[asterisk-users] test delivery for lists.digium.com
Testing the new lists.digium.com server. Apologies for the email noise. Digium IT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why such high latency on internal lan?
It depends on the type of sip end point, and how long it takes to respond to a SIP event. For example if I connect a Cisco 7960 IP phone to my Asterisk server over the LAN, I always see registration times of over 100ms. But if I connect X-Lite I get registration times of under 10ms. Asterisk connected as a SIP client, under 2ms. The higher latency with the Cisco's don't seem to effect performance at all. -- Brad > To: asterisk-users@lists.digium.com > From: seandar...@gmail.com > Date: Sat, 23 Oct 2010 12:31:58 -0400 > Subject: [asterisk-users] Why such high latency on internal lan? > > My internal lan is small, 100mb, all wired. aastra phones. > > sip show peers > ... > 142/... 10.10.10.42 D A 5060 OK (136 ms) > 144/... 10.10.10.44 D A 5060 OK (138 ms) > 145/... 10.10.10.45 D A 5060 OK (133 ms) > > But pings are < 1ms: > > ping 10.10.10.42 > > rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms > > Why are the sip latencies so high? And is it a problem? And if so, how > do I fix it? > > FWIW, latencies to outside providers over nat are close to ping: > > jnctn/ 5060 OK (7 ms) > teliax/... N 5060 OK (7 ms) > > ping > > rtt min/avg/max/mdev = 3.471/4.120/4.466/0.288 ms > > > sean > > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RealTime Voicemail
I am using Asterisk 1.4.36 with Realtime Voicemail from a MySQL database, and whilst I have it all working, I am unable to find a way to customize the content of the email that gets sent to a user when they receive a voicemail. In the past I just edited it in the voicemail.conf file and made the customizations in there, but now that I am using Realtime voicemail from MySQL, my voicemail.conf file has to be an empty file. So does anyone know how it would be possible for me to customize the content of the email, other than hacking the source? Cheers, Brad Hughes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk, Skype For SIP
I have been using SiSky Enterprise Edition to integrate Skype with asterisk. You can even call saved skype users from your asterisk system, by creating speed dials in SiSky. Unfortunately it is not a free product but it is very reasonable. Thank you, Brad Finberg - Original Message - From: Alejandro Imass To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Date: Sunday, July 18 2010 8:57 AM Subject: Re: [asterisk-users] Skype for Asterisk, Skype For SIP On Sun, Jul 18, 2010 at 7:48 AM, Vieri wrote: > Hi, > > I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 > things: > > 1) allow any Asterisk SIP extension to call any Skype "user". I do not need > to call landlines via Skype. > I think this is _explicitly_ not supported in the Skype for SIP docs. > 2) allow Internet Skype "users" to call my Asterisk PBX Skype "user" and > route the call to a specific Asterisk SIP extension. > Here is how it goes from my experience with Skype: each SIP channel will cost you about $5 a month, regardless if you have a landline number with them or not. Your account will be assigned a special Skype number 99x . With that number a Skype user can call you and it will be free. You _cannot_ call Skype users from your PBX, as I stated above, this is an explicit no-no in the docs. If you want to make calls from your PBX to landlines you have to buy Skype credit just like you would with a regular skype client. If you want land-lines to call your PBX you need to purchase a skype number which about $60 a year. > At first, I thought it would be simple and free. However, correct me if I'm > wrong but the Skype "user" I can use within the Asterisk PBX cannot be the > "standard type" (used by eg. desktop Skype applications) but needs to be > created by the Skype User Manager for Business Solutions. I believe this has > a price although Skype For SIP Open Beta seems to be free until Q4 2010. I think you can associate existing skype users to your Business Solutions manager but I still don't understand exactly how or why this is useful, and I don't think it has to do with you being able to call any of them from your PBX. Then again I haven't paid much attention to that and perhaps you have more insight into this. > > Has anyone found a way to make "pure Internet user-to-user" Skype/SIP calls > via Asterisk (no PSTN involved) for free? As I said above, once you have purchased your SIP channel you can make free calls to your PBX using the special number but it's only INBOUND AFAIK. Best, Alejandro Imass > > Thanks, > > Vieri > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rename External Directory
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey Guys, Saw your article at www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx and had a question regarding the directory on a 7960 POS3-08-6 not running call manager. I quickly figured out each directory only holds 32 spots and need to implement an A-M and N-Z but the phone labels the directories as "External Directory" I also realized changing the abc or the abc does nothing... Is there a way to change the "External Directory" to CompanyName A-M etc...? Thanks, Brad -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (Darwin) iEYEARECAAYFAkws5SIACgkQhzJ5NSeNtkheSACcD2FFN3eIV0+sfZnBrJWrN10l 3v8An0IoJHVh/2chw+bcn4Q0WZuveoJ/ =LhNY -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7961 - can't place calls
Thanks for the reply. I am not getting any output from the Asterisk CLI when I
place the call. The phone give busy signal as soon as I push the first digit
of the extension #. When I call the 7961 from another extension I get the
following on the CLI - that works fine.
asterisk*CLI>
-- Executing [0...@inside_sip_phones:1] Verbose("SIP/0206-08522f28",
"1|Extension 0203") in new stack
Extension 0203
-- Executing [0...@inside_sip_phones:2] Dial("SIP/0206-08522f28",
"SIP/0203|30") in new stack
-- Called 0203
-- SIP/0203-08529f68 is ringing
-- SIP/0203-08529f68 answered SIP/0206-08522f28
-- Packet2Packet bridging SIP/0206-08522f28 and SIP/0203-08529f68
== Spawn extension (inside_sip_phones, 0203, 2) exited non-zero on
'SIP/0206-08522f28'
asterisk*CLI>
Thanks.
-Brad
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Friday, November 20, 2009 5:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7961 - can't place calls
There could be many reasons for this. You should show us the output of
your asterisk cli during a failed call attempt, and we can go from
there.
Thanks,
--Warren Selby
On Nov 20, 2009, at 5:23 PM, Brad Darr wrote:
> Hello,
>
>
>
> I have been working on getting a Cisco 7961G to place calls on my *
> server for a while now with no luck. I can receive calls just fine
> but I get a fast busy when I try to place calls. I have googled and
> been to many different sites but the solution has not been found.
> Anyone out there had a similar issue and found the fix?
>
>
>
> Asterisk server is 1.4.26
>
> Cisco 7961G is running SIP version 8.5-2S
>
>
>
> Thanks.
>
>
>
> -Brad
>
>
>
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7961 - can't place calls
Hello, I have been working on getting a Cisco 7961G to place calls on my * server for a while now with no luck. I can receive calls just fine but I get a fast busy when I try to place calls. I have googled and been to many different sites but the solution has not been found. Anyone out there had a similar issue and found the fix? Asterisk server is 1.4.26 Cisco 7961G is running SIP version 8.5-2S Thanks. -Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI TDM440E still has echo on bridged connections
" Hello, Strangely i purchase a TDM440E with the echo canceller onboard and I still receive a horrible echo and i'm only using bridged connections between DAHDI/4 and DAHDI/1. I turned of echo cancellation on bridged connections which seemed to help alittle bit. I ran fxotune -i5 and setup fxotune -s to apply settings on startup, which has helpped but there is still echo on the begining of each call. Any idea's as to why there would be an echo at the beginning of a bridged conversation with echo cancellation turned off? Also this only happens on incomming calls not outgoing calls Thank you, Brad Finberg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call transfer using DTMF
" Yes, In the features.conf under featuremap you need the blindtransfer un-commented blindxfer => ## Then in your extensions.conf you need to have at least a capital T exten => example,1,Dial(ZAP/4/12345,,T) Then during the call you can press ## and asterisk will say transfer. Then dial in the extension you want to transfer too. Thank you, Brad Finberg - Original Message - From: Michael To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Date: Tuesday, July 14 2009 11:07 PM Subject: [asterisk-users] call transfer using DTMF Is there a way to transfer a call, while in the middle of the call, using DTMF? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Asterisk behind a SIP Proxy
We are trying to configure Asterisk (version 1.6.1.0) with some SIP
phones behind a SIP Proxy/NAT device. The phones register properly to
Asterisk, and to get Asterisk to register properly to the external SIP
registrar we added this to the general section of sip.conf (the address
of the Asterisk system on the LAN is 192.168.30.5):
outboundproxy=192.168.30.10
register => myname:mysec...@my.provider.com/100
The problem we are facing is that it appears that the outboundproxy
value is being treated globally by Asterisk so it sends all SIP traffic,
including traffic to the phones, to the proxy. The behavior we want is
that all outbound traffic is sent to the proxy, but inbound SIP traffic
to the phones should be sent direct to the phones.
The result we see is that an inbound Invite is received by Asterisk and
then the Invite for the phone is sent by Asterisk to the outbound proxy.
This causes much confusion.
Can anyone please tell me how to configure Asterisk properly for working
behind a SIP Proxy?
Below you will find our configuration.
Thanks,
Brad
Here is the channel for our SIP provider:
[my_provider]
type=peer
host=my.provider.com
username=100-phone
secret=mysecret
context=incoming
canreinvite=no
qualify=300
insecure=port,invite
Here is a sample phone entry in sip.conf:
[100_phone]
type=friend
username=100-phone
secret=100secret
host=dynamic
context=internal
Here is the relevant part of extensions.conf:
[incoming]
exten => 100,1,Dial(SIP/100_phone,30)
exten => 100,n,Hangup()
[internal]
exten => _X.,1,Dial(SIP/my_provider/${EXTEN})
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] notifyringing=no does not work
"
Hello,
If anybody has any idea's to where I should start looking to fix the below
subscription problem. If there is another mailing list I should post this to
please let me know.
Thank you,
Brad Finberg
- Original Message -----
From: Brad Finberg
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Date: Thursday, April 9 2009 9:42 AM
Subject: notifyringing=no does not work
Hello,
I have been trying to get my Grandstream GXP2000 phones to stop showing ringing
state on monitored extensions. But no matter where I put notifyringing=no
asterisk still sends the ringing state to the phones. Is this a bug I should
report or is there another way around it.
Here is how i have my subscriptions setup:
extensions.conf
[demo]
exten => 6100,hint,SIP/100
exten => 6101,hint,SIP/101
exten => 6102,hint,SIP/102
exten => 6103,hint,SIP/103
exten => 6104,hint,SIP/104
exten => 6105,hint,SIP/105
exten => _1XX,1,SIPAddHeader(Alert-Info:\;info=ring3)
exten => _1XX,2,Dial(SIP/${EXTEN},20,Tt)
exten => _1XX,3,VoiceMail(${ext...@default,u)
exten => _1XX,104,VoiceMail(${ext...@default,b)
sip.conf
[general]
allowsubscribe=yes
;subscribecontext = default
notifyringing=no
notifyhold=yes
;limitonpeers=yes
[100]
type=peer
context=demo
callerid=Back Office <100>
username=100
secret=(Private)
host=dynamic
nat=no
qualify=yes
canreinvite=no
dtmfmode=rfc2833
call-limit=5
mailbox=...@default
disallow=all
allow=ulaw
allow=alaw
;allow=g723.1
allow=g729
;callingpres=allowed_passed_screen
notifyringing=no
callgroup=1
pickupgroup=1
Asterisk CLI:
Extension Changed 6100[demo] new state Ringing for Notify User 105
Extension Changed 6100[demo] new state Ringing for Notify User 104
Extension Changed 6100[demo] new state Ringing for Notify User 102
Extension Changed 6100[demo] new state Ringing for Notify User 101
Extension Changed 6100[demo] new state Ringing for Notify User 103
Thank you,
Brad Finberg___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] notifyringing=no does not work
"
Anybody have any idea's
Thank you,
Brad Finberg
- Original Message -----
From: Brad Finberg
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Date: Thursday, April 9 2009 9:45 AM
Subject: [asterisk-users] notifyringing=no does not work
Hello,
I have been trying to get my Grandstream GXP2000 phones to stop showing ringing
state on monitored extensions. But no matter where I put notifyringing=no
asterisk still sends the ringing state to the phones. Is this a bug I should
report or is there another way around it.
Here is how i have my subscriptions setup:
extensions.conf
[demo]
exten => 6100,hint,SIP/100
exten => 6101,hint,SIP/101
exten => 6102,hint,SIP/102
exten => 6103,hint,SIP/103
exten => 6104,hint,SIP/104
exten => 6105,hint,SIP/105
exten => _1XX,1,SIPAddHeader(Alert-Info:\;info=ring3)
exten => _1XX,2,Dial(SIP/${EXTEN},20,Tt)
exten => _1XX,3,VoiceMail(${ext...@default,u)
exten => _1XX,104,VoiceMail(${ext...@default,b)
sip.conf
[general]
allowsubscribe=yes
;subscribecontext = default
notifyringing=no
notifyhold=yes
;limitonpeers=yes
[100]
type=peer
context=demo
callerid=Back Office <100>
username=100
secret=(Private)
host=dynamic
nat=no
qualify=yes
canreinvite=no
dtmfmode=rfc2833
call-limit=5
mailbox=...@default
disallow=all
allow=ulaw
allow=alaw
;allow=g723.1
allow=g729
;callingpres=allowed_passed_screen
notifyringing=no
callgroup=1
pickupgroup=1
Asterisk CLI:
Extension Changed 6100[demo] new state Ringing for Notify User 105
Extension Changed 6100[demo] new state Ringing for Notify User 104
Extension Changed 6100[demo] new state Ringing for Notify User 102
Extension Changed 6100[demo] new state Ringing for Notify User 101
Extension Changed 6100[demo] new state Ringing for Notify User 103
Thank you,
Brad Finberg
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] notifyringing=no does not work
"
Hello,
I have been trying to get my Grandstream GXP2000 phones to stop showing ringing
state on monitored extensions. But no matter where I put notifyringing=no
asterisk still sends the ringing state to the phones. Is this a bug I should
report or is there another way around it.
Here is how i have my subscriptions setup:
extensions.conf
[demo]
exten => 6100,hint,SIP/100
exten => 6101,hint,SIP/101
exten => 6102,hint,SIP/102
exten => 6103,hint,SIP/103
exten => 6104,hint,SIP/104
exten => 6105,hint,SIP/105
exten => _1XX,1,SIPAddHeader(Alert-Info:\;info=ring3)
exten => _1XX,2,Dial(SIP/${EXTEN},20,Tt)
exten => _1XX,3,VoiceMail(${ext...@default,u)
exten => _1XX,104,VoiceMail(${ext...@default,b)
sip.conf
[general]
allowsubscribe=yes
;subscribecontext = default
notifyringing=no
notifyhold=yes
;limitonpeers=yes
[100]
type=peer
context=demo
callerid=Back Office <100>
username=100
secret=(Private)
host=dynamic
nat=no
qualify=yes
canreinvite=no
dtmfmode=rfc2833
call-limit=5
mailbox=...@default
disallow=all
allow=ulaw
allow=alaw
;allow=g723.1
allow=g729
;callingpres=allowed_passed_screen
notifyringing=no
callgroup=1
pickupgroup=1
Asterisk CLI:
Extension Changed 6100[demo] new state Ringing for Notify User 105
Extension Changed 6100[demo] new state Ringing for Notify User 104
Extension Changed 6100[demo] new state Ringing for Notify User 102
Extension Changed 6100[demo] new state Ringing for Notify User 101
Extension Changed 6100[demo] new state Ringing for Notify User 103
Thank you,
Brad Finberg___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vividial issue
does anyone have a sample dialplan for vici dial that does not include any pri stuff. I am running exclusively SIP for everything and trying to edit the sample dialplan and removing anything to do with a pri card is becoming a nightmare! Thank you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialing a 60anything number issue!
we just did a brand new installation of asterisk 1.4 on ubuntu with a sagnoma t-1 card everything went smooth (other than fighting a little outbound call issue that we are sure is a tdm network to sagnoma issue) inbound calls are fine dialplan is silly basic with outbound channels set to factory specs and inbound dialplan a silly basic to ring one phone. This MUST be a know issue! When we dial any outbound number, everything does what it should BUT if the dial 60anything the call goes into the "busy signal of death". We deleted everything relevant to 60 "as per make samples" installs in extensions.conf searched the other obvious files and naddha! Can some kind soul point us in the right direction please? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic outbound calling issue : a lot closer
I figure it out, asterisk is using the wrong ip address.
I have bind address set to the correct ip address. How to I force asterisk to
use the correct ip address?
--- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:
> From: Brad <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Basic outbound calling issue : a lot closer
> To: asterisk-users@lists.digium.com
> Date: Friday, August 15, 2008, 9:33 PM
> This what they sent me
> You need to send:
> - 11-digit originating # (i.e., 1-NPA-NXX-)
> - 10-digit terminating #
>
> This got me a lot further in extensions.conf
>
> exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
>
> I am getting a 503 error on the phone and asterisk is
> giving me:
>
> == Auto fallthrough, channel 'SIP/100-09ef2cc0'
> status is 'CONGESTION'
> -- Executing [EMAIL PROTECTED]:1]
> Dial("SIP/100-09f2ee18",
> "SIP/[EMAIL PROTECTED]|30|r") in new stack
> -- Called [EMAIL PROTECTED]
> -- Got SIP response 503
> "NoCircuitChannelAvailable" back from
> 64.211.41.115
> -- SIP/64.211.41.115-09ef2cc0 is circuit-busy
> == Everyone is busy/congested at this time (1:0/1/0)
> == Auto fallthrough, channel 'SIP/100-09f2ee18'
> status is 'CONGESTION'
>
>
>
> --- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:
>
> > From: Brad <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-users] Basic outbound calling
> issue
> > To: asterisk-users@lists.digium.com
> > Cc: "Felippe Silvestre"
> <[EMAIL PROTECTED]>
> > Date: Friday, August 15, 2008, 9:06 PM
> > extensions.conf
> >
> > [To_Airspring]
> > exten => 55,1,Playback(demo-echotest) ; Let them
> know
> > what's going on
> > exten => 55,2,Echo ; Do the echo test
> > exten => 55,3,Playback(demo-echodone) ; Let them
> know
> > it's over
> >
> > exten => 100,1,Dial(SIP/100,20)
> >
> > sip.conf
> >
> > ;; twinkle softphone
> > [100]
> > user=100
> > nat=yes
> > type=friend
> > secret=andreasd
> > host=dynamic
> > context=To_Airspring
> >
> >
> > This should ba all I need
> >
> > exten => 100,1,Dial(SIP/100,20) should catch it and
> send
> > it to Sip
> >
> >
> > --- On Fri, 8/15/08, Felippe Silvestre
> > <[EMAIL PROTECTED]> wrote:
> >
> > > From: Felippe Silvestre
> > <[EMAIL PROTECTED]>
> > > Subject: RE: [asterisk-users] Basic outbound
> calling
> > issue
> > > To: [EMAIL PROTECTED], "Asterisk Users
> Mailing
> > List - Non-Commercial Discussion"
> >
> > > Date: Friday, August 15, 2008, 12:25 PM
> > > Check if you have some rule to dial under brad1
> > context
> > >
> > > dialplan [EMAIL PROTECTED]
> > >
> > > Regards
> > >
> > > Felippe Silvestre
> > >
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED]
> On
> > Behalf
> > > Of Brad
> > > Sent: Friday, August 15, 2008 12:09
> > > To: Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > > Subject: [asterisk-users] Basic outbound calling
> issue
> > >
> > > I am trying to lauch a first outbound call.
> > > I am connected to my telco via a peer which is a
> > little
> > > different from what I consider the norm.
> > >
> > > extinsions.conf
> > >
> > > [To_Bandwidth]
> > > ignorepat => 9
> > > exten => 9,1,Dial(Sip/g2/)
> > > exten => 9,2,Congestion
> > >
> > > sip.conf
> > >
> > > [To_Bandwidth]
> > > canreinvite=yes
> > > context=from-pstn
> > > dtmfmode=rfc2833
> > > host=.com
> > > nat=no
> > > outboundproxy=xxx.com
> > > qualify=no
> > > type=peer
> > >
> > >
> > > error
> > >
> > > [Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035
>
> > > handle_request_invite: Call from 'brad1'
> to
> > > extension
> > > '919544790554' rejected because extension
> not
> > > found.
> > >
> > >
> > >
> > >
> > > ___
> > > -- Bandwidth and Colocation Provided by
> > > http://www.api-digital.com --
> > >
> > >
Re: [asterisk-users] Basic outbound calling issue : a lot closer
I get congestion (same error) with
exten => _NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
not dialing 1
exten => _1NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
dialing 1
exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
dialing 9
All the same
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/100-b7c03ce8", "SIP/[EMAIL
PROTECTED]|30|r") in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 503 "NoCircuitChannelAvailable" back from 64.211.41.115
-- SIP/64.211.41.115-09f2ee18 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/100-b7c03ce8' status is 'CONGESTION'
--- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:
> From: Brad <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Basic outbound calling issue : a lot closer
> To: asterisk-users@lists.digium.com
> Date: Friday, August 15, 2008, 9:33 PM
> This what they sent me
> You need to send:
> - 11-digit originating # (i.e., 1-NPA-NXX-)
> - 10-digit terminating #
>
> This got me a lot further in extensions.conf
>
> exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
>
> I am getting a 503 error on the phone and asterisk is
> giving me:
>
> == Auto fallthrough, channel 'SIP/100-09ef2cc0'
> status is 'CONGESTION'
> -- Executing [EMAIL PROTECTED]:1]
> Dial("SIP/100-09f2ee18",
> "SIP/[EMAIL PROTECTED]|30|r") in new stack
> -- Called [EMAIL PROTECTED]
> -- Got SIP response 503
> "NoCircuitChannelAvailable" back from
> xxx.xxx.xxx
> -- SIP/xxx.xxx.xxx-09ef2cc0 is circuit-busy
> == Everyone is busy/congested at this time (1:0/1/0)
> == Auto fallthrough, channel 'SIP/100-09f2ee18'
> status is 'CONGESTION'
>
>
>
> --- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:
>
> > From: Brad <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-users] Basic outbound calling
> issue
> > To: asterisk-users@lists.digium.com
> > Cc: "Felippe Silvestre"
> <[EMAIL PROTECTED]>
> > Date: Friday, August 15, 2008, 9:06 PM
> > extensions.conf
> >
> > [To_Airspring]
> > exten => 55,1,Playback(demo-echotest) ; Let them
> know
> > what's going on
> > exten => 55,2,Echo ; Do the echo test
> > exten => 55,3,Playback(demo-echodone) ; Let them
> know
> > it's over
> >
> > exten => 100,1,Dial(SIP/100,20)
> >
> > sip.conf
> >
> > ;; twinkle softphone
> > [100]
> > user=100
> > nat=yes
> > type=friend
> > secret=andreasd
> > host=dynamic
> > context=To_Airspring
> >
> >
> > This should ba all I need
> >
> > exten => 100,1,Dial(SIP/100,20) should catch it and
> send
> > it to Sip
> >
> >
> > --- On Fri, 8/15/08, Felippe Silvestre
> > <[EMAIL PROTECTED]> wrote:
> >
> > > From: Felippe Silvestre
> > <[EMAIL PROTECTED]>
> > > Subject: RE: [asterisk-users] Basic outbound
> calling
> > issue
> > > To: [EMAIL PROTECTED], "Asterisk Users
> Mailing
> > List - Non-Commercial Discussion"
> >
> > > Date: Friday, August 15, 2008, 12:25 PM
> > > Check if you have some rule to dial under brad1
> > context
> > >
> > > dialplan [EMAIL PROTECTED]
> > >
> > > Regards
> > >
> > > Felippe Silvestre
> > >
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED]
> On
> > Behalf
> > > Of Brad
> > > Sent: Friday, August 15, 2008 12:09
> > > To: Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > > Subject: [asterisk-users] Basic outbound calling
> issue
> > >
> > > I am trying to lauch a first outbound call.
> > > I am connected to my telco via a peer which is a
> > little
> > > different from what I consider the norm.
> > >
> > > extinsions.conf
> > >
> > > [To_Bandwidth]
> > > ignorepat => 9
> > > exten => 9,1,Dial(Sip/g2/)
> > > exten => 9,2,Congestion
> > >
> > > sip.conf
> > >
> > > [To_Bandwidth]
> > > canreinvite=yes
> > > context=from-pstn
> > > dtmfmode=rfc2833
> > > host=.com
> > > nat=no
> > > outboundproxy=xxx.com
> > > quali
Re: [asterisk-users] Basic outbound calling issue : a lot closer
This what they sent me
You need to send:
- 11-digit originating # (i.e., 1-NPA-NXX-)
- 10-digit terminating #
This got me a lot further in extensions.conf
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
I am getting a 503 error on the phone and asterisk is giving me:
== Auto fallthrough, channel 'SIP/100-09ef2cc0' status is 'CONGESTION'
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/100-09f2ee18", "SIP/[EMAIL
PROTECTED]|30|r") in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 503 "NoCircuitChannelAvailable" back from 64.211.41.115
-- SIP/64.211.41.115-09ef2cc0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/100-09f2ee18' status is 'CONGESTION'
--- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote:
> From: Brad <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Basic outbound calling issue
> To: asterisk-users@lists.digium.com
> Cc: "Felippe Silvestre" <[EMAIL PROTECTED]>
> Date: Friday, August 15, 2008, 9:06 PM
> extensions.conf
>
> [To_Airspring]
> exten => 55,1,Playback(demo-echotest) ; Let them know
> what's going on
> exten => 55,2,Echo ; Do the echo test
> exten => 55,3,Playback(demo-echodone) ; Let them know
> it's over
>
> exten => 100,1,Dial(SIP/100,20)
>
> sip.conf
>
> ;; twinkle softphone
> [100]
> user=100
> nat=yes
> type=friend
> secret=andreasd
> host=dynamic
> context=To_Airspring
>
>
> This should ba all I need
>
> exten => 100,1,Dial(SIP/100,20) should catch it and send
> it to Sip
>
>
> --- On Fri, 8/15/08, Felippe Silvestre
> <[EMAIL PROTECTED]> wrote:
>
> > From: Felippe Silvestre
> <[EMAIL PROTECTED]>
> > Subject: RE: [asterisk-users] Basic outbound calling
> issue
> > To: [EMAIL PROTECTED], "Asterisk Users Mailing
> List - Non-Commercial Discussion"
>
> > Date: Friday, August 15, 2008, 12:25 PM
> > Check if you have some rule to dial under brad1
> context
> >
> > dialplan [EMAIL PROTECTED]
> >
> > Regards
> >
> > Felippe Silvestre
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On
> Behalf
> > Of Brad
> > Sent: Friday, August 15, 2008 12:09
> > To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> > Subject: [asterisk-users] Basic outbound calling issue
> >
> > I am trying to lauch a first outbound call.
> > I am connected to my telco via a peer which is a
> little
> > different from what I consider the norm.
> >
> > extinsions.conf
> >
> > [To_Bandwidth]
> > ignorepat => 9
> > exten => 9,1,Dial(Sip/g2/)
> > exten => 9,2,Congestion
> >
> > sip.conf
> >
> > [To_Bandwidth]
> > canreinvite=yes
> > context=from-pstn
> > dtmfmode=rfc2833
> > host=.com
> > nat=no
> > outboundproxy=xxx.com
> > qualify=no
> > type=peer
> >
> >
> > error
> >
> > [Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035
> > handle_request_invite: Call from 'brad1' to
> > extension
> > '919544790554' rejected because extension not
> > found.
> >
> >
> >
> >
> > ___
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register
> >
> > Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Basic outbound calling issue
extensions.conf [To_Airspring] exten => 55,1,Playback(demo-echotest) ; Let them know what's going on exten => 55,2,Echo ; Do the echo test exten => 55,3,Playback(demo-echodone) ; Let them know it's over exten => 100,1,Dial(SIP/100,20) sip.conf ;; twinkle softphone [100] user=100 nat=yes type=friend secret=andreasd host=dynamic context=To_Airspring This should ba all I need exten => 100,1,Dial(SIP/100,20) should catch it and send it to Sip --- On Fri, 8/15/08, Felippe Silvestre <[EMAIL PROTECTED]> wrote: > From: Felippe Silvestre <[EMAIL PROTECTED]> > Subject: RE: [asterisk-users] Basic outbound calling issue > To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial > Discussion" > Date: Friday, August 15, 2008, 12:25 PM > Check if you have some rule to dial under brad1 context > > dialplan [EMAIL PROTECTED] > > Regards > > Felippe Silvestre > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf > Of Brad > Sent: Friday, August 15, 2008 12:09 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Basic outbound calling issue > > I am trying to lauch a first outbound call. > I am connected to my telco via a peer which is a little > different from what I consider the norm. > > extinsions.conf > > [To_Bandwidth] > ignorepat => 9 > exten => 9,1,Dial(Sip/g2/) > exten => 9,2,Congestion > > sip.conf > > [To_Bandwidth] > canreinvite=yes > context=from-pstn > dtmfmode=rfc2833 > host=.com > nat=no > outboundproxy=xxx.com > qualify=no > type=peer > > > error > > [Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035 > handle_request_invite: Call from 'brad1' to > extension > '919544790554' rejected because extension not > found. > > > > > ___ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register > > Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Basic outbound calling issue
I am trying to lauch a first outbound call. I am connected to my telco via a peer which is a little different from what I consider the norm. extinsions.conf [To_Bandwidth] ignorepat => 9 exten => 9,1,Dial(Sip/g2/) exten => 9,2,Congestion sip.conf [To_Bandwidth] canreinvite=yes context=from-pstn dtmfmode=rfc2833 host=.com nat=no outboundproxy=xxx.com qualify=no type=peer error [Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035 handle_request_invite: Call from 'brad1' to extension '919544790554' rejected because extension not found. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VICIDial error
Solved! You have to get to the end of the "scratch install directions" to find the database setup. This information SHOULD be in the standard vicidial install instructions. Classic case of stupid flippn' administrator combined with poor documentation. Install the database. du! --- On Fri, 8/8/08, Brad <[EMAIL PROTECTED]> wrote: > From: Brad <[EMAIL PROTECTED]> > Subject: [asterisk-users] VICIDial error > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Date: Friday, August 8, 2008, 6:02 PM > Warning: Cannot modify header information - headers already > sent by (output started at > /home/telecom/public_html/vicidial/admin.php:1175) in > /home/telecom/public_html/vicidial/admin.php on line 1187 > > Warning: Cannot modify header information - headers already > sent by (output started at > /home/telecom/public_html/vicidial/admin.php:1175) in > /home/telecom/public_html/vicidial/admin.php on line 1188 > > Has anyone ever seen this? > > I am getting a double header sent with all aspects of the > Astisk GUI including VICIDial > > > > > ___ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VICIDial error
Warning: Cannot modify header information - headers already sent by (output started at /home/telecom/public_html/vicidial/admin.php:1175) in /home/telecom/public_html/vicidial/admin.php on line 1187 Warning: Cannot modify header information - headers already sent by (output started at /home/telecom/public_html/vicidial/admin.php:1175) in /home/telecom/public_html/vicidial/admin.php on line 1188 Has anyone ever seen this? I am getting a double header sent with all aspects of the Astisk GUI including VICIDial ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Dialer proof of concept
Yes, everyone will have the same message. You think building the call fill in the spooler is the most effectient? Can you refer me to a page that will explain pulling the info from a sql db into a call file? Last thing, I dial out to an extension, not a registered sip provider, my provider does not require authentication. How would I pull from DB, put into call file, send to "context". Short and pretty. Just trying to get my head back into Asterisk. --- On Fri, 8/8/08, emist <[EMAIL PROTECTED]> wrote: > From: emist <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Auto Dialer proof of concept > To: [EMAIL PROTECTED] > Date: Friday, August 8, 2008, 4:18 PM > Hey Brad, > > The simplest way I thought to implement it for a client who > needed > multiple calls to be placed based on time was to code a > deamon that > would query the db every given interval, check if there > were any calls > that needed to be made and pull those out and build call > files. > > That seemed to work pretty decent. However, if you just > want to call > 2000 people with the same message with the click of a > button all you'd > have to do is have a frontend that would pull the message > off the db > along with each person's number and build call files in > a loop. > > Thats simple and relatively scalable, unless you're > doing 1,000,000 at a > time or something. > > Regards, > > Igor H. > > Brad wrote: > > I read that last night and I was curious about > followme' > > > > Will this give me the ability to dial out 10 - 2000 > simultaneously calls the easiest and control to number of > call? > > > > doing it the file method looks kind of easy for proof > of concept, but not very manageable. I could seeing putting > 2000 files into a directory would be very cumbersome. > > > > Eventually, the number will be coming from a sql > database. > > > > I am just trying to get general concept to prove it > works for now, but do not want to have to reconfigure to > much over the weekend > > > > > > --- On Fri, 8/8/08, emist <[EMAIL PROTECTED]> > wrote: > > > >> From: emist <[EMAIL PROTECTED]> > >> Subject: Re: [asterisk-users] Auto Dialer proof of > concept > >> To: [EMAIL PROTECTED], "Asterisk Users > Mailing List - Non-Commercial Discussion" > > >> Date: Friday, August 8, 2008, 3:57 PM > >> > http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out > >> > >> Bradley Sumrall wrote: > >>> I am a returning Asterisk user. > >>> > >>> It has been a few years since I played with it > and > >> trying to get a server up for proof of concept > >>> What is the easiest method of having asterisk > dial 5 > >> numbers simultainiously and deliver a pre recorded > message? > >>> > >>> > >>> > >>> > ___ > >>> -- Bandwidth and Colocation Provided by > >> http://www.api-digital.com -- > >>> AstriCon 2008 - September 22 - 25 Phoenix, > Arizona > >>> Register Now: http://www.astricon.net > >>> > >>> asterisk-users mailing list > >>> To UNSUBSCRIBE or update options visit: > >>> > >> > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, AudioCodes, Caller ID
Hello all, I'm working on a little project right now and have ran into a snag. Was hoping someone would be kind enough to give me a few pointers to help me get past the current issue... I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...) that I'm trying to get to play nice with Asterisk 1.4. I've got it to the point where the AudioCodes box picks up calls coming in on the FXO ports and routes them to a predefined extension on the Asterisk server. The problem I am having is that I cannot seem to get the AudioCodes box to pass the Caller ID data to Asterisk. I have tested the setup with both my Teltone TLS-5C line simulator and the local telco's POTS lines... Both the teltone and the telco are passing caller ID data onto the line and it is being displayed properly on a standalone CID display box that I hooked up for testing. Yet, that data seems to disappear somewhere in the MP-114... As far as I can tell, I have the AudioCodes box setup to accept caller ID... (Enable Caller ID = yes, type = Bellcore). The AudioCodes documentation is somewhat lack-luster when it comes to real examples, but I did my best to interpret all the various settings throughout the box that seem to affect Caller ID presentation... Does anyone have any experience with getting an AudioCodes MediaPack to work nicely with Asterisk? If so, some advice or sample config snips or anything really would be very helpful... I'd appreciate it... Thanks in advance, Brad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ekiga register problems
204/20466.176.193.46D 5063 Unmonitored It just came up after a reboot on its own??? Go figure, windows problem! Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, May 29, 2007 12:17 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ekiga register problems On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote: > returning newbie. > Trying to register ekiga for the first time to my asterisk server only. > > [204] > user=204 > context=internal > type=friend > secret=xxx > insecure=very > canreinvite=no > host=dynamic > disallow=all > allow=ulaw > allow=alaw > nat=no > > Can anyone tell me what I am missing? > I am not behind NAT or a firewall What exactly is the problem you get? What is the line for "204" in 'sip show peers'? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ekiga register problems
returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not behind NAT or a firewall ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto/forced call
Can anyone guide me to a "how to" on automating a call? I know a little piece of code (normally python) has to be place some where and then a file has to be mv into the spooler. Where do I get the run down? I have a button on another application that sends an email and I want it to also send a text message through asterisk! Brad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dealing with 2 SIP providers
On Fri, May 11, 2007 at 11:06:35AM -0400, Mike wrote: > Hi, > > I have a question of using 2 SIP providers. Let's say I have provider A and > provider B, and I would like my calls to go to A, and then B if A wasn`t > available What would be really cool, but require special code in the chan_sip dialer, would be automatic support of multiple providers in a similar fashion to the way Asterisk can ring two channels and only talk to the first to answer. You can't just do this with outgoing providers, because if you try to ring two at once, you may very well have the second one go to a voicemail and thus answer right away (because the first is ringing) and you would treat that as the success. What I have in mind is something like this: a) Invite to main provider b) Await some intermediate response, such as a RINGING code or some early media c) If you don't get that after a short timeout (more like 5 seconds) then INVITE the second provider d) Upon the receipt of a ringing or early media code from either, CANCEL the other. Now you would have to get your timings right because there could still be risk of doing something bad, such as a 2nd call going to voice mail or residual ringing making a call waiting on the recipient. (I don't know what typical 5ess do with a 2nd call that comes in while still ringing, anybody known?) Anyway, this could be a good course when a provider has known unreliability. Long timeouts and restarts are very annoying to users. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)
I am very confident the 7960G has a sip load. I know for sure the regular 7960 does and the G just means gigabit interface. The 7970 was the only one that didn't because of all the color interface/touch screen, and Cisco was still pushing call manager big time, so skinny was the only load available. If you log into cisco.com, they have it under software. Sometimes people post it on the internet. Asterisk is supposed to be more skinny friendly these days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Howk Sent: Wednesday, April 25, 2007 7:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP) >From reading the SLA docs, SIP hints are use to get the lights on the phone to show the "correct state". I was under the impression that the SIP firmware on the 7960's didn't support the SIP hints properly (or at all), which means that SLA won't work properly on a 7960. If anyone has gotten this to work, I'd like to hear about it. --Jason. John C. Wolosuk Jr. wrote: > Has anyone had any success with getting SLA going between 2 SIP phones? > (Particularly a set of Cisco 79xx's) The SLA document that comes with > the asterisk source is about as clear as mud. > > Does anyone have a working sip.conf, sla.conf, and extensions.conf that > I can use for reference? > > The part I'm most confused about is how to build the lines in sip.conf > and how the phones should behave. It seems apparent that the phones > should not register with asterisk, otherwise all the phones will try to > register to be THE phone for a given extension. should these lines be > built like a trunk/peer? if I could be an example of how lines for SLA > should look in sip.conf, that would be helpful. > > Also I'm somewhat annoyed that I have to compile zaptel drivers that I > don't use in order to compile the app_meetme.so module so I can have the > SLA functions available to the dialplan... > > Any feedback is greatly appreciated! > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Random Asterisk deaths
test -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Jensen Sent: Wednesday, April 25, 2007 7:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Random Asterisk deaths Asterisk 1.2.13 (newest available in Debian Etch) No VMs, nothing strange whatsoever about the setup. In the box: TE405P X100P clone AMD Athlon XP 2400+ 512 MB RAM In the logs I get a lot of "zt hook failed: Device or resource busy" and "Avoided initial deadlock for '0xXX', 10 retries!" but these happen all the time and don't increase or decrease in frequency around the time that Asterisk dies. On 4/24/07, Bryan M. Johns <[EMAIL PROTECTED]> wrote: > What version are you running? Anything creative like VMs or other unique configurations in use? > > Bryan Johns > Partner > > Shelton | Johns > Office: 678.248.2637 > FindMe: 678.229.1809 > http://www.sheltonjohns.com > > - Original Message - > From: "Wayne Jensen" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > Sent: Tuesday, April 24, 2007 7:24:26 PM (GMT-0500) America/New_York > Subject: [asterisk-users] Random Asterisk deaths > > Every once in a while for no apparent reason, Asterisk has been dying > on me, dropping all calls in progress. There's nothing in the log > file or on the Asterisk console that indicates the reason. Some days > it doesn't happen at all. Other days it happens two or three times. > > The problem began on Friday, but the last time anything was changed on > that box was at least a week before that. > > Any suggestions on what to do/where to look to find out what's going > on and fix the problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom SP 601 Reboot Issue- Help!
Hard reset the phone first! Provision and see if it is fixed. No? Upgrade software (watch out for provisioning changes). Still rebooting? Downgrade software. Still rebooting? You now have a new door stop! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Garstang Sent: Wednesday, April 25, 2007 7:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom SP 601 Reboot Issue- Help! That used to happen to us _ALL_ the time. Sometimes you'd just have to press the 'Directory' key and the phone would instantly reboot. It was very easy to reproduce and Polycom where useless at admitting it might be a problem. It occurred on several phones. Funnily enough, the phone it was most reproducable on was a 601 being used as a Receptionist phone with 3 sidecars... and about 35 buddies being watched. Hmmm! Russ Beaupre wrote: > We had a situation where the 601 base went missing and the electrical > connection between the side cars and the 601 was broke. Might be > worth a look to see if the phone got damaged. > > > -Original Message- > From: Jerry Jones <[EMAIL PROTECTED]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: Tue, 24 Apr 2007 12:27:46 -0500 > Subject: Re: [asterisk-users] Polycom SP 601 Reboot Issue- Help! > > The only reboot issue I have with 1 sidecar is the side car deciding > to randonly rebbot, not the phone itself > > Perhaps upgrading to 2.1 will help? > > > On Apr 24, 2007, at 10:51 AM, J French wrote: > > > I have a Polycom 601 with 3 expansion modules running 2.0.3. We > > have Buddywatch set up on around 42 users on the expansion > > modules. We are experiencing reboots on the 601. Today it > > happened twice after users paged through the phones. The page > > groups have about 23 phones each. There is a third page group > > comprising all 46 phones. I'm thinking it may be an issue with > > changing buddywatch state on so many buddies so quickly. Also, > the > > cpu usage is pegged at 100% for around 3 minutes after it reboots, > > FWIW. > > > > Anyone else experiencing rebbots on the 601? Advice is really > needed! > > > > Thanks > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Marketing 101
Personally, I look for specialty applications. Work smart not hard! I myself am looking for outstanding marketers for a fire hot industry / telecom application. I have all of the correct "duckies" in a row, just need to send it to the market the correct way. [EMAIL PROTECTED] _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Wednesday, April 25, 2007 6:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Marketing 101 Agreed. Highly-considered purchases like telco infrastructure are not as much a push as a pull sale. It's about being in the right place at the right time with all the right answers. Almost like buying a home. Since the turnover is SO long with core business process equipment, it's almost a beauty contest when the time comes around. A better analogy would probably be in luxury car buying. You need to look good, have a good feature set, be luxurious to drive, have all the right bells and whistles above and beyond basic requirements, and then of course have a track record of reliability and great service. Just my $.02 -- --- Robert Goodyear Managing Partner Brand Up LLC Knight West 949.542.7001 DIRECT 949.542.7010 FAX 888.272.6387 x501 [EMAIL PROTECTED] [EMAIL PROTECTED] --- On Apr 25, 2007, at 10:52 AM, SIP wrote: Businesses RARELY are in a position to choose new Telco systems providers. Oftentimes, that sort of decision is made by whomever leases them the office space, or was made once back in the beginning, and they've had no real reason to re-evaluate their service/provider. There are, however, plenty of Telco events where the providers hawk their wares and the installers tout their expertise. Cold Call/Networking/Word of Mouth are decent methods of getting your name out there as an alternative, but be prepared to run into a great many situations in which the system or provider they have 'works well enough' so they're not interested in changing. shadowym wrote: Thanks for the advice. Maybe I should clarify what I was asking. It's not so much the how but the what. What are people doing to get PBX Sales/Support business. I know how to get IT business but potential customers still see the Telco business as quite different and are used to using separate companies for that. What I was asking is how the traditional telco guys get new sales/support/consulting business. With IT it's usually a combination of cold call/networking/word of mouth. I'm hoping that Telco is the same but I never see any telco guys at networking events so I am thinking they cold call and advertise targeted at business owners. I'm not sure though. -Original Message- From: dave cantera [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 24, 2007 9:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Marketing 101 shadowym, best thing to do is talk to a lot of consultants, coaches, and marketing people... take the approach you do with learning open source only reverse it... instead of reading source (internal) ask people (external)... it is a big undertaking and the most important task you have... marketing is a bigger task than the technical (for a tech anyway) don't go it alone nothing happens without marketing (and sales)... marketing is *not* sales... daveC shadowym wrote: I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.dig
RE: [asterisk-users] No Audio with SIP to only one provider whenswitching servers
I would not rule your firewall out as the problem! Port 5060 is only the authentication port, the rtp stream is normally 10,000 thru 20,000. Some of your phone may have STUN modules on them. Open 10,000 thru 20,000 and 5060 on the firewall. Stick some holes in it for testing purposes. Verify ports are open with telnet:port number "both ways", telnet is your friend. If it works, close the holes up and consult your firewall docs Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hadar Pedhazur Sent: Wednesday, April 25, 2007 6:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No Audio with SIP to only one provider whenswitching servers I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is configured identically to the old one as well. All of my IAX connections just worked. All but one of my SIP connections just worked as well (which is why I can't believe it's a firewall issue). StanaPhone, which I use for 2 incoming DIDs, registers correctly, and rings my phones correctly when a call comes in. However, once answered, there is dead silence in both directions, on 100% of the calls. There isn't any problem on StanaPhone's side (which has provided a _fantastic_ service ever since I signed up!), because I can connect to them with X-Lite and receive calls with audio. More importantly, if I fire up Asterisk on the old server, it still works!!! I can connect with X-Lite to the new server, so the new server definitely accepts SIP connections, and audio works. It's _not_ a codec problem. I verified that on both the working and non-working servers the connection is established with ulaw on both sides. I have dumped the "peer" and the "channel" on both, while the call was active, and they look identical to me, except for the random bits associated with a particular connection. Here are the ones from the machine that fails: *CLI> sip show peer XX * Name : XX Secret : MD5Secret: Context : default Subscr.Cont. : Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : "" <> Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : sip.stanaphone.com Addr->IP : 204.147.183.18 Port 5060 Defaddr->IP : 0.0.0.0 Port 0 Def. Username: 12345678 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : OK (20 ms) Useragent: Reg. Contact : new*CLI> sip show channel [EMAIL PROTECTED] * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] Our Codec Capability: 4 Non-Codec Capability: 1 Their Codec Capability: 4 Joint Codec Capability: 4 Format ulaw Theoretical Address:204.147.183.18:5060 Received Address: 204.147.183.18:5060 NAT Support:RFC3581 Audio IP: AAA.BBB.CCC.DDD (local) Our Tag:as360c7ca5 Their Tag: 0bd46ffd48e4fbffb3a68f13f8ad2599 SIP User agent: Username: 87654321 Peername: 12345678 Original uri: sip:204.147.183.55:1024 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:204.147.183.18;ftag=as360c7ca5;lr=on DTMF Mode: rfc2833 SIP Options:(none) Finally, I built 1.2.18 from source today, and everything is working perfectly _except_ for StanaPhone, which continued to connect with no problems, but deliver no audio in either direction. I have no idea what else to try, and would appreciate _any_ guidance. Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] call dispatching - legacy application
Then you take the number you get from your database and put it into the asterisk spooler. Remember, the temp file you create has to be moved to the spooler using the mv command. Nothing else works. There might be one other step, I am not sure with Asterisk 1.4. I had a friend help me do it before and he said he had to write a little piece of python code to make it work properly (we were making asterisk call phone automatically). I am not sure if you will need this or not. I know the process because I had it done for me before. I am at the beginning trying to do the same thing, though my php is rusty. Maybe you can hook a brother up with the proper code to "grab caller id and query mysql? To answer your question, Yes, you are on the right track! Brad [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of adriano ghezzi Sent: Wednesday, April 25, 2007 6:13 PM To: asterisk users Subject: [asterisk-users] call dispatching - legacy application Hy all need to preprocess 1) incoming call get caller id lookup some info in my db, 2) based on the result dispatch the call to the right operator step 1 is ok I developped a small .php script that connect manager and parse events, now I have to tell AAH do dispatch call to the right operator questions 1) is this the right practice ? 2) where to find a complete manager api reference, (to buy too) note that there is a legacy application that query the db actually php script send the request to this app and wait for response I'm a programmer at very first installation of AAH , just testing capabilities thanks in advance for any help and suggestion. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SLES?
This question should really be asked at Linux. Basically FC, Red hat, Centos and SUSE are all the same. Some minor security defaults and a few directory changes. Last time I check (it has been some time now); All of the above on their enterprise level basically only supported the install, updates (which are free on Yum anyways) and some minor other stuff. More advanced was a few thousand and $15,000 for priority for a year. Digium rates have gone up for support, but "WELL WORTH IT" when it deals with Asterisk and Linux, minor to advanced! I pay the piper from time to time and always get the job done quickly! Outside of that, this mailing list is a great place for support, we all work together! Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans Witvliet Sent: Wednesday, April 25, 2007 5:24 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SLES? On Mon, 2007-04-23 at 04:46 +0300, Tzafrir Cohen wrote: > On Mon, Apr 23, 2007 at 01:49:12AM +0200, Hans Witvliet wrote: > > Hi all, > > > > Just curious, > > > > Quite a while a go, i was checking for supported SW-platform. > > AFAIR, it was RHES and SLES > > > > Now it's only RHES-4 and FC-3 or FC-4. > > Not a single syllable about CentOS or SLES-9 or SLES-10 > > > > It probably just runs fine, but any chance of getting support for their > > *-enterprise version? (just in case of, if one needs it) > > Asterisk is an official package of SLES. Consider asking them as well > regarding support (including newer versions of Asterisk). > I knew that it included in the retail version (prof-10.x) and in open-suse (no support). And it was surely NOT included in SLES-9. At that time i suggested to get it included with sles-10, but marcus/andreas replied that they considered asterisk not stable enough to be able to have SLA-contracts connected to it, hence they would not include it. I'm pretty sure that one way or another, asterisk will just work fine on SLES-10. Point is however, that management would like to see a possible backup for support, in case the shit hits the fan. Official, with SLA-contracts and so on It took years to get SLES into the organisation, so open-suse, fedora or Centos are out-of-the-question, and RHEL will be another long struggle. hw -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk & Pix firewalls
Pix usually uses NAT, A quick fix is to simply forward the ports in your NAT statements. If the pix is new, call Cisco and cheat like I do so often! Brad _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Wednesday, April 25, 2007 9:31 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk & Pix firewalls Don This may not be a solution to your question, but I would like to share that I’ve been having one way audio issues when connecting point to sight to a PIX 515E using SIP. I changed to IAX and this is working perfectly now. It was paynless to configure IAX2, so you might want to consider it. Ed From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Don E. Wisdom Sent: Tuesday, April 24, 2007 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk & Pix firewalls Hi, I asked this last week but i didn't get any answer So i will elaborate on my question. I need to setup a pix 515 firewall (running 7.2.2 OS) to allow sip traffic thru it from a sip phone wherever i may be. The pix is where all my servers are colocated and i will need to connect thru it from softphones / hardphones wherever i happen to be traveling. I need help setting up the pix for inbound and outbound sip/iax traffic. Any help would be greatly appreciated. Thanks --Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom Provisioning Problems
Access the phones through the web interface, Compare version numbers with the phones that work Compare only with other 501 phones Make sure all settings are identical, most polycom web interfaces will loose there setting adjustments if you click on another tab, so do one page at a time, click save, then let it reboot, then go to next section. Also, is you asterisk server local? If remote, and the above does not work, look at routing. Are you behind NAT? Try and access the phone via telnet from a remote server to the auth ports of the phone and vice verse. (I know you can telnet from a Cisco phone, I would imagine polycom has a similar features) Let me know if this helps!!! Polycom is very picky! Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze Sent: Wednesday, April 25, 2007 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Provisioning Problems Hello I am having some difficulties provisioning a set of polycom 501 phones, while another set of phones are working just fine. My Asterisk box is dual homed. On one network, where the asterisk box runs dhcpd and there are only phones, provisioning works as expected. However, for phones that are connected thru the other interface (and receive their IP address from a separate router), they are not provisioning. To add to the confusion, it seems that they fail in inconsistent ways. Even after specifying the FTP server address, name and password, these phones will complain that they cannot connect to the server, and begin loading the stored configuration. In addition, when they come up, their dates are set to Jan 1, 2001. (I think I can fix this by specifying the snmtp address, but the other phones seem to be able to find the snmtp on their own.) In inspecting the -boot.log files, the phones that fail have CDP enabled, while the phones that succeed have CDP disabled. I think this is Continual Data Protection, but don't see where to disable it on the phone interface. Is this a cause of the failure? Any insight will be greatly appreciated. Thanks -- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk GUI issue, minor
I installed the asterisk GUI, "Asterisk web manager", it loads fine, but if I go to the AGI section, I get a "permission denied" Obviously apache cannot access the /etc/asterisk directory. I added apache as group, but still the same problem. Suggestion any one? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Problem converting a Cisco 7960 to SIP
Indeed this might be the failing point... Unfortunately, because I have no Cisco CCO account anymore, I have no access to firmware... I will try to find a copy of an old firmware for these phones. If I can find one, I hope it fixes my problem. Thanks, Brad > Apologies in advance if this is a stupid comment, but don't you have to > convert to SIP at a much lower version than 8. I had to go all the way > back to version 3.? if I remember correctly to convert from SCCP to SIP. > > > dave -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Problem converting a Cisco 7960 to SIP
Hello all,
I've got myself into a bizzare situation that I can't seem to get myself
out of... Was wondering if anyone had some advice that might get me 'over the
hill' on this...
Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco
7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are
all on a separate LAN. There is no VLAN configuration. The Asterisk box also
is running a TFTP server and DHCP server. The 4 original Cisco's work fine
still. The Polycom IP500's work fine.
The problem is with trying to get this new Cisco 7960 online... It came
pre-loaded with the SCCP image and I cannot get it to convert to SIP.
Currently it is running the following versions:
App Load ID: P0030301MFG2
Boot Load ID: PC0303010200
Version: 3.1(MF.G2)
The phone contacts the DHCP server and gets an IP successfully. The
dhcpd.conf file:
##
# dhcpd.conf - dhcp config file for eth1 / sip phones
##
authoritative;
ddns-update-style interim;
ignore client-updates;
local-address 192.168.1.1;
option tftp-boot-server code 150 = ip-address;
option tftp-boot-server 192.168.1.1;
subnet 192.168.1.0 netmask 255.255.255.0 {
option routers 192.168.1.1;
option subnet-mask 255.255.255.0;
option domain-name-servers 192.168.1.1;
option time-offset -18000; # Eastern Standard Time
option ntp-servers 192.168.1.1;
option tftp-server-name "192.168.1.1";
default-lease-time 43200;
max-lease-time 86400;
pool {
range 192.168.1.100 192.168.1.150;
}
}
Then the phone contacts the TFTP server. Below are the logs:
Mar 29 12:09:15 asterisk1.local atftpd[32276.-1208575056]: Serving OS79XX.TXT
to 192.168.1.144:49427
Mar 29 12:09:16 asterisk1.local atftpd[32276.-1208575056]: Serving
SEP001795B05B1D.cnf.xml to 192.168.1.144:49428
Mar 29 12:09:16 asterisk1.local atftpd[32276.-1208575056]: Serving
XMLDefault.cnf.xml to 192.168.1.144:49429
Mar 29 12:09:16 asterisk1.local atftpd[32276.-1208575056]: Serving
SEP001795B05B1D.cnf to 192.168.1.144:49430
OS79XX.TXT contains:
P003-08-6-00
Originally the SEP001795B05B1D.cnf file didn't exist. Since it was for
CallManager, I didn't bother to configure it and just setup the SIPmac.cnf
file instead. The phone never requested the SIPmac.cnf file...
I found a trick via google that uses the SEPmac.cnf file to change
firmware. The SEP file now contains:
2000
192.168.1.1
P003-08-6-00
The TFTP directory contains:
0004f20049bc-app.log
0004f20049bc-boot.log
SEP001795B05B1D.cnf
polycom_brad.cfg
sip.cfg
WORKING_POLYCOM_sip.cfg
WORKING_POLYCOM.cfg
phone1.cfg
0004f20049bc.cfg
0004f20049bc-phone.cfg
0004f20049bc-appFlash.log
SoundPointIPLocalization
.cfg
-directory~.xml
SoundPointIPWelcome.wav
sip.ld
sip.ver
bootrom.ld
SIP001795B05B1D.cnf
snom.cnf
SIP0012DABF2AAA.cnf
SIP0012D9B94C72.cnf
SIP001280B9D6E1.cnf
SIP001280F3AFC7.cnf
SIPDefault.cnf
DSM2ColorLogo_3.bmp
OS79XX.TXT
P003-08-6-00.bin
P003-08-6-00.sbn
P0S3-08-6-00.loads
P0S3-08-6-00.sb2
797x_template.cnf.xml
cisco_util
Desktops
dialplan.xml
merlin2.pcm
RINGLIST.DAT
syncinfo.xml
All other phones work fine. Therefore, I assume all the firmware is in the
right place... They all converted to SIP firmware fine...
When I try to do the **# unlocking, it does nothing... Everything still
shows locked. The phone doesn't have an Unlock Settings function (assuming
firmware is too old)
The phone, when it boots, goes through an endless loop consisting of:
Configuring VLAN
Configuring IP
Then it starts over.
What in the heck am I doing wrong? I thought that the OS79XX.TXT file
should have taken care of pushing out the new image. And the phone is
grabbing the file via TFTP, but it's like it ignores the idea of changing
firmware.
Also, when I try to do a factory reset (holding down #, power cycling) it
never asks for the reset key sequence and never said it detected the key
sequence.
Any advice would be appreciated.
Thanks,
Brad
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help making a voice record server $$$
Hey there folks, Looking to my favorite mailing list for assistance and have a few bucks to pay you for your time. Me: Played with asterisk for a while in the early days and getting stuck on silly stuff on a time sensitive project for a friend. Project: PSTN incoming call to asterisk and then back to PSTN again, asterisk will hold and record the RTP stream. Upon disconnect, asterisk will name the record file by CID and Date. That's it! E mail me with how much you want for your time and this will surely grow into other project that are later going to be implemented on this server. Sincerely, Brad [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Question about DSP in Digium card
Whether it is IAX, SIP, H323 or ? These are authentication handshakes to establish an rtp stream. SIP = user name and password in a standardized IP packet IAX = same H.323 = same Is also has to do with what codec are supported as well. As far as NAT is concerned! Yep, tell your ISP to forward the authentication port or just junk their gear and get something like a low end Cisco. Or Get IP Phones with STUN (a little pricey) Or Trick Use some type of tunneling gear to an outside IP (outside your NAT) and then bounce your authentication from this new gateway!!! i.e. establish a VPN connection to an outside router from an internal router and drive the call through there. Brad _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of A. Levy Sent: Tuesday, March 27, 2007 6:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Question about DSP in Digium card well, ...,we did not choose SIP because our customers are located behind NAT router (using private IP's) and those routers are not managed by them but by the ISP so it is very difficult to establish full duplex phone calls because you can not initiate voice over ip session from the internet (outside) to LAN side (inside) with private IP's. We could not establish 2-way phone calls, I mean, the conversation is listened in 1-way only. As I mentioned before, we can not configure PAT into the NAT router neither because is handled by the ISP and the passwords are unknown That's why we decided to use IAX instead of SIP, I mean, IAX is more robust than SIP when the NAT router is 3th-party managed and the PAT feature is not enable. On the other and we tested IAX over dialup links and it worked fine Those are the reasons we choose IAX as "acess protocol" to our SIP/H323 Network. You know, the access networks of the customers are different completely: Private IP Address over DSL lines (NAT Router), Public IP Address over DSL lines, Corporate Networks over dedicated Links (Public and IP Addresses), Dialup links, .. Any comment would be welcomed, thanks a lot Levy.- 2007/3/24, A. Levy <[EMAIL PROTECTED]>: Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX <-> ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Refresher course needed!
Hello everyone
My name is Brad, I am an old Asterisk Vet of the very early days just coming
back to join the group.
Ok, for starters, I feel like the "monkey with the light bulb" looking at
extensions.conf and sip.conf.
It has been some time.
A friend ask me to set up a asterisk server that records phone calls.
FC4
Asterisk 1.4
And all the latest and greatest
Problem number 1
Some good "get back into the grove" literature.
"I work CLI only", never much for graphics and gui's
Problem number 2
We have asterisk logged into teliax but cannot see the inbound call come up
on the CLI
Tethereal says this;
1660 3.829799 207.174.202.4 -> 66.109.17.92 SIP Status: 100 Trying(1
bindings)
1661 3.831357 207.174.202.4 -> 66.109.17.92 SIP Status: 200 OK(1
bindings)
Asterisk says this;
*CLI>
Nothing, notta!
My extensions.conf
(yes, I loaded the samples)
[general]
static=yes
writeprotect=no
clearglobalvars=no
;#include "filename.conf"
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;From here is brads stuff
exten => _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten => YOURNUMBER,1,Answer()
exten => YOURNUMBER,1,DIAL(SIP/user,20)
Thanks to all!
Brad
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Refund from SellVoip?
On Sat, Mar 24, 2007 at 12:13:25PM -0700, Martin Joseph wrote: > On 2007-03-23 14:37:18 -0700, "Tom Lynn" <[EMAIL PROTECTED]> said: > > > > > > >Now I know where they've been spending my remaining balance... > > I still use Sellvoip as my primary terminator, and have found the call > quality to be superior to any other ITSP from my location (Seattle). > > I agree completely that there is no support from this company, which is > a major issue if you are trying to support other customers. > > Still, I remain a happy customer of sellvoip, with Teliax and Nufone > configured as backups... > > I wouldn't expect a refund for cancellation of prepaid phone usage, > does the original agreement you have with then suggest that they owe > you a refund? Sure, if you had a bunch of DIDs with them, and you sent in requests to cancel them and they never acted on them and kept billing until your accounts ran out of money. Or if due to a bug, they started billing another customer's calls to you, quickly depleting your account and giving you access to that other customer's private CDR data, and they never answered any tickets, calls or emails about the matter for days. Yes, I could see wanting a refund. The "good call quality and no customer service" routine seems good, until something goes wrong and you have no idea when they will fix it. You can use backup providers for termination, but not origination. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel silly issue
I am geet this error, I assume because I have zero digium hardware installed. This is to be an entirely web based PBX. Can anyone point me to an easy 123 for installing zaptel in dummy form? I need music on hold for a VPS server. Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, March 19, 2007 2:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Configuring Faxs any help :) younss azzayani wrote: > Hi everybody, > after installing hylafax & iaxmodem i get this email > == > > The HylaFAX software thinks that there is a problem with the modem > on device /dev/ttyIAX that needs attention; repeated attempts to > initialize the modem have failed. This would be better off on the HylaFAX+ mailing list. Please, when posting there, include your configuration files. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Refund from SellVoip?
On Fri, Mar 16, 2007 at 04:16:21PM -0700, Tom Lynn wrote: > At this point, I'm simply contacting the State of Washington Attorney > General's office. They're ignoring my e-mails and I'm done monkeying > around. > It makes no sense. The put together a good system on the tech end, Asterisk based, decent call quality and faster call completions than any of the other folks I have been trying, at good prices. And then dropped it all on the floor, not responding to calls, emails or tickets often for weeks and months, if at all. Their interface needed work but that I can tolerate. Not being able to reach somebody for an urgent problem makes no sense. Does anybody know Jed Stafford? As far as I can tell this ended up being a one-man or two-man operation. It's just sad. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Refund from SellVoip?
On Fri, Mar 16, 2007 at 11:32:31AM -0700, Tom Lynn wrote: > Has anyone been successful in getting a refund from SellVoip when you've > cancelled service? You were able to cancel service with Sellvoip? That's impressive, that implies they actually responded to a request you made to cancel service. It was rare I could get them to respond to any request or ticket before I gave up. Their service (as in call quality) was pretty good. It's a shame they could not, while I was using them, afford to provide customer service. It's one of the few companies I ever wrote to to say, "Could you please raise your prices, so that you can afford to hire a customer service rep?" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Bluetooth Re: [asterisk-users] Nomination for Coolest App in 2007
Another idea that has just come to me regarding bluetooth and a PBX is like this. Many people would like to use headsets with their IP phones. Some support wired headsets, but bluetooth headsets can be a good choice for a headset -- no wires, many people often have one, and there is a rich competitive market that makes them cheaper than many of the headset products available.There are a few hardphones that will take a bluetooth headset -- this seems to me like an obvious idea that's not very expensive to implement -- and I have a plantronics bluetooth base station for use with any phone that works on my Cisco 7960 and other phones with a headset jack. But it's expensive, and in the latter case, goes digital-analog-digital. So with a bluetooth channel that can talk to a bluetooth headset, you could have Asterisk itself give you your bluetooth headset on your desk phone.To do this, you would: a) Send calls to your desk phone to both the phone and headset. You can answer on either. See caller id on phone. b) Tranfer calls from desk phone to bluetooth headset. (Unfortunately requires the cumbersome transfer function of many phones.) c) Better still, have it so if the bluetooth headset opens a connection, and the "paired" desk phone is in a call or has a call on hold, auto-grab that call and put it on the headset. d) If the BT headset hangs up, instead of a normal hangup, consider that a transfer back to the desk phone, which will ring, and can then take over the call for any phone functions (real transfers, etc.) If you really meant to hang up, it does mean you have to answer and immediatly hang up this final call.The user could program if she wants this, and from which modes she wants it. Other than some minor inconveniences of (d), you get something much like the ability to have a bluetooth headset as a handsfree headset for any phone on the system, even analog phones. Unfortunately, you can only have a limited number of BT headsets operating at once from any bluetooth dongle (typically 8, and at that point you also get interference issues.) However, dongles are only $10, so you can have several on one server. If people will be far from the server, you have to do this functionality from remote machines, which might be best done with an IP phone softphone module. In that case, you can have more UI, include a choice on termination. The general idea of a "secondary phone" which, if it connects, automatically grabs any call on the "main phone" is handy. For real phones with dials, you can have this be a magic extension to dial. Bluetooth headsets can't dial and can have it simply happen on connection. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nomination for Coolest App in 2007
On Wed, Mar 14, 2007 at 09:37:45AM -0500, Steve Totaro wrote: > > Another interesting (from an American's perspective anyways) is that > inbound calls on cell phones are free. Even if you buy a SIM with a > little pre-paid time and use up the time, you can still receive inbound > calls for free for a couple months. Inbound calls on cell phones outside North America are alas, not free, though people pretend they are free. They are "caller pays for airtime." The only free incoming call systems I have seen are some mobile to mobile free call plans, and a small number of North American mobile plans that, for a flat monthly or daily fee, offer free incoming. The caller-pays system found outside North America is, in my view -- though I know some differ -- one of the last, great curses of old world telephony on our new environment. With my VoIP terminators, I can call most of the world's landline's for a price so low I think of it as free, with one exception -- the damn caller-pays cell phones which cost over an order of mangitude more because the fact that the payer doesn't negotiate the price removes the competition that would normally drive the price down. (And has driven it down in the receiver-pays countries.) However, for people in those countries, the bluetooth module does seem like a good idea. Obviously in places with no landlines, but also in places with these bizarre prices, so that if you call one mobile from another mobile, it's cheap, but if you call from a SIP terminator, it's 25 cents/minute. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nomination for Coolest App in 2007
On Tue, Mar 06, 2007 at 11:14:15PM -0500, Steve Totaro wrote: > Mine goes to chan_bluetooth. Somewhat of a pain getting it going but I > am totally floored with how cool it is! > > Thanks, > Steve Totaro > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users My, that is a cool app. I look forward to running it when it's a bit more stable. While the outgoing call ability seems of limited use since cell call quality is not that exciting to even unlimited night and weekend minuets are probably not too attractive compared to 1 cent/minute SIP terminations, there are a number of interesting possibilities: a) If your target has an unlimited calls to other customers/family/etc. plan, you would want to call them this way to save minutes. b) Handy on some carriers for checking cell voice mail. (I have found that with many US carriers, however, you can call your cell phone with CID set to your cell number, and it goes directly to voice mail Make sure you have a password!) c) Incoming calls, obviously handy. d) During daytime, program to receive incoming calls and say, "I am at my desk. Please call me at xxx- or press 1 to have me call you back at " so you get better quality and don't bill cell minutes. In the evening, assuming unlimited weekends, you might forward directly. Can it send and receive SMS via bluetooth too? I also like a lot the talk of coming softphones with bluetooth headset support. This would allow you to use your bluetooth headset as an extension on your Asterisk pbx. I happen to have a bluetooth headset that plugs into my hard phone -- I wish more hardphones supported them natively -- and that's handy. This could be just as good. To really get it right you would want some speech recognition so you could place calls from the bluetooth headset by saying names and digits, as many cell phones can already do. Of course, a linux softphone could reside right on the asterisk box. You could multi-dial your bluetooth headset and your hard phones and answer where you like. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
On Sun, Mar 04, 2007 at 02:34:21PM -0600, Kevin P. Fleming wrote: > Brad Templeton wrote: > > In many packages there is some file (usually the change log) which always > > tells you what version of the program you have in your hands, in terms > > of the program's current version number -- of course you can see the > > svn revision numbers and dates but they don't trivially translate. > > I would be surprised to see such a file in a direct checkout from the > project's SCM system. Even if that file existed, it would exist for a > very short time as the moment a new commit occurred that branch would no > longer 'be' 1.4.1, for example. Yup, typically it's a changelog and significant changes are noted along with version number bumps. I'm presuming the /branch/1.4 is "the latest stable version of 1.4 with the latest patches."Since there is a 1.4.1 it means it is also 1.4.1 with the latest patches -- or so I presume. Having a file means people can look at see what they have, without having to ask here :-) Now that I know I can interpret what it means.I'm assuming that the latest /branch/1.4 is "the one to run" if you want a stable system with all known and tested patches and fixes but only modest new functionality -- or should one really be running /tags/1.4.1 and regularly updating your tag in order to get that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
On Sun, Mar 04, 2007 at 08:50:48AM -0600, Kevin P. Fleming wrote: > Brad Templeton wrote: > > I did an svn up and there are new files, but nothing in the change > > files about it being 1.4.1.Many packages with various minor > > versions tend to have the master branch (like 1.4) mean "The latest > > stable version of 1.4, be it 1.4.0 or 1.4.whatever", while if you > > check out 1.4.1 that means you stay at 1.4.1 even if there is a 1.4.8. > > You will never see ChangeLog files in the branches in our Subversion > repository, because we only create them in the tags as we make releases. > > Since you didn't give us the output of 'svn info', we don't know what > you have already checked out... but if you checked out > http://svn.digium.com/svn/asterisk/branches/1.4, then you have > everything that is in Asterisk 1.4.1 plus whatever changes have been > committed to the 1.4 branch since the release was made. Thanks, Kevin. Yes, I have http://svn.digium.com/svn/asterisk/branches/1.4. I was running /trunk before but it wasn't stable enough to be a production system (no surprise.) I am presuming that the above is intended to be stable in this fashion. In many packages there is some file (usually the change log) which always tells you what version of the program you have in your hands, in terms of the program's current version number -- of course you can see the svn revision numbers and dates but they don't trivially translate. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
Do I get 1.4.1 by just doing an svn up in a checkout of the 1.4 branch, or do I have to switch to a new tag or branch for what I have checked out? I did an svn up and there are new files, but nothing in the change files about it being 1.4.1.Many packages with various minor versions tend to have the master branch (like 1.4) mean "The latest stable version of 1.4, be it 1.4.0 or 1.4.whatever", while if you check out 1.4.1 that means you stay at 1.4.1 even if there is a 1.4.8. What's the procedure here? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sellvoip configuration....Please Help!!!!
On Fri, Feb 23, 2007 at 09:11:18AM +0100, [EMAIL PROTECTED] wrote: > hi guy, i have a problem, i have an sellvoip account and i want > configure asterisk for outbound calls. Alas, the best sellvoip configuration, I eventually had to conclude, was not to use sellvoip. They have good quality service, which makes this even more frustrating, but they are woefully understaffed, and can take months -- yes months, not hours, not days, not weeks -- to respond to support requests and tickets.They really are a good value when they work, but I had to abandon them, because problems can appear and you have no idea when they will be fixed. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
On Tue, Feb 20, 2007 at 12:08:15PM -0700, Natambu Obleton wrote: > Why not make it like DNS and have each provider have their lookups > deligated to a local server and then each ISP will run a caching > server that will use a serial number system to get updates.. just like > DNS. > > I know there are lot more DNS lookups then CNAM lookups per hour... > isn't there? :) > Hey, we could even build a system where DNS can be used to take any phone number and look up data about it, not just a name, but even a URI to redirect calls to for it, a source of presence info and more. What a great idea! Unfortunately, since phone numbers are believed to be owned by telcos and not by individuals, such a system would probably make the mistake of delegating control over the numbers to the telcos, who would feel no particular motive to help people bypass what they sell, and so I predict it will languish for a long time with no real deployment in the USA. :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
On Mon, Feb 19, 2007 at 09:02:56PM -0500, C F wrote: > I doubt it's CNAM since it has old an outdated listings. > > On 2/19/07, Paul <[EMAIL PROTECTED]> wrote: > >Does google really have the true CNAM database? When I enter my number, > >I get a search result for my business listing at yellowpages.com > > > >Are you referring to something available in a google area other than the > >search engine? Well, at one time Google got a large telno to name database. I don't know if they have updated it. They can certainly afford to. There are other web sites that do reverse number lookups as well. Still, starting with their database seems a good choice. They might not like you scraping it at once but a thousand * boxes pulling records one call at a time is not something they are going to be bothered by. If this, combined with other info from other sources (including contributions from people who have CNAM) builds a workable database, you will eventually get the LECs contributing their data to it. People want their name to show up correctly. If millions start using a database, the LECs will want their data in it, especially if entry is free or near free for bulk entries. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
On Mon, Feb 19, 2007 at 03:01:50PM -0500, Paul wrote: > I think terms of service for most CNAM providers prohibits sharing the > data and limits the amount of time it can be cached for your own reuse. > I don't know why they manage to get this level of control over the cnam database so that they can charge a penny per lookup as well as monthly fees. Does anybody know how this happens? Clearly some people buy the database at a good price. Google for example has it, and there are asterisk hacks to do google lookup (if you query a 10 digit phone number in google, you'll get not just name but address etc.) Perhaps they are just paying. One way to build a free database would be to simply have people share the results of all sorts of searches. People who pay for CNAM as end users, for example, have signed no contract to not share the data. So they could, if trusted, forward those records to be stored in the shared database. People who don't could take any number they get, and if it's not in the shared database already, do a google query, and if that gives a result, store that in the shared database. (Also store negative results with a timestamp so that you know that the google lookup provides no info.) http://www.google.com/search?q=nn&pb=r Eventually you would get a pretty good database, perhaps one big enough that CLECs start wanting to update it directly? Now there may still need to be something to pay for all of this, but the fees could be much lower. Charge fees for the latest copy or real time query but just have the regular database out there for download and local lookup. Or perhaps just use the google api? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
On Tue, Jan 30, 2007 at 10:23:09PM +0100, Benny Amorsen wrote: > > "PC" == Patrick Cervicek <[EMAIL PROTECTED]> writes: > > PC> But then all RTP Traffic of my internal phones will go over > PC> Asterisk. I want RTP to go "Peer-to-Peer". ==> "Intern-2-Intern" > PC> and "Extern-to-Extern" should go P2P and "Intern-2-Extern" should > PC> go over Asterisk, see picture > > I understand what you want. I am telling you that you cannot get what > you want, and the best compromise you can achieve. Either your > internal-to-internal calls go direct, or your external-to-external do. > Pick one. You can almost get it, if your NAT will hairpin, by having your "internal" phones all present their external addresses. Then all phones will appear external, and all can talk to one another (though there can still be port change problems on reinvite if you don't do explicit ports) -- but internal to internal (and all other internal involved calls) will go through your NAT box, not your Asterisk box. However, the NAT must hairpin audio, and a lot of them don't. Here is Cullen's latest chart of who hairpins: http://tools.ietf.org/wg/behave/draft-jennings-behave-test-results-03.txt Only a few do it, and only a few make his "group A" category at all. Of course, many of these boxes are not expensive so it may be worth switching, though many of us love the WRT54G and its clones because of the ability to use open source firmware. However, the bastard does not hairpin, I don't know if any of the new firmwares do. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
On Tue, Jan 30, 2007 at 12:00:17PM +0100, Patrick Cervicek wrote:
> Brad Templeton schrieb:
> >On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote:
> >
> >>PC> When I set for Extern1/2 canreinvite=yes it works, but
> >>PC> "Intern-2-Extern" doesn't work because Asteisk gives out the
> >>PC> private IP-Adresses of Int1/2
> >>
> >>Asterisk can't give out a public IP-address for Int1/2. Where
> >>would it get one from?
> >>
> >
> >Correct that it doesn't. But some kind sould could indeed code a
> >variety of techniques to get it, such as:
>
> Again: My Problem is not "Intern-to-Extern" (NAT,Stun). My Problem is
> "Extern-to-Extern", that the external phones are not talking RTP
> *directly* to each other. This is bad, when Asterisk is in Europe and
> the Phones are in Asia.
> ___
That you can usually make work with most phones today, as long as
the phone has some NAT penetration in it.The most versatile
approach is STUN -- see if you can configure STUN on your phones.
Some phones that don't have STUN do support a hard-configured external
address. However, they often require a numeric IP, which means
that they only work if you have a static IP, and your RTP port number
never changes on the outside.
A few phones also support extracting information from rport fields etc.
However, they tend to have STUN too.
Then you must also keep a port in the NAT open, by one of the following
methods.
a) Manual hole in the NAT -- most NATs support this ("port forwarding")
and it is recommended. Set your phone to a fixed SIP Port
and port forward that to your phone. Requires phone have a fixed
internal IP usually.
b) Keep alive transmitted by phone (will be on phone's config)
c) qualify=yes in asterisk sip.conf sends keep alive from asterisk.
Works, but if it ever fails your phone won't ring until it
re-registers.
d) Very short registration time, like 30 seconds to 1 minute. This
will effectively do a keep alive from the phone. I like this
least.
If your phone does not understand NAT properly it will forward
a useless SDP that is internal. This is what some Cisco phones
do. For those, you are currently screwed. Some day Asterisk might
support rewriting SDPs that contain unreachable addresses but it
does not, at present, do this.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote: > PC> When I set for Extern1/2 canreinvite=yes it works, but > PC> "Intern-2-Extern" doesn't work because Asteisk gives out the > PC> private IP-Adresses of Int1/2 > > Asterisk can't give out a public IP-address for Int1/2. Where > would it get one from? > Correct that it doesn't. But some kind sould could indeed code a variety of techniques to get it, such as: a) If the phone supports STUN or a static external IP mode, it often will include the known external IP in its SIP headers. Asterisk would ignore those headers and talk directly to the internal address (equivalent to what it does with NAT=yes, but for RTP too) when talking directly to the phone, but when connecting the phone to outside addresses, it would pass these addresses and SDPs. b) In many cases, such phones sit behind a NAT which can have hard coded port forwardings so that Asterisk could simply be told, in sip.conf, what the external IP and port of the phone are, and it could rewrite SDPs when needed. c) The phone could attempt to reach Asterisk through an external address, even though the phone is on the same natwork as the Asterisk. This would tell Asterisk the external IP of the phone, and it could rewrite SDPs. However, since many NATs don't support hairpin of audio, for those NATs asterisk must use internal SDPs to connect devices behind the same NAT, including itself if that is the case. Anyway, as noted, there are many answers to your question of how asterisk _could_ get the info from, but at present it does not. They are all kludges of course. Just about all NAT penetration involves some kludging. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
On Sat, Jan 27, 2007 at 01:55:31PM -0500, Jerry wrote: > > On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote: > >> > >> Unless you are monitoring calls, want full CDR etc, > >> then that's what you want anyway. > > > > CDR are not affected by how the audio flows. > > While technically true, I believe (it may have changed in 1.4) that if you > allow reinvites, the signalling path follows the audio path, and you end > up with reported calls lasting 3 seconds. > > So, if you want full (ie accurate as to the length of time) CDR, then I > think asterisk has to remain in the call path. That would have been an odd bug. Signalling in SIP only moves when you do a REFER or similar. Reinvites can't change it. Having the signalling flow differently from the audio is a feature, not a bug, a very important one. With SIP INFO (or its planned successor) you can get the DTMF without having to get the audio, which is highly valuable. Right now Asterisk needs to stay in the audio stream to get DTMF, and that is one of the prime reasons it does. (The others are NAT, recording and meetme, the latter 2 of which should be a small minority of calls.) This is an important thing. Done properly, audio should almost never flow through the switching machine, or only flow for a portion of the call. The result can be orders of mangitude difference in bandwith requirements. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote: > >For a remote phone, not on the same network as the Asterisk > >box (in which event the NAT worries are different) you definitely > >want to use the same protocol for the phone as for your > >term/orig provider. Otherwise you will be forced to hairpin > >your audio through your asterisk server, adding latency and > >wasting bandwidth and cpu for little reason. > > Unless you are monitoring calls, want full CDR etc, > then that's what you want anyway. CDR are not affected by how the audio flows. Monitoring calls does require hairpin of the audio. Most people who are not call centers do not wish to monitor all calls or even more than few calls. (In fact in many states it is illegal unless you inform the other party, mostly limiting it to call center use.) If you had a call center * server in the USA hairpinning a call between India and the UK it would be really dumb, but even over shorter links it's dumb. > > I agree. Single SIP phones can usually be got to work behind > a reasonable NAT router. And with some work could be made to work without special config with all but the rarest NATs. Hopefully in 1.6. > > For a single phone - you are quite right. For multiple phones, > I'm not sure I agree - multiple SIP phones behind a NAT router > is going to require some extensive config , or a SIP proxy in the > router. Not really, other than the issue of NATS that won't hairpin between the phones. I have this situation, and our 2nd home I have 2 phones, on the * server at my main home. While I have linux computers at the 2nd home, it would be silly to put up a * server for the two phones if they can work through the NAT. It's not a big deal to tell the WRT54G I have to forward two ports to the 2 phones (well 3 if you include the wifi phone). There is no need at the 2nd house for intercom, so I would not put in a local server just for that. However, it does mean the remote location can't have SIP phones without things like STUN. > Ah, but it isn't just asterisk you have to change - it is > all the SIP implementations and all the routers :-) STUN is quite common in SIP phones, in fact the only major modern ones to not do it seem to be the Ciscos, though I have not tried the 8.0 firmware on them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
On Thu, Jan 25, 2007 at 10:19:06PM -0800, Yuan LIU wrote:
> Asterisk1 <--> NAT1 --- { Internet } --- NAT2 <--> Asterisk2
>
> If Asterisk1 can talk to Asterisk2 at trunk level, I'll be happy.
While I'm not sure of what tricks * plays at all levels, you
can certainly make this work if you have control of the NATs to
open ports, or if the asterisk servers know the address of their
partner and thus can keep the NAT "open" by sending keep-alives.
>
> The way Jeff Pulver puts it, ICE has conquered the world :-) Would love
> to learn more.
ICE is a methodology. You list every way you might be reached
(LAN, external addresses and addresses of outside relays) and the
other endpoint tries every way it can, ranked in order of quality,
and picks the best one. So if you're both on the same LAN it will
see that and use it. If you can't reach one another except through
a relay it identifies that and uses a relay. If, of course, you have
a willing relay.
(Skype solved that last problem :-)
>
> Is this the concept of STUN? Does this also create latency (by adding an
> additional leg in the route), packet loss, even jitter?
STUN is something else. Using a relay does indeed increase latency
(and thus echo) and may increase jitter and packet loss, though latency
is the big issue.
>
> I should have used FWD as an example. One can't say it uses proprietary
> clients. Does it stay away from voice path?
It provides a relay if one is needed. I don't know about today but
they started using jasomi boxes sold to deal with this question.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
On Wed, Jan 24, 2007 at 11:09:21PM -0800, Yuan LIU wrote: > >From: Brad Templeton <[EMAIL PROTECTED]> > > > >On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote: > >> In the meanwhile, use IAX, which understands about NAT pretty well. > >> If you have multiple SIP phones on a LAN behind a NATing router, just > >> put a small asterisk box on the LAN. It can manage your hairpin > >> calls internally, save you bandwidth by trunking the IAX traffic > >> to the central asterisk and avoid all the NAT hassle by using > >> a single port (outgoing) and refreshing it often enough for the > >> router to hold it open. > >> > >> Tim Panton > >> > >> www.mexuar.net > >> www.westhawk.co.uk/ > > > >IAX is a fine protocol as far as it goes, however this answer > >is really not a workable one. There are only a few IAX phones, > >and they are not nearly as solid and full featured as the many > >SIP phones. There are some IAX termination and origination > >providers, but there are far more SIP providers. > ... > >IAX is great but SIP is also a reality, and putting > >Asterisk into the "just works" category is a really > >important milestone. One I think that is intended > >to be improved a lot for 1.6. > > I have a really dumb question. It appears that Yahoo, MSN, AIM, you name > them, they don't have a NAT problem, and some use SIP. I don't think they > all stay in voice path, either. What takes? When you control both ends of the path, you can eliminate all NAT problems. Skype also deals almost perfectly with NAT (by using other nodes as relays if necessary) as does IAX. SIP was designed without much attention to NAT and it's had to be added on later and the different phones are all at different levels of implementation. Some time ago, actually, the SIP and SDP groups devised the ICE protocol for highly reliable NAT penetration, but it is still some distance from wide adoption, and I don't know when anybody will code up Asterisk adoption. Larger services like you describe often solve NAT by relaying traffic through their servers. They use a "trick", that if they suspect an endpoint is behind NAT, they just ignore what they see in the SDP, and send all traffic back to the source port/host that the traffic comes from. For RTP, they wait for packets to arrive at the (external, routable) RTP port they provided, and send the traffic back there instead of the often unroutable address in the SDP. Asterisk, if you set nat=yes, will do step 1 (SIP traffic back to the source it came from, ignoring Contact header) but it does not yet do the same for the RTP. If it did, you would be unlikely to get NAT trouble on phone to Asterisk calls, or calls hairpinned through Asterisk. But you don't want to hairpin unless absolutely necessary. It costs bandwidth and adds latency. Latency no only makes calls annoying, it increases the chance of echo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: High Quality Wireless Headset for Cisco IP Phones and *
On Tue, Jan 23, 2007 at 02:01:43PM -0600, Tom wrote: > Has anyone found a high quality wireless headset that works well with > Cisco 7960 IP phones on an asterisk system? > > I tried the vxxi offering but the sound quality was pretty bad. > > Since these are pricey, I don't want to sample blindly. > I've got one of the Plantronics bluetooth ones. It's OK, but frankly, with bluetooth hardware costing just a couple of bucks, you would think we should just see bluetooth becoming standard in every non-budget IP phone. People already have the headsets in many cases, and you can go digital all the way, and even rely on the headset's echo cancellation if you like. SNOM has a high end phone with this but otherwise it's been much slower to come than you would think. Alas, this doesn't really answer your question. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote: > In the meanwhile, use IAX, which understands about NAT pretty well. > If you have multiple SIP phones on a LAN behind a NATing router, just > put a small asterisk box on the LAN. It can manage your hairpin > calls internally, save you bandwidth by trunking the IAX traffic > to the central asterisk and avoid all the NAT hassle by using > a single port (outgoing) and refreshing it often enough for the > router to hold it open. > > > Tim Panton > > www.mexuar.net > www.westhawk.co.uk/ IAX is a fine protocol as far as it goes, however this answer is really not a workable one. There are only a few IAX phones, and they are not nearly as solid and full featured as the many SIP phones. There are some IAX termination and origination providers, but there are far more SIP providers. For a remote phone, not on the same network as the Asterisk box (in which event the NAT worries are different) you definitely want to use the same protocol for the phone as for your term/orig provider. Otherwise you will be forced to hairpin your audio through your asterisk server, adding latency and wasting bandwidth and cpu for little reason. In addition, many people just want to do things like give family or employees a phone they can take home, or take to a remote location and use on the PBX. They probably can't "just" put up an Asterisk server to make this happen, and nor should they want to. An additional server is not only more work and requires an always-on server computer, it's another thing that can go wrong. No thanks. Even if you can run Asterisk on a WRT54G, and thus don't have the $200/year power expense of a server, it's still not what you really want. IAX is great but SIP is also a reality, and putting Asterisk into the "just works" category is a really important milestone. One I think that is intended to be improved a lot for 1.6. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
Some NAT problems you can solve, some you never will. Many modern phones have NAT support in them, via STUN, or a static external IP address. Most NATs also offer port forwarding, so you can open a hole for the SIP port in the NAT so all outside can reach it. (With port forwarding, you need a constant address for each SIP phone, so that means either static IP for the phone, or a DHCP server with the ability to always bind a device to the same address - the latter is preferable because you can move your phone to other networks more easily.) Many devices also feature NAT keep alive on the SIP port. That is a must if you can't open ports, but it sure generates a lot of annoying debug output when you turn on sip debug. Nothing beats a permanent NAT entry point though. Some devices, notably Ciscos, just don't support NAT as well. They don't have STUN, and while they may have a static external IP mapping, that's no good if your NAT itself has a dynamic address, as most home broadband NATs do. Asterisk, if you set nat=yes (or often even without that) will take incoming packets from a natted phone, and look at the incoming address, and send back to it regardless of what the phone says in its SIP headers. That's handy, but unfortunately it does not do the same thing for the SDP, so if the phone hands out an SDP with an unreachable address, Asterisk handles it badly. Some SIP gateways are smarter, and if they see an unreachable address in the SDP, ignore it and send to whatever address they get incoming RTP from. You'll have better luck connecting to such endpoints. Many termination providers do this, so you may find your phones can talk to the term provider, but not to other phones on the same * box. Many consumer nats will not hairpin audio. That means if you do all this work to rewrite the addresses in your SIP headers/SDP via STUN so you look like an externally routable device, and Asterisk hooks you up with another device behind your same NAT, you will get one way audio. I get this problem -- I have a * box at one location, with most of the phones (no problem for those) and some other phones at another location behind NAT. These phones can talk to the main location, but not to one another, due to the hairpin. What fun. A new method, called ICE, was drafted a while ago but is getting slow adoption. In ICE, devices are given a list of possible ways they could reach one another (directly, through nats, via RTP forwarders etc.) They try them all and pick the best. In the end it will always work through the RTP forwarders, but that costs bandwidth and latency. So far, however, support is limited. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote: > Brad Templeton wrote: > > > > >For SIP phone calling * box, relay to other * box and out to SIP > >phone, you definitely want SIP all the way. > > > Unless bandwidth between the * servers is a concern, then you're better > off keeping the link between servers as IAX. (preferably trunked) The bandwidth of the audio stream dwarfs the bandwidth of signalling traffic by orders of mangitude. So in fact, I think this is exactly wrong. If bandwidth to or between the servers is a concern, that's where you most want to not be in the audio path. > > It is worth remembering in this sort of setup, often the phones at one > site will not have a route to the phons on the other site, so the calls > wont be re-invited off to the handsets anyway. > If it's phone-on-NAT to phone-on-different-NAT, it typically will not work. That doesn't mean it can't work if bandwidth is important. I think the complete solution, not yet in Asterisk as I understand it is for Asterisk to be aware of both the internal and external addresses of a phone, and to connect internal phones with their internal addresses, but to connect internal phones to external endpoints through their external addresses. Ideally audio never flows through asterisk unless it's doing an IVR dialogue or otherwise explicitly wants it to. (In fact, ideally DTMF goes via SIP INFO or its successors so that Asterisk can listen to the DTMF without being in on the audio.) Flowing audio through your box costs not just bandwidth, it adds latency, and very slight extra risks of packet loss. Latency is the bane of voip calls, it also worsens echo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DiD for less then $4
On Fri, Jan 05, 2007 at 02:29:03PM -0800, CM Rahman wrote: > Anybody selling DID flat rate for less then $4 with sip or iax incoming? Let > me know > > Thanks vbuzzer charges $2 for flat rate DIDs, not quite sure how they do it. However, I have had some clicks and pops of dropped packets with them I don't get with other DID providers, you should check how it does with you. And, worst of all, there is a bug in either their code or asterisk on reinvites. Asterisk hangs up the call after receiving the somewhat unusual reinvite OK that vbuzzer sends. However, if you don't need reinvites (ie. no IVR followed by xfer to phone with reinvite or other xfers) it could be good for you. The other question is, will you do more than 300 minutes (6 hours) incoming a month on average? If not, many people sell DIDs for $1/month and 1 cent/minute. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Best inexpensive home office router for VoIP(QoS with maybe PoE)
On Fri, Jan 05, 2007 at 05:37:22PM -0500, Allen Casteran wrote: > Mike wrote: > >You're quite right, I typed before thinking. Upload is the problem > >anyways, since it usually (in homes) uses much more limited bandwidth > >than downloading does. > > > >No answer to my question though: How do you people handle QoS without > >relying on the phones to do that? I'd like a box that can be purchased > >and installed easily (Linksys type of product) > > > > Mike, > > Unless your ISP specifically supports QOS on your internet connection > there is NO QOS beyond your router. Only within your network will the > QOS be effective. Once the packets go through your router all control is > lost. :) > > This also means that you have little control over the priority of the > traffic coming through the router's WAN port. The most you could do with > QOS in this case is to limit outbound traffic from your PC if it would > interfere with a voice call. The same is not true for the return (ie > inbound) packets. True, but for many people the upstream path is the biggest, and sometimes the only bottleneck in their internet traffic, especially to a good termination provider that has not underprovisioned. So this is the one place QoS can make a difference. For downstream, it can be an issue. Though in theory a clever router can notice the amount of high-priority RTP traffic that is going through, and then cause incoming TCP traffic to back off to leave room for the RTP traffic. I don't know if the cheap boxes do this. The D-link DI-102 qos box literature seems to talk mostly about upstream so I don't think it does this. On the other hand, I tested my wrt54g with qos firmware on, and while downloading at full speed I detected no dropped packets in incoming voice, so perhaps it does that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
On Fri, Jan 05, 2007 at 11:33:02AM +, Gordon Henderson wrote:
> On Thu, 4 Jan 2007, Noah Miller wrote:
>
> >Hi Damon -
> >
> >>Can anyone comment on the overhead added when a SIP call comes into one
> >>asterisk box, is routed to another with IAX instead of SIP, and is then
> >>sent
> >>to the UA from the second box with SIP?
> >>
> >>DTMF passthrough issues?
> >
> >I've got a client with sip phones on several different servers and
> >IAX links between the servers, so I guess that's pretty similar to
> >your setup. I've never bothered to check for overhead since it was
> >never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram,
> >with never more than 3-4 calls going through any one of the IAX
> >links). I can say that DTMF works fine in this setup.
>
> I'm doing the same on 1GHz processors - CPU usage is virtually nil unless
> there's transcoding going on (about 4% per GSM transcode)
>
> ADSL bandwidth is more of a concern for me in these applications )-:
While it would be work to set up, you actually ideally want to
trunk with the same protocol being used by the external phones
or endpoints. When connecting a SIP to SIP call (presuming you
don't have annoying nat problems or have turned canreinvite off)
the audio should go directly from endpoint to endpoint and not
via asterisk.Ditto on IAX to IAX calls.
For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.
In some ways, an ideal solution would have two "trunk" connections
between the boxes (really just two config entries in iax.conf and
sip.conf) and go between the boxes with whatever protocol the
calling channel is using. You could write dialplan scripts to
pull out the channel and choose the right * to * protocol (as
opposed to inter-asterisk protocol which has another meaning.
:-)
It can also be worth having a termination provider that you
can talk to with both IAX and SIP, and sending them the call
with the same protocol the phone used.
Annoyingly, IAX and SIP channels use different interfaces
to provide the address, so you can't do
DIAL(${chantype}/[EMAIL PROTECTED])
A cute patch would be to support that with a consistent syntax over
channels.
Note if you use various flags on Dial which require asterisk
to hear dtmf or do other audio, you are stuck hairpinning.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id ring tones for Asterisk Phone
On Wed, Jan 03, 2007 at 11:15:19PM -0500, Jeronimo Romero wrote: > I'm going to be rolling out asterisk at a small office and one requested > feature was the ability to have a phone that can be configured so that > ringtones can be configured according to the callerid of the caller. > Does anyone have Asterisk experience with such a phone? Any suggestions > would be greatly appreciated. > > Thanks in advance!!! > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Many phones can do this. Some have only a limited set of tones that don't vary much. Most phones can do the basics. Some let you have some uploaded wav file ringtones. A smaller number such as the SNOM phones and a few others can actually be given the URL of an audio file as the ringtone, and the phone will download it and play that. I haven't tried it, but it should be possible on the SNOM to: a) Have festival, cepstral or other TTS turn the caller id into an audio file (ideally cached) b) Put that audio file on a local web server c) Set the URL of the audio file as the ring tone. You usually set the ring tone with the SIP Alert-Info header, however various phones use different syntaxes. Do a search on voip-info for terms like "ringtone" and "alert-info" for instructions on how to set them. Of course, you can also do things like generate the audio and have your computer, or a nearby computer, play the sound so you get it reading the name or number. Or you could generate your own audio files for the people who call you regularly rather than that trick. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 segfaulting when manager client is connected
I was just trying astman with the latest svn trunk from Dec 31. It
connects, but if I attempt to make a call, asterisk segfaults, but
in pthread_kill in /lib/tls/libpthread.so not in the asterisk code.
Is this something others have seen? This is with glibc-2.3.4-2
I just upgraded to 2.3.6 (the lastest for Fedora core 3) and it's
the same.
Not much of a traceback, it's happening here:
static struct eventqent *unref_event(struct eventqent *e)
{
struct eventqent *ret = AST_LIST_NEXT(e, eq_next);
if (ast_atomic_dec_and_test(&e->usecount) && ret)
pthread_kill(accept_thread_ptr, SIGURG);
return ret;
}
Should I file a bug on this? I would presume if it's as trivial
to duplicate as it is for me that others would have seen it.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 200 behind NAT and other xmas woes
On Sat, Dec 23, 2006 at 02:02:18AM -0800, Brad Templeton wrote: > I've tried all the various NAT settings on the SNOM 200 (with > the last firmware rev they made) but reports are that's broken. > The SDPs and Contact headers it sends out are always the natted > address, even if I tell it to use STUN or static or UPNP etc. > If the nat traversal is broken not much I can do on that end. > Well, I tried a bit more, including a numeric STUN server address and a reboot and that seemed to have helped. So now I'm down to the DTMF not working (I've tried both inband and rfc2833 and INFO settings but will try again.) And other issue. I have 5 line buttons set to 5 accounts. Incoming calls to these accounts light the right line button, and make the right ringtone. However, pressing the line buttons to make calls still results in all the calls coming from the first line's identity in the From: header. What's a good way to get the source line to identify on the SNOM 200? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM 200 behind NAT and other xmas woes
I decided to give the whole family IP phones for christmas, all hooked into my asterisk server, so all the nephews can have their own lines. However, one of the phones I got was the SNOM 200. That's worked fine for me on my own network, but I'm having bad luck getting it to work behind NAT talking to Asterisk. It talks to my termination/origination provider, which seems to ruthlessly ignore SDPs and send audio to the address it gets audio from, which works pretty well behind NAT. I've tried all the various NAT settings on the SNOM 200 (with the last firmware rev they made) but reports are that's broken. The SDPs and Contact headers it sends out are always the natted address, even if I tell it to use STUN or static or UPNP etc. If the nat traversal is broken not much I can do on that end. Asterisk, on the other hand, should be handling this with nat=yes on the channel, but it's not. It handles it for the SIP packets, responding to those on the address the requests came from (ignoring the contact header) but it seems to accept the SDP, which contains and address Asterisk can't see. The docs say nat=yes will fix addresses in SDPs. I'm running svn trunk from a few days ago. Is there a way to get Asterisk to send audio to the address the incoming audio comes from, or to take the SDP and replace the IP address in it with the IP address the SIP came from? Otherwise while the phone will be able to make PSTN calls, it is unable to call Asterisk for voicemail and the rest. -- Some other issues: Anybody tried to use vbuzzer with Asterisk with an IVR on the DID? I find when I do this, after the IVR connext to an extension, the reinvite Asterisk sends to vbuzzer is responded to by a very simplified 200 state response which Asterisk seems to get upset at. Asterisk immediately hangs up the channel, but nothing in the debug logs (even level 9) seems to say why. I will use another DID for now. Also, I bought an SPA-2000 on ebay that claimed to be unlocked. It isn't. I will probably get my money back but that's not going to help at christmas. Anybody know a way to unlock these things at the hardware level (shorting pins etc.) The factory reset codes all demand a PW, which means it's locked. Web access disabled too. I have a Cisco ATA 186 but they only do NAT traversal with a static address, no STUN etc. I may end up setting up the SNOM and replacing it if no other solution shows up. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial own extension to get to voicemail.
On Wed, Dec 20, 2006 at 02:34:36PM -0500, Phil Finkler wrote:
> I've gotten this Polycom 501 pretty much licked, but I need to know if
> there's a way in a dialplan to say if someone dials their own extension
> it goes straight to voicemail and asks them for their password. I
> thought I saw an example of this on the web but I can't seem to find it.
> Any advice appreciated!
>
You can do it, but it's more work than having an extension (the standard one
seems to be 86 now) that goes to:
VoicemailMain(s${CALLERID(num)[EMAIL PROTECTED]);
(But only in a context where the callerid can be trusted.)
To do what you want, you would need to have your extension processing
macro test if CALLERID(num) = ${EXTEN}, and then invoke the above
expression instead of dialing the extension.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco devices (without STUN) and dynamic NAT
Cisco devices (7912, ata-168, 7960 etc.) don't support STUN. However, they do let you define a static external NAT IP address, and parameters to send a keep-alive out through the NAT on a regular basis. However, I want to make these devices work in an environment where they are behind a NAT which has a dynamic IP which might change (though in fact it changes only rarely.) They're talking to an external Asterisk server which is of course not behind NAT. The docs say that nat=yes will cause Asterisk to ignore the IPs in the SIP headers and SDP, and replace them with the actual address the packets are coming from. I thought this meant that if I put any address in the Cisco's NATIP field, this would work because Asterisk would rewrite the SDP to the real address, which might have changed since the NATIP field was set. However, Asterisk is not doing this (though it's doing some other interesting things, now noticing that the NAT address in the Via header is local to it and talking directly to that) and it's trying to do native bridged channels to other devices, which isn't working with the wrong address in the SDP. I want native bridging of course, in fact it's a must if you have a phone on the east coast, an asterisk server on the west coast and a SIP terminator in the middle. No way you want to hairpin the audio, and with STUN supporting SIP phones this works fine. canreinvite is not involved here because there is no need for reinvite on a simple call. I'm using svn trunk, as of a few days ago. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT and Dial to two channels at once
On Mon, Dec 11, 2006 at 06:48:18PM -0500, Andrew Joakimsen wrote: > You need to understand how NAT works, if you can chan2 and chan2 is behind a > NAT and suddenly someone else is invited to chan2's IP address port 5060 > chan2's router willl say "WTF I dont have an estabished connection on port > 5060" (to the client being reinvited to chan2) and it wont work. You need to > have the media path go through asterisk in that case. Actually, it's more complex than that. If the NAT box has had a hole poked (in its config) for the RTP port (SIP port is only used by Asterisk) then any machine can send it RTP on that port. In addition, if the NAT is of the "full cone" type, any host can send to your port once you have sent a packet out that port. With Restricted cone and Port restricted cones, it also works as long as the Natted IP phone is sending packets out to the other host already. Which it should be if we have symmetric RTP. Symmetric NATs, which are rare, will change the port number when they start talking to a different host for RTP. This will screw up all but the cleverest implementations. (Though there are endpoints that notice if the RTP is coming from a port other than they were told, and start sending to that instead of the one in the SDP) What doesn't work is assymetric RTP with NAT. In this case we have the audio going through asterisk in one direction, and directly in the other direction. That will fail if the direct direction tries to go into a nat (it should work if it's only leaving a nat) Asterisk currently does assymetric RTP if it thinks it only has to listen to one end of the audio path. That's a good idea in general -- but not one that works through anything but a manually opened NAT. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any possibility of Vonage Integration
On Wed, Dec 06, 2006 at 08:13:00AM -0500, Paul wrote: > Also, I should have mentioned that many of these providers advertise > "business" plans on their website. How can anyone honestly advertise > phone, fax, email hosting, web hosting, etc. to the business community > without 24/7 support? I like 24/7 support, but I would have to guess that most businesses would be mostly interested in support during working hours (which is more like 6am to 11pm for most companies.) Not that there aren't employees around at night sometimes, but I'm just talking about what's most important. I think with any company, Vonage included, it is good to have a redundant backup, for when their network or your network is down. Be it a PSTN line or a cell phone. Some companies offer PSTN failover on DIDs, which I think is a good idea. Works at least if your equipment, or their middle equipment is down but doesn't work if the PSTN failover equipment itself is down. Vonage does offer PSTN failover if your ATA is not responding. But having an FXO box talk to your Vonage ATA is just nuts. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP firmware for Siemens Optipoint 410 Economy?
I have not seen anybody on the web to have found this so I thought I would check here. Anybody got this firmware? I've found firmware for the 400, but it doesn't seem to load in the 410. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any possibility of Vonage Integration
On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: > And if you get someone over at Vonage that knows that to do you can > connect without the FXO > It is like FWD you have to get the KEY from Vonage for this to work. > And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Fri, Dec 01, 2006 at 07:37:35PM -0700, Ken Williams wrote: > I was able to set a program to speed dial the park extension. Then a user > just hits TNFR followed by the line I've programmed to speed dial park. > > If you get the HOLD button to do this, I'd love to hear how :). Oh, that would require new code in Asterisk, a new commmand that is able to get all channels that are currently on hold, and connect to one if only one or give a menu and connect to one if more than one. Don't know if it would require any fancy changes to holding itself. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Fri, Dec 01, 2006 at 04:55:51PM -0500, John Novack wrote: > In most hybrid business systems one does NOT place a call on hold, but > begins a transfer, either a specific function button or intercom button > which automatically places the call on hold, gives a new dialtone and > another extension is dialed. IF the called party answers, the > transferrer can announce the call, and if the called party wants to > accept the call, they simply hang up. Blind transfer is done the same Alas, it's not possible in these days to make it that simple. Today, almost all calls will be answered (by a voice mail) if not by the person. So you need an additional UI for attended transfer, which allows you to say "No, I got the voice mail, disconnect the voice mail and bring me back to my call." I guess you needed that for endless ring in the old days too. You're right that there is no big difference between attended and unattended in the UI when it works, but when the attended call "fails" with voicemail or unlimited ring, or for that matter busy signal, you need a means to go back to the caller. That's one thing soft buttons are good for, you can create soft buttons for specialty functions like this. If you have "line" buttons on your phone, normally the original call is on one line button, and the 2nd call on another line button, so you just press the first line button to abandon the call attempt. On my Asterisk system, I have done another thing which is handy. My extension macro looks at the caller-id. Calls within the house do not ever go to voicemail. Calls from outside (including ones transferred) will go to voicemail after the timeout. So I never get voicemail but I do get endless ring. Many PBXs also offered a feature that if you blind transfer, and the call goes into endless ring that it transfers back to you after some timeout. Today, voicemail has largely eliminated that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Thu, Nov 30, 2006 at 02:50:21PM -0500, Tom Rymes wrote: > for example: In your example above where they can't figure out how to > transfer, why don't you edit features.conf and define the transfer > key as # or something. Then, when they have a call for "Bill" across > they way, they can do this: In this case don't they need to have a t in every Dial as well? And then there's the other direction. Sometimes (actually quite often) I like to transfer a call that I dialed, which requires the "T" but means you interfere with typing touch tones to IVRs that you call. No, almost all IP phones have a transfer button, the nice thing would be if somehow the UI for that could have been standardized, at least for the phones that don't have screens and soft buttons (which can extend the interface because they can show it to you). This is not generally Asterik's fault, of course. PBX interfaces are, as I said, notorious. Most users have no idea how to use most of the featurs on the PBXs they use, and a disturbing number don't even know how to transfer unless they do it frequently. That's where screen phones (or computer pop-ups) are a win. There are too many PBX interfaces that can't be improved. The nice thing about hold => park is that it is darn simple. The only thing it doesn't do is easily let you park on an extension whose number you don't know. The old parking lot can exist for that, I guess. Or just write the extension on the phone, or have an extension you can call that tells you your extension. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Thu, Nov 30, 2006 at 12:03:24AM -0600, Lacy Moore - Aspendora wrote: > The question is what is the best interface? On our old system, we put the > caller on hold, went to another phone, pressed pickup and then entered the > extension where the call is on hold. I never liked that, especially if I > was at an extension that wasn't mine. By the time, I got to where I needed > to be, or someone called me and told me to pick that call, I would forget > what extension. The same thing, I believe, will happen with the current > park method. I don't know what would help with that, maybe better vitamins > to prevent memory loss? :-) I don't know. Maybe a receptionist console > that could tell who is on park, their phone number and caller id info along > with who put them on park? If you integrate with the voice mail, so that you can pull a user's audio name for an extension, the pickup extension can say "Do you want to pick up the call put on hold by 'Lacy Moore' or 'Joe Smith' or 'waiting room' or 'extension 242'" Hopefully little need for memory. > > I'm wondering if maybe we are looking at having to have different ways of > doing it. Being able to transfer the call to a line button, and being able > to press that line button to pick up the call, and having the status shown, > may be the better solution for small companies. Problem there is only some phones have line buttons, and when they have them they are scarce and there's many things you might like to do with them, and dedicating them to this would be low on my list. Dedicating one speed dial to a "pickup call" command that picks up the solo call or reads you the names/numbers of the calls on hold, or puts them on your screen if you have a screen -- that makes more sense, and it does well on every phone. Then if you want to have line buttons which read hints based on the number of calls held. > > I'm going to show my ignorance here. Since the phone displays the number we > dialed,or the incoming caller information on the screen (we're talking those > with displays), is there anyway to have it so that when the call is parked, > it also shows the parking spot the caller is parked on? Kind of like hold > does now? I know nothing about the SIP protocol, so I don't know if this is > possible or not. Yes, some phones can receive text messages back from the server. Not all of them. But if you have a system where "parking" is just pressing hold, then all you need to know at worst is the name or extension of the phone you're on, and that's usualy already on the phone screen or even written on in pen! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Port 5060
On Wed, Nov 29, 2006 at 04:49:38PM -0700, Joseph wrote: > What I have is that each device is listening on different port ex. > > [pstn-5665] ; incoming/outgoing calls on FXO port > type=friend > ... > port=5066 ; port on Pstn line > ... > > [318] ; incoming/outgoing calls on FXS Sipura-2002 > type=friend > ... > port=5069 ; port on FXS line > ... > > etc. > > Though I'm not sure if bindport will work, try it! > My understanding was that the "port=" field on a particular SIP channel defines the port used at the remote end, ie. The user's phone will be talking on port X of their IP address, it does not alter what SIP port Asterisk is listening on on the Asterisk box. That is what bindport does, and that's a global setting, I was not aware you could have multiple bindports but that is very useful if it works. The idea of having a different bindport per channel would be handy too as yet another means of identification. There is some merit in not using port 5060 for your bindport, though it comes at a cost. a) If you use another port, all clients must be configured with a slightly more complex URL for you including a port b) A few broken clients can't call you at all because they don't let you set the port (mythphone is one example.) but... a) You might get around carrier SIP blocking b) If there is ever a virus/DOS aimed at SIP devices or asterisk, you may well escape it. No guarantees becuase many mal-tools will scan for other ports but it's more work for them. c) If you want to run more than one instance it is the only way ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Wed, Nov 29, 2006 at 06:05:31PM -0500, Steve Sobol wrote: > On Mon, 27 Nov 2006, Brad Templeton wrote: > > > On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote: > > > Can you explain how ValetParking and twenty minutes worth of "dialplan > > > creativitiy" can't do the same EXACT thing you are describing? Sometimes > > > the > > > simplest answer is never the most obvious > > > > Yeah. With valet parking (or any parking) you have to explicitly park > > your call. With what I propose, or with SLA, or with many key systems > > or simple multiline phones, all you do is put the call on hold, and > > that makes it possible to grab it from elsewhere. > > Back on track... > > OK. I understand that "press the hold button" won't do what I want. > > The next best thing is ... instead of using the built-in call parking > feature, where the call gets parked at a random extension, I need to be > able to park calls from extension X at a specific other extension Y. > > Parking at a random extension, then picking it up, is fine if there's ever > only one call on hold, but I expect that there will be times where I need > to have more than one call on hold. We have four DID lines, all plugged > into our Asterisk server, and we do a lot of telephone support. :) > You could pull this off in a small system because the parking lot is big enough, I think with the valet parking add on. You can create a parking slot for each extension, and then grab them with a special extension. So if your extensions are say 30 through 39 you could use the valet add-on to always park extension 32 in slot 732, and then you could always pick it up that way. But I'm hoping that a better interface than lots can be devised with time. Lots are useful if you are a receptionist having to handle tons of calls, put more than one call on hold at once and pick them up in different unpredicted places. But otherwise their UI is not particularly good for ease of use. The best interface, I think, is either hold implicitly becomes pickupable after 5 seconds, or if you want a transfer, to have a parking lot you blind xfer to because you don't need to listen to the slot number, you just press a single button called park if possible. Let's face it, PBXs are notorious, even with new fancy screen phones, at being hard to use, and the UIs not remembered by their users, and also varying from phone to phone. There's much room for improvement in all PBX interfaces. Not just Asterisk. To my mind every PBX might do well just to have a single function button on the phone which causes a dialog box to pop up on the PC next to the phone, and do it all through that (though possibly by pushing buttons on the phone in most cases instead of the keyboard if you like.) A UI with clear buttons, grayed out buttons for what you can't do, help screens, warnings where appropriate, etc. SIP didn't go that way, it went towards the phone being full of features itself. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Mon, Nov 27, 2006 at 11:20:27PM -0500, Andrew Joakimsen wrote: > Can you explain how ValetParking and twenty minutes worth of "dialplan > creativitiy" can't do the same EXACT thing you are describing? Sometimes the > simplest answer is never the most obvious Yeah. With valet parking (or any parking) you have to explicitly park your call. With what I propose, or with SLA, or with many key systems or simple multiline phones, all you do is put the call on hold, and that makes it possible to grab it from elsewhere. Frankly, call park by transfer requires that the user be comfortable with transfer. You may be aware that quite often they aren't, and in fact, transfer doesn't always even work on some phones or takes work to get going. On some phones it's #700. On some it's "hit xfer, dial 700, hit xfer" On some it's put on hold, then dial, then hit xfer. Etc. etc. almost all phones however, that can put on hold. do so with a single well marked button, same UI everywhere. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Mon, Nov 27, 2006 at 04:05:34AM -0800, Steve Langstaff wrote: > > > > What I describe is different. There are no shared lines, but if > > you put a call on hold on one phone on a non-shared line you > > can go to another -- any other in the pickup group, whether > > it is registered to have the shared line or not, and pick it > > up, as you can (in a more cumbersome way) with call parking. > > I'm a bit unclear on where you say 'push a button to pick it up' > - does this mean that there can be only one held call 'shared' between > the extensions, or is there some logic somewhere that 'knows' which held > call should be picked up when you press the button? In my view of the SOHO environment, you would put a call on hold. That's one button on most phones. At the target phone, a speed-dial button would be configured to call the "pick up held call" extension. If there is only one call on hold in the pickup group of that extension, it would simply connect that call. Let's say the pickup extension is 600. If there were more than one held call (quite rare in a home PBX) it would instead say, "There are calls held for extensions 123 and 456. Please enter the extension you wish (followed by pound sign if there are ambiguous patterns like extension 22 and 222 but hopefully nobody does that! Otherwise you have to wait a few seconds after 22 is pressed.) In addition, you could also define so that extensions of the form 600xxx are a hard pickup of a call held by extension xxx. Thus if you want to be more reliable, or have a speed dial aimed at picking up only a very specific extension with no chance of a menu, you could do that. I would implement this with a PickupHeld command, which can take an extension argument, or no extension (meaning pickup any or give menu), or possibly a pickup group argument so dialplans could allow pickup from other pickup groups if you want to allow that for security reasons. Anyway, for the user, the UI is very, very simple, especially in a SOHO where mostly we're talking one call on hold at a time. For security reasons, as noted, you would not be able to pick up a call that was just put on hold in the last few seconds. And an extension could define if it wanted that calls it puts on hold are not available for remote pickup to avoid any risk of accidental pickup. Even then, they might not want to use the parking lot system (which has no real security) but just have to do an explicit transfer (like the xfer to 700, but no need to wait for a number to remember) For a large PBX serving several offices, you might want to expand the pickup group concept to have a "master" grouping, or allow pickup group numbers to be versioned. Ie. If I say "pickupgroup=2.1" then "2" would define my company, and the 1 would be the traditional pickup group function within the company. Just a thought on a good UI. Obviously what's really more valauable is code. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Mon, Nov 27, 2006 at 01:46:58AM -0800, Steve Langstaff wrote: > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Brad Templeton > > Sent: 25 November 2006 21:02 > > [snip] > > > ...the UI I think most people want, which is, just put > > the call on hold, and go somewhere else and push a button to > > pick it up. > > > > That is not only an eaiser interface, it's actually a more > > powerful one, because it gives you the ability to put a call > > on hold to go away to check on something, and then decided > > after the fact that you > > want to pick it up from another extension. Indeed, you could create > > an interface if you wanted to so that you could pick up the > > call from any pstn phone (ie. cell phone) by dialing a magic > > number and entering a code, without having decided to > > explicitly park it first. > > I think that you are describing shared line appearances. I don't believe so, not as I understand them, but perhaps they are planned to be different from my conception in Asterisk. My understanding of a shared line is it mimics the traditional analog phones, a line is shared on many phones, calls ring on all phones, if you put on hold on one phone and pick up on another it works. What I describe is different. There are no shared lines, but if you put a call on hold on one phone on a non-shared line you can go to another -- any other in the pickup group, whether it is registered to have the shared line or not, and pick it up, as you can (in a more cumbersome way) with call parking. Or do I have it wrong, and with SLAs all lines will effectively be shared, and accessible from every phone? Both SLAs and what I describe allow you to do a very quick move from one extension to another by putting a call on hold and picking it up. However, what I describe is far more general.In some systems, shared line also includes multiple extensions ringing for a call on the line (already possible with Dial to multiple channels). It also means barging, in that I can pick up a shared line that is already in use and I join the conversation. The latter is done by shared lines and not by what I propose. Some PBXs have a barging system where you can say, "Barge in on the call on extension xxx" (this needs permissions typically.) This is also more general but not as intuitive as shared line if people all have phones with line appearances. In my home, for example, we really just have two people. There is no desire for a shared line. However, there is desire for a phone in the living room that rings whether her office line is being called or my office line. And there is desire for very simple call park, so I can pick up the call on the living room phone, and quickly put it on hold and go up to my office and pick it up quickly without the annoyance of current parking. I could do much of this with shared lines as I understand them but not in as nice a way. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Sat, Nov 25, 2006 at 07:17:47AM -0500, marvin horst wrote:
> The valet system gets us partway from what I read, but it still uses the
> >arbitrary number slots. It still requires the user know to transfer a
> >call to the valet.
> >
> >no you can park to a specific number (lotname)
>
> exten =>
> _6XX,1,ValetParkCall(auto|8${EXTEN:1:2}|180|${CALLEDEXTEN}|1|internal)
>
> ; Valet unpark from an extension
> exten => _5XX,1,Playback(beep)
> exten => _5XX,n,ValetUnParkCall(filo|8${EXTEN:1:2})
>
Definitely better, and I will install this add-on, but still not at the
UI I think most people want, which is, just put the call on hold, and
go somewhere else and push a button to pick it up.
That is not only an eaiser interface, it's actually a more powerful
one, because it gives you the ability to put a call on hold to go
away to check on something, and then decided after the fact that you
want to pick it up from another extension. Indeed, you could create
an interface if you wanted to so that you could pick up the call from
any pstn phone (ie. cell phone) by dialing a magic number and entering
a code, without having decided to explicitly park it first.
The other reason it's superior is that call transfer differs on various
phones, and sometimes transfer to the parking lot doesn't work right,
it's one more thing to go wrong. However, almost all phones have the
same interface for putting a call on hold (a hold button) and it is
more likely to work.
The only downside to the implicit park UI is that somebody else can grab a call
you put on hold that you didn't intend to park. That's not an issue
in a house or small office, though. And it's low risk. For example,
you can wait 5 seconds to put a call into implicit park so that if you
are putting them on hold to do attended transfer, this can be spotted
so that there is no implicit park. Implicit park would require you put
the call on hold and do nothing do it for some amount of time if you
like.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Wed, Nov 22, 2006 at 04:51:26PM -0800, Ira wrote: > At 03:14 PM 11/22/2006, you wrote: > >The missing piece of the puzzle: I'm extension 203. I want any call I park > >to get parked at extension 2203. I want a call my boss parks to park at > >2205, since he's ext. 205. In other words, I want calls parked FROM > >extension XYZ to be parked AT extension (XYZ+2000). > > > >I don't see a way to force parked calls to a specific extension. I'm > >probably just missing the answer, but I've googled for it and I can't find > >it. > > > That doesn't seem to be the way parking was designed. It's a first > available distribution of a series of numbers you choose. The problem > with your plan is that it can't handle a second call on an extension. > Coming up in V1.4 is something called SLA or shared line appearance > which might do what you want depending upon how it's implemented. For > the moment you just need to tell people extension to pick up to > retrieve a parked call. Here it's always 701 as we've never yet As I was noting in an earlier message, the parking lot concept is to my view not a thrilling interface at best, and I can't see many times one would want it in a SOHO environment.It seems best for a large PBX where people are moving to random places to pick up calls, and many calls may be parked at any given time. For many people, a far simpler interface is to just put the call on hold -- by pressing just one hold button, and then go pick it up as easily as possible somewhere else.Shared line systems help to do that but from a different direction. The parking lot approach has you remember a somewhat random number told to you, and then to go dial it.People can remember their own extension much more easily, so one good interface in that case is a way to dial a number "NNN" to pick up a call held on a specific extension (in my pickup group). Or more simply, to dial the pickup number, and if there is only one call on hold, it gives it to you, and if there is more than one, it lets you dial the extension that put it on hold and reads the extensions that have calls on hold to remind you. This is a better interface in an environment were the small security risk here is minimal, such as a home or small office. The nice thing about this interface is that a phone speed-dial function button can be programmed to the pickup number. This means that parking and getting a call can amount to pressing one button to put the call on hold, moving to another phone and pushing another button to get the call, which is about the simplest interface and the one found on key systems and some pbx. Where security is a concern (and the current call parking lot does not actually provide a great deal) you can have a call transferred to a valet, but not require the user to remember a parking lot number if they know the number of the extension that put the call on hold. The valet system gets us partway from what I read, but it still uses the arbitrary number slots. It still requires the user know to transfer a call to the valet. Of course if you know what phone you are going to, you can just do unattended xfer to it, as long as there is not too short a voicemail timeout. But again that's a way more complex interface than "push hold" and "push pickup." ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why Aastra uses 48V whereas other IP Phones usemuch less, i.e. 5-12V
On Wed, Nov 22, 2006 at 11:29:01PM -0500, Michelle Dupuis wrote: > 48VDC is a long time telco standard - and has become the Power over Ethernet > standard. > > Keep in mind that 'electricity' isn't the measure - it's power. Power is > not synonymous with voltage. More to the point, there is a tradeoff. For a given power, the higher the voltage, the lower the current. The lower the current, the thinner the wire you can get away with. Power over ethernet uses very thin wire, so you want high voltage and low current. Power transmission lines use very high voltage because they need (comparatively) low current through the wires. The higher the voltage, the more power you can put through the same wire. To a point. As voltage gets higher, it also gets more dangerous, and needs a bit more insulation. It's very hard to hurt somebody with 12 volts. And 48 volts, while not quite as safe, is still pretty safe. It's been chosen as a voltage that mixes the right combination of safety and power. The higher the voltage, the more heat you can generate if you have the current behind it. (If you are current limited or fuse/breaker protected you are just as safe from fire if things are calibrated right.) In the past, we often drove things with batteries, or wanted to sometimes. Getting 48v with batteries takes a lot of cells with most technologies. Phone central offices had big banks of batteries -- no problem. Today, with advanced switched-mode power supply technology, we can turn just about any voltage into any voltage. So we don't care as much about being able to run on batteries as low voltage, though it's still nice in portable tech. And of course the chips all run on very low voltages today (TTL was 5 volts and it's getting rarer) and they want to be low power.Most of the PoE phones that take 48 volts are converting it down to lower voltages to use. But 48 is a good voltage to be sending on the wires. The USA uses 120v for house current. That's enough to hurt you and can kill you if you touch it wrong, though I've touched it a few times. A lot of the world uses 220. This causes enough of a spark that they require all receptacles to have a switch on them so you don't plug things in live. On the other hand, 220 can deliver twice the power in the same current. Kettles in the 220 world are _really_ fast. Your dryer and oven run on 220 even in the 110 world, only way to get enough power. Same with electric car chargers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Powering SNOM 200 phones?
A follow up on my message about my SNOM 200 phones now powering from my 802.3af Netgear FS108p PoE box. To follow up for those finding this thread on searches... I purchased some PowerDSine 6001 units (very cheap on ebay) and they power the SNOM 200 fine. Some Buffalo units also did this. So it seems that either the Netgear is too picky about its detection, or the SNOM 200 not fully compliant. The powerdsines are big and require an extra cable as all external injectors will, but they work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call park on Linksys 922 and similar phones?
I'm having an issue with call park on my new Linksys 922. It has soft menu keys for doing call transfer (which I always think is a good idea because it's amazing how every phone has a different xfer interface and people always get confused). However, I can't get a good call park working on it. It doesn't respond to the use of "#" for transfer (nor should I want it to, since it has soft transfer keys). If I hit xfer and call 700, the parker does announce the call being parked at 701, but then instead of disconnecting me I hear hold music on the 722 (and continue to hear hold music on the calling phone.) If I hit resume, I am back talking to the calling phone. If I hit xfer again (which is normally how to complete a transfer) both phones disconnect, and the console says that the 922 "got tired of parking." --- I must admit, on a side note, I have never been particularly happy with the parking interface. I know a number of other people feel the same since there have been calls and development efforts for ways to improve it, including hints for BLF, shared/bridged line functionality etc. For the SOHO application, ie. a home pbx, the idea of a parking lot with numbered slots is generally overkill. Such a home is extremely unlikely to ever have more than one call parked in a pickup group, or per PBX frankly. I think a much nicer interface would be to have the first phone simply put the call on hold (which is the typical approach in many key systems) and then dial an extension to "pick up the call that's on hold" in my pickup group. If, as will rarely be the case, more than one call is on hold, I think the best way to deal with it would be to present an IVR that says: "3 Calls are on hold. Please enter the extension that placed the call on hold. Available extensions are 305, 49 and 902." But 99 times out of 100, the interface would amount to putting the call on hold, going to another phone and hitting the "pick up held call" speed dial. Which is what people tend to like in SOHO settings. You can sort of do this if you just insist there is only one parking slot, but it won't handle the rare double-hold case and it's much more to do when putting the call on hold. Any effort been made in this direction? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Powering SNOM 200 phones?
On Fri, Nov 10, 2006 at 02:34:31PM +1100, Paul Hales wrote: > > I had a 200, and it worked fine with POE. > > The standard power connector was the RJ-11 style as mentioned below. > Weird item that one. > > The successor to the 200, known as a 190 does NOT support poe, while the > 320 does. > Yeah, these have an extra unmarked rj-11 on the bottom next to two covered holes (with nothing but pc board behind) where the ethernet would be on the old model of snom 200 if I read the manual right. So that's the power. So I guess the only way to find out if they just don't talk to my netgear POE (which does power my grandstream 2000) is to find different POEs. Or buy the power supplies which don't seem to be very expensive -- or are there different models of snom power supplies? It is suggested the 190 takes 5v, not 48v. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Powering SNOM 200 phones?
Ok, not exactly an Asterisk problem, but... I picked up some SNOM 200 phones because SNOM's have been recommended for use with Asterisk and they have line buttons that can subscribe to presence. However, they don't appear to power up when connected to my Negear FS108P, which is an 802.3af Power-over-ethernet capable hub. I am pretty sure these are the SNOM 200b, in that the ethernet connectors are at the back rather than on the bottom, and there doesn't even seem to be a jack for plugging in any other kind of power adapter (and I don't have another one.) Anybody had experience with these phones and powering them? Is it just an icompatability with the Netgear, or do I have 2 dead phones? Would getting a different PoE box be a good idea? (Frys has the airlink for $29 from time to time, which is a great price. Otherwise many older PoE boxes tend to cost more than the modern cheaper phones they might power.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Java Web Phone
On Thu, Nov 02, 2006 at 11:23:08AM -0500, Guillermo Salas M. wrote: > On Wed, 2006-11-01 at 16:05 -0500, Vladimir Montealegre Estailes wrote: > > Hello list partners > > > > you know about a softphone made in java attachable in a web page? > > > > GNU! > > > > > I'm using JIAXClient [1] to permit to any user to join one meetme room > [2] with the IAX2 protocol, works very great for me, and is very easy to > install and modify to your needs. > > > > [1] http://www.hem.za.org/jiaxclient/ > [2] http://www.rmsenecuador.info/jiaxclient/index.html Useful, but it requires a signed client with permissions to install a DLL. Once you are able to do that, you can do anything -- install a full softphone, send the voice to any destination (not just back to the server) -- and of course take over the other person's machine so they should be very wary of approving the client. What would be more useful is an applet that can run as a pure applet. That's forced to only talk to the server that served it, but it doesn't need approval. Or, in theory the voip client going into the new flashplayer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk both behind a NAT and outside at the same time
On Wed, Nov 01, 2006 at 06:16:25PM +0100, Benny Amorsen wrote: > >>>>> "BT" == Brad Templeton <[EMAIL PROTECTED]> writes: > > BT> The correct behaviour, as I see it is: > > BT> a) Native bridge when connecting two external channels -- > BT> everybody is on the real internet b) Native bridge when connecting > BT> two internal channels -- everybody is on the 192.168.* network c) > BT> Route RTP through Asterisk when connecting internal and external > BT> d) When a channel is to a device behind a remote NAT, the usual > BT> rules apply (either use STUN or other smart NAT, or route RTP > BT> through Asterisk) > > You won't get asterisk to do what you want. That kind of logic simply > isn't implemented, and no amount of fiddling with configuration files > will make it happen. > > I'm sure patches are welcome. Thanks. Will look into it. Probably need to switch to 1.4 before I start writing more patches though. Though to my surprise I am now discovering something worse. It doesn't seem to work in the lastest 1.2 even with canreinvite=no and nat=yes on the natted (internal) phone with a connection coming in from outside.The outsider has to presume it's calling a natted phone rather than a non-natted asterisk, the invalid SDP is leaking out. I'll see if I can pin that down a bit better. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Opinions on the best wholesale origination/term providers
On Tue, Oct 31, 2006 at 08:03:56PM -0800, Martin Joseph wrote: > On 2006-10-31 17:29:47 -0800, Brad Templeton <[EMAIL PROTECTED]> > said: > > > > >I've been losing patience with my current provider, a small company > >called Sellvoip. Their termination is good, and they are > >asterisk based, but they are understaffed and have no concept > >of customer service. So I'm shopping. > I also use Sellvoip and I am close to them (Seattle). They by FAR > produce the best call quality for me, when compared to nufone and > Teliax, although both of those companies do ok, my routes to them > aren't nearly as clean. > > I recommend Teliax for good support. > Their DIDs ($5/month plus 2 cents/minute) are much too high, their termination is 2 cents which is tolerable but in general too high for a wholesale service. But thanks for the comment. The sellvoip guys (guy?) are indeed producing good quality. Another thing they are doing, which I really like, is processing termination quickly, in that when I do the invite it's ringing within a fraction of a second. A few other termination providers I have tried are taking 3-4 seconds to ring after invite. You thought I wrote a lot and I didn't even put that on my list. We just have to convince Jed at Sellvoip to hire some some support techs, even if he has to add a couple of tenths per minute. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
