 
> Chris Coulthurst 
> 
> [EMAIL PROTECTED] 
> 
>   
> 
>   
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Re: [Asterisk-Users] In/out calls from/to same sip provider

[Adrian A]

If you have a [provider] peer in sip.conf, what happens if you use a
register command in sip.conf such as:

register => user:[EMAIL PROTECTED]/provider

and in extensions.conf you have:

Dial(SIP/${EXTEN:[EMAIL PROTECTED])


On 5/13/05, Pizco Dominguez <[EMAIL PROTECTED]> wrote:
> Hi.
> 
> I'm new to asterisk and, one way or the other, I manage to get it working
> for me.
> 
> But I'm having a hard time getting calls going to and coming from the
> same provider, since the definition of the peer in sip.conf seems to be
> different AND not compatible for incoming and outgoing call.
> 
> Outgoing calls need a "secret" and "username" definition in the peer
> context of sip.conf, while incoming ones will have nothing to do with
> those fields.
> 
> So I can have incoming or outgoing calls regarding one provider, but not
> both.
> 
> I've also tried the sample sintax
> 
> "exten =>_42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)"
> 
> that comes with the distribution (debian-sarge), but only to get asterisk
> unable to create sip channel because
> 
> "host [EMAIL PROTECTED] doesn't exist". The address
> is that of the provider.
> 
> voip.org and asteriskdocs.org seems to lead me nowhere.
> 
> I must be missing something obvious, but can't figure out what it is.
> 
> Anybody?
> 
> Thanks.
> 
> --
> Pizco Dominguez
> --
> 
> --
> GPGKEY: gpg --keyserver pgp.rediris.es --recv-key 8DE37A4D
> FINGERPRINT:85CB 4323 F322 5837 EDB5  2033 6FB2 C326 8DE3 7A4D
> --
> 
> --
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Re: [Asterisk-Users] * Server

[Adrian A]

It's mentioned in the description that:
"The VoIP Connection™ Asterisk Voice Server combines the functionality
of a PBX, SIP proxy, Voice Mail server, and more."
As far as I know, Asterisk does not act as a proxy. Does it have SER
included or is it just a confusion of terms?
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[Asterisk-Users] SER Asterisk and NAT

[Adrian A]

I have been trying to setup Asterisk in combination with SER on the
same box as a PBX with SIP clients.  I would like to have it available
for both external and internal users so I have the box setup with
external and internal IP address.  I am running into all kinds of
troubles with this configuration, specifically with forwarding
voicemail to Asterisk from SER.

Does anyone have a similar setup that is working?
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