Re: [asterisk-users] sipgate outgoing calls

[Jamie A. Stapleton]

Probably worth noting that sipgate will close (at least in the U.S.) on Oct. 
31:  
http://www.besttechie.com/2013/09/13/voip-provider-sipgate-will-close-oct-31/


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten Wemheuer
Sent: Thursday, September 19, 2013 10:54 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] sipgate outgoing calls

Hi,

Am Mittwoch, den 18.09.2013, 14:29 +0100 schrieb
gpxctawjc...@irational.org:
> Hello
> 
> i am trying to setup sipgate gateway
> 
> i can get incoming calls fine, but when i dial in and then try to dial 
> out i get this in asterisk command line

What Sipgate product are You using? At least in Germany there are different 
configurations for the different products necessary. For Sipgate trunking and 
Sipgate team You have to configure an outboundproxy (which differs between both 
products). For Sipgate Basic you don't need an outboundproxy. As far as I 
remember there was an issue with some asterisk versions and the outboundproxy 
for Sipdate team.

Karsten



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Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[Jamie A. Stapleton]

What is your provider seeing?  Many providers send re-INVITEs at 15 minutes.  
Many firewalls have closed their port before this due to UDP timeouts.  I have 
a whitepaper that I wrote on this subject; I will see if I can dig it up.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Florian Wolters
Sent: Thursday, March 21, 2013 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

Hello,

> I solved it by moving Asterisk 1.6 to Asterisk 1.4.
>
> Try asterisk 1.4 or 1.8  on a test box and see how it goes.

I did try the latest 1.8.2x release already without any improvement.
Currently running is a Asterisk PBX 1.8.10.1~dfsg-1ubuntu1 as the tcpdump says 
(little mistake to my last mail).

I also played around with "canreinvite". But regardless of the setting
(yes/no) I still get disconnects after 15 minutes. I just tried to accept 
session-timers, but this has no connection to this issue either.

So I turned on SIP debug for this host and analyszed it with wireshark.
The last packets show an INVITE from my provider, that is answered by my 
Asterisk with "200 OK, with session description". What follows is an ACK by the 
provider and immediately a BYE sent by the provider. So for me it looks like 
the provider is disconnecting the call.

I could not see any reason or hangup cause for this in the dump. Are there 
error messages for this that can be seen in the protocol?

The tcpdump (the last few packets) shows:


--- 8< snip ---

13:37:28.258566 IP (tos 0x0, ttl 64, id 44187, offset 0, flags [DF], proto TCP 
(6), length 611)
172.16.0.2.44929 > 217.0.17.170.5060: Flags [P.], cksum 0xf764 (incorrect 
-> 0xd1be), seq 4568:5139, ack 4057, win 45600, length 571
13:37:28.277390 IP (tos 0xc0, ttl 55, id 4807, offset 0, flags [DF], proto TCP 
(6), length 547)
217.0.17.170.5060 > 172.16.0.2.44929: Flags [P.], cksum 0x2c63 (correct), 
seq 4057:4564, ack 5139, win 65535, length 507
13:37:28.277415 IP (tos 0x0, ttl 64, id 44188, offset 0, flags [DF], proto TCP 
(6), length 40)
172.16.0.2.44929 > 217.0.17.170.5060: Flags [.], cksum 0xf529 (incorrect -> 
0xdc6d), ack 4564, win 45600, length 0
13:37:54.240304 IP (tos 0xc0, ttl 25, id 14090, offset 0, flags [none], proto 
UDP (17), length 1255)
217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 1227
INVITE sip:090066@79.253.136.104:5060 SIP/2.0
Via: SIP/2.0/UDP
217.0.17.170:5060;branch=z9hG4bK354b7c6fb4f56e91ecb6a22cd865b03b.0e83c94b
Via: SIP/2.0/TCP
62.156.82.55:5060;branch=z9hG4bK88ecd3ab295c71dfc9ea9ab6c3134b08jaaiaaj23clqa3Zqkv7akae3e3wetjnxm
Via: SIP/2.0/TCP 62.156.80.48:5083;branch=z9hG4bK2609300495-692831843
Max-Forwards: 70
To: ;tag=as77f2fb84
From: ;tag=8f233b97
Call-ID: 83de2b0c3faf0ef9@217.0.17.170
Contact:
;+g.3gpp.icsi-ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.icsi_ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel"
CSeq: 1939619 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, 
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 297

v=0
o=- 558131575 1701401067 IN IP4 217.0.17.170
s=Phone Call via hiQ9200 SIPCA
c=IN IP4 217.0.1.67
t=0 0
m=audio 16884 RTP/AVP 8 100
b=AS:110
b=RS:1375
b=RR:4125
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sqn: 0
a=sendrecv
a=ptime:20

13:37:54.240497 IP (tos 0xc0, ttl 25, id 14091, offset 0, flags [none], proto 
UDP (17), length 1222)
217.0.17.170.5060 > 172.16.0.2.5060: SIP, length: 1194
INVITE sip:090066@79.253.136.104:5060;transport=TCP SIP/2.0
Via: SIP/2.0/UDP
217.0.17.170:5060;branch=z9hG4bK86e8a99ff6e15f3e72053880758bf877.9856a4e1
Via: SIP/2.0/TCP
62.156.82.55:5060;branch=z9hG4bKb0d91d40f6f840a4ff61c11c2afe12d6jaaiaahr0zo2a3Zqkv7awon0rib4uosfa
Via: SIP/2.0/TCP 62.156.80.48:5082;branch=z9hG4bK2609108269-1839751709
Max-Forwards: 70
To: 090066 ;tag=as09bca4fd
From: ;tag=f18b4044
Call-ID: 248ef1b5553e5756490d6556573a1...@tel.t-online.de
Contact:
;+g.3gpp.icsi-ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.icsi_ref="urn%3Aurn-xxx%3A3gpp-service.ims.icsi.mmtel"
CSeq: 1939639 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, PUBLISH, REFER, 
REGISTER, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 224

v=0
o=- 1028575251 1704720679 IN IP4 217.0.17.170
s=Basic Session
c=IN IP4 217.0.1.81
t=0 0
m=audio 17120 RTP/AVP 8 101

[asterisk-users] Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2

[Jamie A. Stapleton]

After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a 
Segmentation fault.

[root@localhost asterisk-11.1.2]# asterisk -vvc
Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
  == Parsing '/etc/asterisk/logger.conf': Found
  == Manager registered action DBGet
  == Manager registered action DBPut
  == Manager registered action DBDel
  == Manager registered action DBDelTree
  == Registered custom function 'MESSAGE'
  == Registered custom function 'MESSAGE_DATA'
  == Registered application 'MessageSend'
  == Manager registered action MessageSend
  == Manager registered action DataGet
  == Parsing '/etc/asterisk/codecs.conf': Found
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
  == Parsing '/etc/asterisk/dnsmgr.conf': Found
[2013-01-10 14:20:10] ERROR[27062]: config_options.c:512 aco_process_config: 
Unable to load config file 'acl.conf'
  == Parsing '/etc/asterisk/http.conf': Found
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Login
  == Manager registered action Challenge
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action GetConfig
  == Manager registered action GetConfigJSON
  == Manager registered action UpdateConfig
  == Manager registered action CreateConfig
  == Manager registered action ListCategories
  == Manager registered action Redirect
  == Manager registered action Atxfer
  == Manager registered action Originate
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action PresenceState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxStatus
  == Manager registered action MailboxCount
  == Manager registered action ListCommands
  == Manager registered action SendText
  == Manager registered action UserEvent
  == Manager registered action WaitEvent
  == Manager registered action CoreSettings
  == Manager registered action CoreStatus
  == Manager registered action Reload
  == Manager registered action CoreShowChannels
  == Manager registered action ModuleLoad
  == Manager registered action ModuleCheck
  == Manager registered action AOCMessage
  == Manager registered action Filter
  == Registered custom function 'AMI_CLIENT'
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_humbug.conf': Found
[2013-01-10 14:20:10] NOTICE[27062]: manager.c:7545 __init_manager: Invalid 
keyword  =  in manager.conf [general]
  == Parsing '/etc/asterisk/users.conf': Found
  == Parsing '/etc/asterisk/cdr.conf': Found
[2013-01-10 14:20:10] NOTICE[27062]: cdr.c:1613 do_reload: CDR logging 
disabled, data will be lost.
-- CEL logging disabled.
  == Parsing '/etc/asterisk/udptl.conf': Found
[2013-01-10 14:20:10] WARNING[27062]: udptl.c:1413 removed_options_handler: 
t38faxudpec in udptl.conf is no longer supported; use the t38pt_udptl 
configuration option in sip.conf instead.
[2013-01-10 14:20:10] WARNING[27062]: udptl.c:1415 removed_options_handler: 
t38faxmaxdatagram in udptl.conf is no longer supported; value is now supplied 
by T.38 applications.
Asterisk PBX Core Initializing
Registering builtin applications:
  == Registered custom function 'EXCEPTION'
  == Registered custom function 'TESTTIME'
[Answer]
  == Registered application 'Answer'
[BackGround]
  == Registered application 'BackGround'
[Busy]
  == Registered application 'Busy'
[Congestion]
  == Registered application 'Congestion'
[ExecIfTime]
  == Registered application 'ExecIfTime'
[Goto]
  == Registered application 'Goto'
[GotoIf]
  == Registered application 'GotoIf'
[GotoIfTime]
  == Registered application 'GotoIfTime'
[ImportVar]
  == Registered application 'ImportVar'
[Hangup]
  == Registered application 'Hangup'
[Incomplete]
  == Registered application 'Incomplete'
[NoOp]
  == Registered application 'NoOp'
[Proceeding]
  == Registered application 'Proceeding'
[Progress]
  == Registered application 'Progress'
[RaiseException]
  == Registered application 'RaiseException'
[ResetCDR]
  == Registered application 'ResetCDR'
[Ringing]
  == Registered application 'Ringing'
[SayAlpha]
  == Registered application 'SayAlpha'
[SayDigits]
  == Registered application 'SayDigits'
[SayNumber]
  == Registered application 'SayNumber'
[SayPhonetic]
  == Regist

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

[Jamie A. Stapleton]

I am seeing this as well.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy Abshire
Sent: Monday, January 07, 2013 1:22 PM
To: Asterisk Users
Subject: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After 
Pick-up

Outoing calls I make using Motif Google Voice Calls continue ringing even after 
the other end picks up.

I have to restart Asterisk to resolve the issue.
I don't see any errors.

It's not recognizing that the other party picked up the phone and restarting 
Asterisk fixes it only for a day.

--
Co-op Vacation Rentals
www.coopvr.com
15218 Summit Ave
Suite #300-354
Fontana, CA 92336
Phone/Fax (855) 760-COOP (2667)


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Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

[Jamie A. Stapleton]

ADTRAN has some interesting Voice Quality Monitoring built into their switches, 
routers, etc:  http://adtran.com/web/url/vqm

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Wednesday, June 20, 2012 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

Hello,

1) I am wondering what is the best practice to monitor if there are or were 
problems with SIP calls on my Asterisk box. E.g. how about a software that 
extracts all calls from the /var/log/asterisk/full (I have permanently enabled 
verbose 10 and sip debug) log and tells me on which of them were problems? 
Checking the logs manually is very hard, but as SIP is a standardized 
protocoll, there should be tools doing that for you? As an example, a person 
calling me recently got a 488 Not acceptable error as reply from my Asterisk 
box. Nothing came through to my SIP phone, so I didn't know anything about the 
call or the problems (which were on his phone btw). I would like to be informed 
about such cases, know that there was a call to my Asterisk box that made 
problems.

2) How about monitoring speech quality? E.g. sometimes it seems like a packet 
is missing (I then have a short pause during the call), how to monitor such 
things and create statistics out of this data?

So basically I want to monitor my Asterisk installation proactively for 
reliability/problems and (speech) quality.

Thanks for any hints!

Best regards
Stefan
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Re: [asterisk-users] Ongoing attack from 188.138.100.16

[Jamie A. Stapleton]

Block them.  They are one of the Internet's top bad IP addresses.  
http://www.threatstop.com/checkip

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Tuesday, March 06, 2012 7:29 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Ongoing attack from 188.138.100.16

I've been logging sip registrations from this IP address for 2 days now.  I've 
emailed the domain's admin, but nothing seems to come of it.

I've routed him into oblivion, but still, I think 50 requests a second for 2 
days is a bit much.

Any ideas?

-- 

Take care and have fun,
Mike Diehl.

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

[Jamie A. Stapleton]

Snom is an OEM of the Konftel.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of brya...@zktech.com
Sent: Sunday, January 08, 2012 12:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best non polycom SIP conference room phone

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?

Bryant Zimmerman 

On Jan 8, 2012, at 10:59 AM, C F  wrote:

> I find that the bottom line of all polycom haters is ones inability of
> comprehending the config files and not in its quality.
> However check out Panasonic. They make a sip conference phone.
> 
> On 1/5/12, Carlos Alvarez  wrote:
>> On Thu, Jan 5, 2012 at 5:10 PM, C F  wrote:
>> 
>>> On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman 
>>> wrote:
 I am looking for a really good SIP conference room phone for use with
 asterisk. I do not like Polycom at all.
>>> 
>>> You have a really bad taste.
>>> 
>> 
>> There was an interesting flamewar one day in the Asterisk IRC channel over
>> Polycom love/hate.  We fall into the hate category here, and hope to never
>> have to deal with them.  If there was an SPA-series conference phone, we'd
>> all rejoice.
>> 
>> --
>> Carlos Alvarez
>> TelEvolve
>> 602-889-3003
>> 
> 
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Re: [asterisk-users] Best non polycom SIP conference room phone

[Jamie A. Stapleton]

Some ideas:
* http://www.clearone.com/voip-conference-phones.html
* http://www.konftel.com/Products/Konftel300IP
* 
http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_duo.html

We have tested all of these in our lab but I would prefer not to be too verbose 
about my preferences on a mailing list.

Please feel free to call me if you want more detail,
-jamie
(804) 412-1601
sip: ja...@cbsiva.com
Skype:  cbsi_jamie

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of José Pablo Méndez 
Soto
Sent: Thursday, January 05, 2012 1:06 PM
To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best non polycom SIP conference room phone

Hello Bryant,

Have you seen the snom meetingpoint?
http://www.snom.com/en/products/sip-conference-phone/snom-meetingpoint/

I don't own one, but it looks like a fine piece of hardware. And snom is 
manufacturer of supported phones for Microsoft's Lync server (must say 
something their quality right?)

http://technet.microsoft.com/en-us/lync/gg278172.aspx

Doubles Polycom price though...

 José Pablo Méndez


On Thu, Jan 5, 2012 at 11:19 AM, Bryant Zimmerman 
mailto:brya...@zktech.com>> wrote:
I am looking for a really good SIP conference room phone for use with asterisk. 
I do not like Polycom at all. What would you all recommend? I have to be able 
to get them in the US. I found several that looked good but could not get them. 
And yes cost does matter but quality is the most important thing.
Thanks

Bryant

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Re: [asterisk-users] Hint'ing with XMPP?

[Jamie A. Stapleton]

Yes, we are using it.  Most of the docs on the Internet are for 1.4.  However, 
we now have it working with 1.8 (after some work).

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, December 06, 2011 5:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hint'ing with XMPP?


2011/12/5 Jamie A. Stapleton 
mailto:jstaple...@computer-business.com>>
I have not ever done what you are talking about.

However, I can tell you that our Openfire XMPP server has similar functionality 
because of their Asterisk-IM Plugin.
Are you currently using it ?
With which asterisk version ?

From: 
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
 
[mailto:asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>]
 On Behalf Of Jay R. Worthington
Sent: Saturday, December 03, 2011 8:11 AM
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Subject: [asterisk-users] Hint'ing with XMPP?

Hiya,

can i use an XMPP Client to see the presence of a hint? I have configured 
asterisk in component-mode, seem's to work, but all users 
(xmpp:1...@asterisk.dohmain.com<mailto:xmpp%3a...@asterisk.dohmain.com> are 
online, even if 123 isn't a configured hint). Any good howto's out there, all 
the stuff on voip-info.org<http://voip-info.org> is completely outdated, i'm 
using asterisk 10...

Regards

Jay

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Re: [asterisk-users] Hint'ing with XMPP?

[Jamie A. Stapleton]

I have not ever done what you are talking about.

However, I can tell you that our Openfire XMPP server has similar functionality 
because of their Asterisk-IM Plugin.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay R. Worthington
Sent: Saturday, December 03, 2011 8:11 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hint'ing with XMPP?

Hiya,

can i use an XMPP Client to see the presence of a hint? I have configured 
asterisk in component-mode, seem's to work, but all users 
(xmpp:1...@asterisk.dohmain.com are 
online, even if 123 isn't a configured hint). Any good howto's out there, all 
the stuff on voip-info.org is completely outdated, i'm 
using asterisk 10...

Regards

Jay
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Re: [asterisk-users] Best VoIP conferencing phone ?

[Jamie A. Stapleton]

Some ideas:
* http://www.clearone.com/voip-conference-phones.html
* http://www.konftel.com/Products/Konftel300IP
* 
http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_duo.html

We have tested all of these in our lab but I would prefer not to be too verbose 
about my preferences on a mailing list.

Please feel free to call me if you want more detail,
-jamie
(804) 412-1601
sip: ja...@cbsiva.com
Skype:  cbsi_jamie



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, November 30, 2011 3:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best VoIP conferencing phone ?

Hi Faisal,

Thanks for reply but I want hardware wase VoIP device. If know please gussed 
me. From google I fould the list of below devices but I am not sure that these 
are best for used or have an issue 

 1)Polycom SoundStation IP 7000

Why it's best: The Polycom SoundStation IP 7000 is the most advanced conference 
phone from the Polycom SoundStation lineup and leaves little to be desired. 
With an amazing 20' 360 radius, the 7000 is perfect for large conference rooms. 
The new HD voice quality (22 kHz) allows.

2) Polycom Voicestation 500

Why it's a best pick: The Polycom VoiceStation 500 is one of the best 
conference phones for a wide variety of reasons. The VoiceStation 500 features 
amazing call quality, 7' 360 radius, Bluetooth connectivity, wired connection, 
background noise reduction, and an attractive design.

3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S

Why it's a best pick: With a 360 10' radius and 8 microphones, everyone is sure 
to be heard with the Panasonic KX-TS730S. The multiple microphones allows for 
everyone sitting in on the conference to be heard uniformly without distortion.

4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone

Why it's a best pick: The Cisco 7937G works via VoIP connection, has stunning 
call clarity, and features a simplistic but expensive design that is easy to 
use. Cisco is an industry leader in IT communication products, and the 7937G is 
no different. The 360 design allows everyone to be heard.

5)Polycom SoundStation VTX 1000

Why it's a best pick: The SoundStation VTX 1000 is an incredible conference 
phone, but it is very pricey and not as good as advertised. The VTX 1000 is 
designed for large conference rooms and features upgradable software (which is 
a huge benefit since the cost is so high), 20' 360 radius.
6)Polycom(r) SoundStation(r) IP 5000
7) GXP2120 6-line Executive HD IP Phone

On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif 
mailto:fai...@vopium.com>> wrote:
I have tried EyeBeam and it worked fine with x members audio conference however 
it need resources (Processing + RAM) per additional line.

Regards,

Faisal Hanif

From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of virendra bhati
Sent: Wednesday, November 30, 2011 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; 
Sam Govind
Subject: [asterisk-users] Best VoIP conferencing phone ?

Hi ,

I know it's might not the right way to asking such stupid question. But I want 
to take help from experts into VoIP fields so I have to decided to post here.

Please help me which will be the best VoIP conferencing phone which will cover 
10 Persians into conferencing with best audio support ?

--

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer


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Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer

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Re: [asterisk-users] Goggle voice incoming dialplan

[Jamie A. Stapleton]

exten => accou...@gmail.com,1,Answer()
exten => accou...@gmail.com,n,Wait(2)
exten => accou...@gmail.com,n,SendDTMF(1)
exten => accou...@gmail.com,n,Dial(SIP/device1)

exten => accou...@gmail.com,1,Answer()
exten => accou...@gmail.com,n,Wait(2)
exten => accou...@gmail.com,n,SendDTMF(1)
exten => accou...@gmail.com,n,Dial(SIP/device2) 



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk
Sent: Wednesday, June 15, 2011 11:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Goggle voice incoming dialplan


Hi,

I am a question to handle incoming goggle voice. I have put several GV accounts 
into the jabber.conf. How can I direct different accounts to different 
extensions?

Help with example is much appreciate

Thanks,

CK


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Re: [asterisk-users] Asterisk issue or VoIP provider issue ?

[Jamie A. Stapleton]

Many providers do not allow for caller ID name to be sent.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Friday, June 10, 2011 5:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk issue or VoIP provider issue ?

Hi List,

I want to set my caller ID and name with asterisk. So that when I make outgoing 
calls then destination end will see my name with number.

from asterisk end I set both the things into dialplan.
---
--
exten => _X.,n,Set(CALLERID(num)=9172341457)
exten => _X.,n,Set(CALLERID(name)="Virendra Bhati")

But when call reach to destination number then only number is display, name was 
display as unknown

Is this issue of voip provider or Asterisk 1.6.2.18 ?
I contact them they replay me that it's your end issue not my end.


-
Thanks and regards

 Virendra Bhati
+91-9172341457
Asterisk Engineer

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Re: [asterisk-users] SIP/IAX guest access?

[Jamie A. Stapleton]

In IAX, you specify context= inside of a [guest] context.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan Gofferje
Sent: Thursday, June 09, 2011 2:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP/IAX guest access?

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

On 06/09/2011 08:50 PM, Jamie A. Stapleton wrote:
> Guest calls go to the context specified in [general] of sip.conf.

Thx. Is this valid for IAX2 also?

- -S


- -- 
 (o_   Stefan Gofferje| SCLT, MCP, CCSA
 //\   Reg'd Linux User #247167   | VCP #2263
 V_/_  Heckler & Koch - the original point and click interface
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.16 (GNU/Linux)

iEYEARECAAYFAk3xDBEACgkQbQKZlCdPOMMogQCeOX1QWdLQJ9SQGnSHNoh9UGFO
iWkAnjwp4oBhbNdGn+lz0fHb3hokH+/f
=la5a
-END PGP SIGNATURE-


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Re: [asterisk-users] SIP/IAX guest access?

[Jamie A. Stapleton]

Guest calls go to the context specified in [general] of sip.conf.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan Gofferje
Sent: Thursday, June 09, 2011 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP/IAX guest access?

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi, I have a general question about SIP access for nonregistered users.

I would like to make it possible for basically anybody to make a SIP
call to my asterisk without having to have a user account, but in a
specific context. So that e.g. somebody could make a SIP call to
SIP/ste...@my.asterix.pbx and it would go like this:

[incoming_guest]
exten => stefan,1,dial(SIP/300&SIP/301)
exten => stefan,2,voicemail(300,u)

For IAX I created a user [guest] with a specific context
[incoming_guest] in which I handle the calls but I also haven't really
figured out how to create the "stefan@.." solution.
To reach this context, people have to call IAX/gu...@my.asterisk.pbx

How do I create a context in which all calls from nonregistered  clients
are handled?

- -S

- -- 
 (o_   Stefan Gofferje| SCLT, MCP, CCSA
 //\   Reg'd Linux User #247167   | VCP #2263
 V_/_  Heckler & Koch - the original point and click interface
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.16 (GNU/Linux)

iEYEARECAAYFAk3xBXYACgkQbQKZlCdPOMN6qgCfR+TBVpVCSKDZyzUJk6r53VYS
dvYAoJrbb76zqFY3c1K0YzA9j3dowPE4
=2SCj
-END PGP SIGNATURE-


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Re: [asterisk-users] Cannot call to my server with SIP

[Jamie A. Stapleton]

If you want anonymous callers to be able to place calls to Asterisk, you need 
to set allowguest=yes.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Saturday, April 23, 2011 9:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cannot call to my server with SIP

Op 22-04-11 23:49, Jamie A. Stapleton schreef:
> I can see your server just fine...
> 
> -bash-3.2# ./svmap.py xen8.vandervlis.nl
> | SIP Device | User Agent  | Fingerprint |
> --
> | 91.198.178.28:5060 | Asterisk PBX 1.6.2.9-2+squeeze1 | disabled|
> 
> However, if I try to call, Asterisk is saying:
> -- Called p...@vandervlis.nl
> [2011-04-22 17:47:13] NOTICE[10639]: chan_sip.c:19036 handle_response_invite: 
> Failed to authenticate on INVITE to ...;tag=as131f7b6a'

Ah, this is very good information. I see you, but I don't understand why
I don't see myself when I try this. Maybe my sip client (Ekiga) is not OK.

Asterisk log:
[Apr 22 23:46:50] NOTICE[29497] chan_sip.c: Sending fake auth rejection
for device "Jamie A. Stapleton"
;tag=0wqaLzsAyMQwTdfcP2r0mG2FkPBQjEQF

Firewall log:
Apr 22 23:46:50 xen8 kernel: [3824476.043190] FW:IN=eth0 OUT=
MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150
DST=91.198.178.28 LEN=1320 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP
SPT=5060 DPT=5060 LEN=1300
Apr 22 23:46:50 xen8 kernel: [3824476.043556] FW:IN= OUT=eth0
SRC=91.198.178.28 DST=81.23.228.150 LEN=782 TOS=0x00 PREC=0x00 TTL=64
ID=17809 PROTO=UDP SPT=5060 DPT=5060 LEN=762
Apr 22 23:46:50 xen8 kernel: [3824476.048153] FW:IN=eth0 OUT=
MAC=00:16:3e:72:6f:93:00:14:f6:7e:d7:f0:08:00 SRC=81.23.228.150
DST=91.198.178.28 LEN=411 TOS=0x08 PREC=0x00 TTL=61 ID=0 DF PROTO=UDP
SPT=5060 DPT=5060 LEN=391

> What do you have allowguest 
> (http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest) set to?

I was testing security. It's like this:

sip.conf:
---
[general]
context=default
allowguest=no
alwaysauthreject=yes
(...)

[guests]
context=default
allowguest=yes

[trunk]
context=dialout
(...)

[phone-paul]
context=dialout
(...)

[phone-ann]
context=dialout
(...)
---

extensions.conf:
-
[default]
include => users

[dialout]
include => users
exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT)

[users]
exten=>6001,1,Dial(SIP/paul,20)
exten=>6002,1,Dial(SIP/ann,20)
(...)


Thanks for your help!

With regards,
Paul van der Vlis.




-- 
http://www.vandervlis.nl/



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Re: [asterisk-users] Cannot call to my server with SIP

[Jamie A. Stapleton]

I can see your server just fine...

-bash-3.2# ./svmap.py xen8.vandervlis.nl
| SIP Device | User Agent  | Fingerprint |
--
| 91.198.178.28:5060 | Asterisk PBX 1.6.2.9-2+squeeze1 | disabled|

However, if I try to call, Asterisk is saying:
-- Called p...@vandervlis.nl
[2011-04-22 17:47:13] NOTICE[10639]: chan_sip.c:19036 handle_response_invite: 
Failed to authenticate on INVITE to ...;tag=as131f7b6a'

What do you have allowguest 
(http://www.voip-info.org/wiki/view/Asterisk+sip+allowguest) set to?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Friday, April 22, 2011 11:03 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Cannot call to my server with SIP

Hello,

I cannot call my server over the internet with SIP anymore.

Even when I do a maximum logging on my firewall, I don't see packets
coming from outside. I've tried it from an ekiga.net account and an
sip2sip.info account. What could be wrong?  I would expect incoming
traffic on port 5060 UDP...

The account is "p...@vandervlis.nl". This should connect trought DNS to
the machine xen8.vandervlis.nl:
---
paul@server2:~$ host -t SRV _sip._udp.vandervlis.nl
_sip._udp.vandervlis.nl has SRV record 0 5 5060 xen8.vandervlis.nl.
---

Is here maybe somebody with an idea, or a way to debug this?
Maybe with a nice Linux commandline tool?

With regards,
Paul van der Vlis.




-- 
http://www.vandervlis.nl/



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Re: [asterisk-users] Can gtalk.conf work with multiple GoogleVoice numbers?

[Jamie A. Stapleton]

No problem.  You just specify accountn...@gmail.com.

exten => accountn...@gmail.com,1,Answer()
exten => accountn...@gmail.com,n,Wait(2)
exten => accountn...@gmail.com,n,SendDTMF(1)
exten => accountn...@gmail.com,n,Dial(SIP/devicename)

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle
Sent: Friday, April 01, 2011 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Can gtalk.conf work with multiple GoogleVoice numbers?

Hello.  I would like to configure Asterisk to accept incoming calls from two 
different GoogleVoice numbers via gtalk and jabber.  I'm running Asterisk 
1.8.3.2 and I can get one number working just fine.  However, I can't figure 
out how to modify the gtalk.conf file shown on the Asterisk wiki site to work 
with two different jabber profiles.  Do all incoming GoogleVoice calls have to 
go through the [guest] context in gtalk.conf?  If so, it seems that would limit 
you to working with only one GoogleVoice number.  My configs basically match 
what's at the wiki site here:

https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google

Any advice would be appreciated.  Thanks!

-- 
Chris

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Re: [asterisk-users] Issues with Digum Repos / AsteriskNOW & Bad Packages

[Jamie A. Stapleton]

Have you read page 312 of Asterisk: The Future of Telephony 
(http://cdn.oreilly.com/books/9780596510480.pdf)?

"there are a few things that need to be
added in order to get it to function. First off, Asterisk needs to have an IMAP 
client
installed so that it can communicate with the IMAP server. Pretty much any IMAP
server works (even Exchange Server), and the authors have personally tested IMAP
voicemail support with both the Courier-IMAP and Dovecot IMAP servers. The IMAP
server may be on the same physical machine as the Asterisk installation, or it 
may be
on the other side of the globe. To be able to access the IMAP server, Asterisk 
requires
an IMAP client library. This library is the University of Washington's free 
IMAP client,
named c-client. To install the c-client you simply need to navigate to your 
/usr/src directory
and run the following commands:
# wget ftp://ftp.cac.washington.edu/mail/imap.tar.Z
This downloads the source code. Extract it with:
# tar zxvf imap.tar.Z"

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Joakimsen
Sent: Wednesday, March 23, 2011 11:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Issues with Digum Repos / AsteriskNOW & Bad Packages

I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
voicemail storage and Asterisk 1.4.

After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI
I run the yum package manager and replace voicemail with imap
voicemail and attempt to start Asterisk, however the voicemail module
is not loaded:

[Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module:
Error loading module 'app_voicemail_imapstorage.so':
/usr/lib/libc-client.so.1: undefined symbol: mm_dlog
[Mar 23 23:30:03] WARNING[12592]: loader.c:777 load_resource: Module
'app_voicemail_imapstorage.so' could not be loaded.

Is there some way to have this working?


-- 
Med Vennlig Hilsen,

A. Helge Joakimsen

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Re: [asterisk-users] fail2ban + asterisk

[Jamie A. Stapleton]

iptables -L -v

will give you the IP address that was banned

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell
Sent: Monday, March 07, 2011 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] fail2ban + asterisk

On Sat, Mar 5, 2011 at 8:54 PM, Pezhman Lali  wrote:
> Dear
> this note is only for fresh administrators don't think about asterisk
> security.


Do you know where you go to 'un-ban' an IP if they made some mistake?

Using webmin I was not able to find the IP address that was was banned.

Thanks,
Matt

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Re: [asterisk-users] SIP Provider Recommendation in US

[Jamie A. Stapleton]

We have had very good results with nexVortex.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent A. Torrenga
Sent: Thursday, March 03, 2011 11:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Provider Recommendation in US

I am becoming frustrated with our current VOIP provider.  Does anyone have any 
suggestions for a provider that supports asterisk well and provides solid 
service?  Voip-info.org has a husge list of providers, but it is impossible to 
tell the fly-by-night operations from the reputable providers.
--Brent
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Re: [asterisk-users] asterisk security....again

[Jamie A. Stapleton]

http://sipera.com/ is one such product.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Monday, February 28, 2011 9:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk securityagain

Thanks Mr. Kevin.

Can anyone please also tell me which firewall is best suited for asterisk/sip 
attack prevention. Is there any firewall built specially to address sip 
security problems?
On Mon, Feb 28, 2011 at 6:38 PM, Kevin P. Fleming 
mailto:kpflem...@digium.com>> wrote:
On 02/28/2011 07:27 AM, Rizwan Hisham wrote:
Any suggestions on encrypting the sip and rtp. I have done some googling
on it. looks like it is not supported by most end point devices or
service providers. But still your thoughts will be appreciated on this
subject.

You cannot protect a remote SIP endpoint from attacks via your server; that SIP 
endpoint is an endpoint itself, and if it can receive IP packets from 
attackers, it will process them. These packets don't go through your server, 
and encrypting the legitimate traffic between your server and the remote 
endpoint isn't going to make any difference at all.

The *only* way to address attacks like this is to modify the configuration of 
the remote endpoint to ignore all incoming packets that aren't from your 
server(s). Even that is not a perfect solution, though, because the attacker 
(if they are actually aware of your server and customers) can spoof the IP 
addresses of your server(s) in order to get the remote endpoints to at least 
accept an INVITE (they can't place a successful call through them using 
spoofing though).

--
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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--
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com

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Re: [asterisk-users] Call sip:u...@domain.com?

[Jamie A. Stapleton]

Just add something like this to your dialplan:

exten=>1234,1,Dial(SIP/u...@domain.com)

Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Thursday, December 16, 2010 5:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call sip:u...@domain.com?

Hello

At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT
set up with a VOSP trunk that I can use to make/receive calls to/from
the PSTN.

Now, I'd like to be able to call any number on the Net that is
advertised as "sip:u...@domain.com", such as those:

www.voip-info.org/wiki/view/Phone+Numbers

Do I need to register a second trunk (FWD, etc.) through which those
calls will be made? Can't my VOSP perform both tasks (landline +
Internet calls)? Can I just let my Asterisk server connect to the
remote SIP server through the SRV DNS record and have it dial the
extension?

Any example appreciated, thank you.


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Re: [asterisk-users] Asterisk 1.6.2.10 & video

[Jamie A. Stapleton]

1.  Per http://www.voip-info.org/wiki/view/Asterisk+video: Asterisk does not 
provide any video transcoding capabilities
2.  You can turn off video support on a peer like this:
disallow=h261 
disallow=h263 
disallow=h263p 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Monday, December 13, 2010 3:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6.2.10 & video

Hello,

1. is it possible that Asterisk does not translate between codecs H263 and H264 
?

2 If I set videosupport=yes in sip.conf [general], can I turn off the video 
support on a peer ?



Kind regards,
Jonas.

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Re: [asterisk-users] Elementary question - accessing feature codes from cell phone

[Jamie A. Stapleton]

We use DISA (http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA) to access 
our entire [features] context from our cell phones.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal
Sent: Friday, November 05, 2010 11:11 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Elementary question - accessing feature codes from 
cell phone

Hi, please forgive me for this (hopefully) simple question. I cannot seem to 
find an answer or solution while searching around.

I want to be able to call in to my server using my cell phone and be able to 
set call forwarding for my extension and enter a phone number and also be able 
to call in to that extension and disable the call forwarding. I see I can do 
this through the ARI web interface but wish to do this with my cell phone. 
Currently, it seems I can only get into my voicemail and attempting to run 
feature codes like *72 don't get recognized.

I currently have a DID assigned to my extension if this helps. I am not sure if 
it is required.

Thanks much.
jellydog

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[asterisk-users] FW: Under heavy attack

[Jamie A. Stapleton]

Only 100?  We had a single server over 300.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria
Sent: Saturday, October 30, 2010 9:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Under heavy attack

My count has reached 100 for the day. The server serves doesn't serve 
international calls anyways, I wonder how would it benefit any hacker in any 
way.

--
Zeeshan

Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak 
mailto:jmas...@antelope.net>> wrote:
No.  It seems that opening up some sort of automatic blocking could cause an 
attacker forging packets to block legitimate endpoints. It also seems like they 
won't get in with good passwords, so it isn't actually accomplishing something 
to worry about the script kiddies if you have good passwords.  And this 
blocking won't actually stop someone with a zero day attack or who is 
sophisticated and can attack from many IP addresses - these are the real 
threats for people with good passwords.

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Re: [asterisk-users] SIP 800 Origination/Termination - International

[Jamie A. Stapleton]

nexVortex (http://bit.ly/9bEw9e) can do this.  They use Global for TF.  They 
can support both US and CA origination.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Freeman
Sent: Tuesday, September 14, 2010 10:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 800 Origination/Termination - International

Anyone have a good provider for International (US/Canada at least) 800 
termination/origination? I have a customer that had us port one of their 
800 numbers and apparently didn't realize that they had published that 
number in Canada as well. Our current origination/termination provider 
can't handle Canadian inbound calls to that number, so I need to find 
another provider that can.

Thanks-
Joe

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[asterisk-users] Cisco 7975g running 8.3.4

[Jamie A. Stapleton]

Have a Cisco 7975g running SIP firmware version 8.3.4.  Many things are broken 
with Asterisk.

1) BLF doesn't work
2) MWI doesn't work
3) Sometimes the calls get "stuck" on the display
4) Sometimes MOH works
5) Headset jack doesn't work

Can anyone recommend a version of the SIP firmware for the Cisco 7975g that 
they know works well with Asterisk?


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[asterisk-users] ITSP with DDIs (or DIDs) from India

[Jamie A. Stapleton]

Anyone know of an ITSP that can offer DDIs (or DIDs) from India?
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Re: [asterisk-users] Monitor asterisk

[Jamie A. Stapleton]

Might be worth your time to check out:  http://www.humbuglabs.org/

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu
Sent: Saturday, August 07, 2010 3:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Monitor asterisk


Hey guys,

I have my asterisk box running without a gui. I now need to monitor usage, 
calls, traffic of voice calls on this asterisk server. I cannot now install a 
gui because the configs will be wiped out, how can i go about monitoring all 
the above?

--
Richard Zulu
Managing Director
Time Information Company
P.O Box 31842
Clock Tower
Kampala, Uganda
www.time.co.ug

Mobile :+256752624006
Skype: zulu.richard

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Re: [asterisk-users] 488 Not Acceptable Here

[Jamie A. Stapleton]

A packet capture would be most useful.  Then, you could review your SDP with 
your provider.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Beak
Sent: Friday, July 23, 2010 7:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 488 Not Acceptable Here

Hi,

I'm having real difficulty in getting calls to go through with 
Asterisk.  I've managed to check that my SIP connection is made to my 
provider.  Below is an email I received from them:

snipsnipsnip
I am not certain of the reason for rejection but it has to do with the 
SDP,  it does not seem to be a codec issue, the response is as you have 
seen:

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 
192.168.0.14;received=172.28.20.106;branch=z9hG4bK42d2ea03;rport=60017
From: "Andy" ;tag=as5c784926
To: ;tag=SD24jn898-4C46B8A2-5688CB2-0ADE2C09
Call-ID: 32d506cd3489aa81031937f467ef6...@192.168.0.14
CSeq: 102 INVITE
Reason: Q.850 ;cause=127 ;text="Interworking, unspecified"
Content-Length: 0

There looks to be a non-standard element in your SDP that is not 
supported by any of the networks.
snipsnipsnip

Which configuration file is possibly incorrect in this scenario?

What dumps are likely to be useful to me?

Thanks,
  Andy

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Re: [asterisk-users] How to change the IP in the SIP contact header

[Jamie A. Stapleton]

Have you tried setting

externip=

In the [general] of your sip.conf?

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman
Sent: Monday, July 05, 2010 1:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to change the IP in the SIP contact header

Hello,

I'm trying to use a SIP trunk service and the provider ask me to have the IP 
address of the contact header as my public IP and not as my private one, how 
can I do it?

See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is 
my public address:

sipINVITE sip:144@ a.b.c.d SIP/2.0
Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport
Max-Forwards: 70
From: "Polycom" ;tag=as7435100b
To: 
Contact: 
Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 05 Jul 2010 15:49:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 1812163927 1812163927 IN IP4 10.100.101.107
s=Asterisk PBX 1.6.1.20
c=IN IP4 10.100.101.107
t=0 0
m=audio 18848 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


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Re: [asterisk-users] Brute force attacks

[Jamie A. Stapleton]

The IP 69.175.35.186 has just been banned by Fail2Ban after 293 attempts 
against our server.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Timms
Sent: Thursday, July 01, 2010 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Brute force attacks

On Thu, Jul 1, 2010 at 9:16 AM, Ishfaq Malik 
mailto:i...@pack-net.co.uk>> wrote:
Hi

We've just noticed attempts (close to 20 attempts, sequential peer numbers) 
at guessing peers on 2 of out servers and thought I'd share the originating IPs 
with the list in case anyone wants to firewall them as we have done

109.170.106.59
112.142.55.18
124.157.161.67

Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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We have noticed the same sort of activity on our server. The originating IP 
addresses attempting access were:

204.9.204.145 (hosted at U.S. Colo, I believe)
91.203.132.149 (Nephax)
130.70.157.186 (University of Louisiana)
61.160.121.46 (Chinanet)
109.170.0.10 (ReasonUP Ltd)

--
John Timms
IT Department - Gnoso Inc.
j...@gnoso.com
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Re: [asterisk-users] Problems for Skype for Asterisk

[Jamie A. Stapleton]

We are running Asterisk 1.6.2.7-rc1 and SfA without problem.  What version are 
you running?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Tuesday, April 27, 2010 9:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problems for Skype for Asterisk

Is there an issue with running it with the latest from the 1.6.2 branch?
I did an svn update and make install and now when somebody comes in via
Skype, I get an infinite loop of:

[Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: 
Invalid argument
[Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: 
Invalid argument
[Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: 
Invalid argument
[Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: 
Invalid argument

and there's no voice path in either direction.

If there's an issue, what's the latest svn revision number I can use?

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Re: [asterisk-users] [Conference] Audio/Video

[Jamie A. Stapleton]

http://www.projectdiastar.org/ looks promising...

On Apr 14, 2010, at 7:04 PM, Stéphane Bauland wrote:

Le 04/15/2010 12:11 AM, Hans Witvliet a écrit :
On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote:
Hi guys,

I'm planning of creating a speech/video conference application. This
application will provide a system to see/listen to each personn present
in the conference.

So each ppl will have a audio and video stream.

I'm wondering if you know a way to do this with asterisk or if it's
supported ?

If it is, i'm asking you about some documentation or related article (if
you know ones) where i could find more informations.

Else, do you know any other way to do this ?

Best regards,


Would love to see a _working_ video conf.
afaicr it's currently vapor-ware
Are you thinking of letting asterisk doing video multiplexing?
Or are you aiming just for a conference with a small number of
participants?

hw


We (cause we are a team) are planning of doing a multi user conference
software at a end school project.

The way we go is, we are looking throught jungle (xmpp ext for jabber)
to create conference between many people. We don't want to set a
limitation about "how many participant of a conference).

But right now, i'm discovering asterisk, and i need some informations
from people like you that know the soft and his capatibilities...

So i think yes, we want to do video multiplexing.

Do you think a software like that could use asterisk as a backend ?

And, do you know any other software that is doing the same thing using
asterisk ?

--
Stéphane Bauland

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Jamie A. Stapleton
CBSi - Connecting your problems with solutions.
Telephone:  (804) 412-1601
Facsimile:  (804) 412-1611
VideoConf:  callto:jstapleton.computer-business.com

Meet me on LinkedIn<http://www.linkedin.com/in/jstapleton>

Have I exceeded your expectations?  Please share your experience with our 
Founder, Fred W. Brumbaugh<mailto:fbrumba...@computer-business.com>

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Re: [asterisk-users] Which H.323 to use in Ast 1.6

[Jamie A. Stapleton]

Which Avaya system are you running?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Wednesday, February 24, 2010 5:52 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] Which H.323 to use in Ast 1.6

Could you share your config for the Asterisk and Avaya side too?  Thanks 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A.
Stapleton
Sent: Wednesday, February 24, 2010 3:37 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Which H.323 to use in Ast 1.6

I have always used ooh323 between Avaya and Asterisk.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Tuesday, February 23, 2010 2:24 PM
To: 'Asterisk Users List'
Subject: [asterisk-users] Which H.323 to use in Ast 1.6

We're doing a project that requires H.323 to an Avaya.  Does anyone have
experience to share on which H.323 driver to use in asterisk 1.6?  Is the
diference between h323 and ooh323 still worth the extra effort?  (We've only
installed h323 under 1.4)
 
If you have setup/config experience with this setup in Asterisk 1.6 please
share! Thanks,
 
MD

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Re: [asterisk-users] Which H.323 to use in Ast 1.6

[Jamie A. Stapleton]

I have always used ooh323 between Avaya and Asterisk.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Tuesday, February 23, 2010 2:24 PM
To: 'Asterisk Users List'
Subject: [asterisk-users] Which H.323 to use in Ast 1.6

We're doing a project that requires H.323 to an Avaya.  Does anyone have 
experience to share on which H.323 driver to use in asterisk 1.6?  Is the 
diference between h323 and ooh323 still worth the extra effort?  (We've only 
installed h323 under 1.4)
 
If you have setup/config experience with this setup in Asterisk 1.6 please 
share! Thanks,
 
MD

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Re: [asterisk-users] SIP tunnel

[Jamie A. Stapleton]

Have you considered using IAX instead of SIP?  IAX2 is a VoIP protocol that 
carries both signaling and media on the same port: 
http://en.wikipedia.org/wiki/Inter-Asterisk_eXchange

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mosbah.abdelkader
Sent: Thursday, February 11, 2010 8:37 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP tunnel

Hello,



I have the following situation: A firewall is blocking all SIP and RTP traffic 
in the side of some of my clients. My clients cannot change settings of the 
firewall.



I need to solve this problem and I need some help from you.



I have this idea: implement a SIP user agent which does not use well known SIP 
ports (uses http port 80 for example) and use other ports that are not blocked 
by the firewall for RTP (FTP, https, ssh, ...ports). Then, configure Asterisk 
to use the same ports to interact with the client.



Is this idea feasible? if not what are the problems? please give me your 
opinions about the situation?



Thank you.
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Re: [asterisk-users] Connecting to an External EPBX without an SIP provider

[Jamie A. Stapleton]

This all depends on your EPBX...  For example

1)  If you put a 2 port FXO card in an Asterisk server, you need 2 FXS ports on 
your EPBX to connect to
2)  If you put a 4 port FXO card in an Asterisk server, you need 4 FXS ports on 
your EPBX to connect to
3)  If you put a T1/E1 card in an Asterisk server, you need a matching T1/E1 
port on your EPBX to connect to

Don't hesitate to email me directly if you still have questions...

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siju George
Sent: Wednesday, January 27, 2010 11:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Connecting to an External EPBX without an SIP 
provider

Thanks for the reply jamie :-)

Does ordinary EPBXs in US have those ports or do you need special EPBXs?

--Siju

On Wed, Jan 27, 2010 at 8:32 PM, Jamie A. Stapleton
 wrote:
> In this case, a SIP provider would not be required.
>
> Obviously, you will need ports on your EPBX to connect the Digium card to.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siju George
> Sent: Wednesday, January 27, 2010 5:01 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Connecting to an External EPBX without an SIP 
> provider
>
> Hi,
>
> If I get a Dignum Card and fit it into my computer do I still need an
> SIP provider to connect through my EPBX to a Public Telephone System?
>
> Thanks
>
> --Siju
>
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Re: [asterisk-users] Connecting to an External EPBX without an SIP provider

[Jamie A. Stapleton]

In this case, a SIP provider would not be required.

Obviously, you will need ports on your EPBX to connect the Digium card to.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siju George
Sent: Wednesday, January 27, 2010 5:01 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Connecting to an External EPBX without an SIP provider

Hi,

If I get a Dignum Card and fit it into my computer do I still need an
SIP provider to connect through my EPBX to a Public Telephone System?

Thanks

--Siju

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Re: [asterisk-users] Really Silly Question From Total Newbie

[Jamie A. Stapleton]

Could use the free http://www.sipgate.com/one for some testing (assuming that 
Asterisk is connected to the Internet)

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT
Sent: Tuesday, January 05, 2010 2:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Really Silly Question From Total Newbie

Hello All -

I've been poking around the past few weeks, trying to familiarize
myself with all of this.  I am new to Linux, VoIP and Asterisk -- to
be complete.   This is my first exposure to all of these technologies.

I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge
2400 and the install went well.   I can log in and poke around in
Linux and I even configured the box to be recognized on my windows
network.  However, is there a GUI that I can access to help me set
things up?  I've gotten so far as what looks to me like "DOS" windows
that I can change various things in the OS...

I do not have any other hardware installed.  No cards and no VoIP
phones.   I havent got to the point where I can make a test call or
anything like that.  I dont know how to tell if Asterisk is up and
running and how I can tweak it, etc.   I've been reading a lot of
different things, and have become a bit confused. I think that in time
it will come to me but I needed to stop and ask because I need to know
if I am on the wrong path for what I'd like to do someday

My main question is: CAN I make call from that box to my cell phone
using a soft-phone?   If so, how can I do that?   Also, can I use my
cell phone to call into that box?   I dont know if I have to get a
phone number, or do I NEED a phone number?   At the moment, I do not
have any dollars to throw at this project.   Its purely for learning,
proof of concept sort of thing for myself on my spare time in the
evenings.  I would simply like to be able to call out and be able to
call into that box.  Later on down the road maybe I will get into
setting up an IVR using a database so I can call into that system from
wherever and get information read back to me.  But, first things
first  I'd like to know if I am heading down the wrong path here.

Sorry for what might seem as really silly questions, but I am not sure
how to proceed.

Thanks in advance for any insight that you folks can provide!

Mike

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Re: [asterisk-users] dahdi restart kills server

[Jamie A. Stapleton]

you have to stop asterisk before restarting dahdi service

On Dec 8, 2009, at 7:06 PM, Mike wrote:

I`ve just experience a dead server, because I ran /etc/init.d/dahdi restart .  
I had to reboot the server.

Should I worry about something not being right in my install, or is there a 
known problem with doing this while Asterisk is running?

I expected DAHDI channels to die, but not the whole server!

Mike
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Jamie A. Stapleton
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