RE: [asterisk-users] Asterisk servers being greedy and not letting goof the media path. (using IAX2 channels)
For the benefit of the archives, my problem was a simple one. I hadn't forwarded the IAX port on the router of the remote * server connection, so when voip provider was trying to connect directly to the remote * server, it couldn't. Hurray for wasting an entire day over a simple silly little thing ;-) Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mat Stace Sent: 06 November 2006 17:42 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk servers being greedy and not letting goof the media path. (using IAX2 channels) Evening everyone (obviously depends on when you're readin this, but hey). I'm trying to set up a multi * server situation, and am falling over at the second server, and after a day of google etc, have come up against somewhat of a brick wall. I can make calls each way between the two servers no problem, and can include the required extension at the remote * server as part of my main incoming dialplan. My problem comes with * attempting to pass the media path to the other server. What is happening is: Incoming call from iax2 provider to main * server -- dial sip extension on main * server -- setup IAX2 channel to remote * server (which then rings extension) Pickup call on extension on remote * server -- main server sip extension stops ringing -- ast console on main server I get : - -- Attempting native bridge of IAX2/voipprovider/6 and IAX2/remote*server/7 -- Channel 'IAX2/voipprovider/6' unable to transfer -- Channel 'IAX2/remote*server/7' unable to transfer - In the user/friend declarations (user for incoming voip provider, friend for remote * server) in the two iax.conf files I have notransfer=no, and also up in the [general] section of the iax.conf. The problem is that when remote * user answers the phone, and then transfers the call to an extension on the main * server, there is massive (ie 2 seconds) delay, and using IAX2 show channels at the two consoles, the call is doing the following: PSTN - VOIP PROVIDER - main * server - remote * server - main * server - SIP extension on main * server. Anyone have any ideas on how to make the * servers give up the media path? Cheers Mat ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.13.28/518 - Release Date: 04/11/2006 17:30 ___ --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 servers being greedy and not letting go of the media path. (using IAX2 channels)
Evening everyone (obviously depends on when you're readin this, but hey). I'm trying to set up a multi * server situation, and am falling over at the second server, and after a day of google etc, have come up against somewhat of a brick wall. I can make calls each way between the two servers no problem, and can include the required extension at the remote * server as part of my main incoming dialplan. My problem comes with * attempting to pass the media path to the other server. What is happening is: Incoming call from iax2 provider to main * server -- dial sip extension on main * server -- setup IAX2 channel to remote * server (which then rings extension) Pickup call on extension on remote * server -- main server sip extension stops ringing -- ast console on main server I get : - -- Attempting native bridge of IAX2/voipprovider/6 and IAX2/remote*server/7 -- Channel 'IAX2/voipprovider/6' unable to transfer -- Channel 'IAX2/remote*server/7' unable to transfer - In the user/friend declarations (user for incoming voip provider, friend for remote * server) in the two iax.conf files I have notransfer=no, and also up in the [general] section of the iax.conf. The problem is that when remote * user answers the phone, and then transfers the call to an extension on the main * server, there is massive (ie 2 seconds) delay, and using IAX2 show channels at the two consoles, the call is doing the following: PSTN - VOIP PROVIDER - main * server - remote * server - main * server - SIP extension on main * server. Anyone have any ideas on how to make the * servers give up the media path? Cheers Mat ___ --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] freepbx dial plan, add and remove at the same time
Hi Mike,
It's a while since I did this one myself, but I was doing the exact same
thing when using voipbuster (or whichever of it's sisters services I was
using at the time).
I'm thinking that in the dial command you want
+44{EXTEN:1}
HTH,
Mat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mike Williams
Sent: 22 September 2006 10:32
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] freepbx dial plan, add and remove
at the same time
Hi,
I'm try to setup a dial plan in freepbx to work properly with
ENUM lookups. However, the only example I can find that works
in the UK is somewhat complex.
(http://www.voipuser.org/forum_topic_6651.html)
Basically, it has 3 outbound routes (local, national,
internation) to strip
certain leading digits in a specific order, before a trunk
does some more
work.
I got very close to doing it with a single outbound route
(the default, strip
the 9, pass the rest) and a single trunk.
Where I got stuck was changing 01234567890 into 441234567890.
I did see this example:
61+0|NXXX
Which to me suggests it will add 61 and strip a leading 0,
but either way
round it didn't work (even with the correct 10 digits).
Can a dial plan infact add and remove numbers at the same
time? If so, how?
Asterisk 1.2.11, FreePBX 2.1.2.
Thanks
--
Mike Williams
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RE: [asterisk-users] freepbx dial plan, add and remove at the same time
I can have a go at explaining.
I've had a quick dig through my extensions.conf, and I've got it in an
outgoing sipgate dial command.
exten = _0.,1,Dial(SIP/+44${EXTEN:[EMAIL PROTECTED],30,t)
What it does is in the dial command, it sends +44, then the extension which
you dialled, minus the first digit (the leading 0)
Cheers
Mat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mike Williams
Sent: 22 September 2006 13:49
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] freepbx dial plan,add and
remove at the same time
On Friday 22 September 2006 13:36, Mat Stace wrote:
It's a while since I did this one myself, but I was doing the exact
same thing when using voipbuster (or whichever of it's sisters
services I was using at the time).
I'm thinking that in the dial command you want
+44{EXTEN:1}
Thanks, but could you explain how that works?
The {EXTEN:1} suggests the first digit is removed, or perhaps
more precisely
that's a place holder for the number dialed starting one digit in?
--
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RE: [asterisk-users] Change the from@ using the voicemail.conf
Hi Dean, In the voicemail.conf, in the [general] section near the top, I've got ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] My e-mails now come from [EMAIL PROTECTED], making to easy to set up a filter in my e-mail client to move voicemail messages into a specific folder HTH Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean @ INKnBITs Sent: 28 July 2006 14:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change the from@ using the voicemail.conf Hi, I'm trying to setup the voicemail.conf to email messages, but my mail server fails because the from user is [EMAIL PROTECTED] Does anybody know away to change the user part from root? I'm using exim4 to send the emails. Thanks, Dean. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/402 - Release Date: 27/07/2006 ___ --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] Change the from@ using the voicemail.conf
Bad form replying to myself, I know, but it looks like my outlook stripped the carriage return. Should be ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] With the comment on the line above the serveremail line Cheers Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mat Stace Sent: 28 July 2006 14:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Change the from@ using the voicemail.conf Hi Dean, In the voicemail.conf, in the [general] section near the top, I've got ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] My e-mails now come from [EMAIL PROTECTED], making to easy to set up a filter in my e-mail client to move voicemail messages into a specific folder HTH Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean @ INKnBITs Sent: 28 July 2006 14:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change the from@ using the voicemail.conf Hi, I'm trying to setup the voicemail.conf to email messages, but my mail server fails because the from user is [EMAIL PROTECTED] Does anybody know away to change the user part from root? I'm using exim4 to send the emails. Thanks, Dean. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/402 - Release Date: 27/07/2006 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/402 - Release Date: 27/07/2006 ___ --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] Two phone numbers, one SIP provider
Title: Message That won't help either. Context is always 'default', but what I want is a different context on any number. Maybe oej'speermatch branch solves the problem. But I cannot compile it, There are lots of ' merge right' tags in chan_sip.c. How about a slight modification of my solution: extensions.conf [incoming_sip_provider] exten = ,1,Goto(_context,s,1) exten =,1,Goto(_context,s,1) That should seperate out the two incoming calls, and give each one access to what they need access to? Cheers Mat ___ --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] Two phone numbers, one SIP provider
Title: Message I'm not exactly sure on the /how/ * mathes items from the sip.conf (I suspect it goes to the latter for whichever provider), but the way configured my extenions.conf to handle multiple incoming accounts from sipgate is like this (obviously much simplified for ease of explanation): [incoming_sipgate] exten = ,1,Answer exten = ,2,Dial(SIP/ciscophone,12) exten = ,1,Answer exten = ,2,Dial(SIP/pcsoftphone,12) Also, in the sip.conf, each peer has context=incoming_sipgate in it. HTH, Mat -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin StockerSent: 20 July 2006 16:05To: Asterisk Users Mailing ListSubject: [asterisk-users] Two phone numbers, one SIP provider HiI have two phone numbers from my SIP provider sippro.com, say and . I use two sip.conf entries to register this phone numbers:register = :[EMAIL PROTECTED]/register = :[EMAIL PROTECTED]/[]type=friendusername=secret=passinsecure=veryhost= sip.sippro.comcontext=incoming-[]type=friendusername=secret=passinsecure=veryhost=sip.sippro.comcontext=incoming-Now, from my dialplan I can use them to do outgoing calls, like Dial(SIP/[EMAIL PROTECTED]). That works pretty fine. The problem are incoming calls. According to [1] asterisk should lookup a match in sip.conf when somebody (outside sippro.com) calls or . For example, a call to should look for a extension in context 'incoming-'. A call for should go to context incoming-. But in the above scenario, asterisk always gets a match on ''. As a result, context 'incoming-' is always used. How does asterisk search for a match in sip,conf for incoming calls and how can I get it to use the context specified in the account settings?1. http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf --No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.10.2/393 - Release Date: 19/07/2006 ___ --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] One extension can transfer internal calls, can't transfer incoming external calls
Greetings list,
I've been bashing my head against a brick wall for a couple of weeks now to
try and get this sorted, have been scouring google/the asterisk-users list
archives to no avail.
The problem I am having is that one extension (an off-site iaxy) cannot
transfer incoming calls from our IAX provider, but can transfer internal
calls. We can transfer incoming external calls on site using our cisco
7960's, just not remotely with the iaxy.
I thought I had cracked it this morning when I found out about the
notransfer=yes option for the IAX2 peers, to prevent the call from being
reinvited by the iaxy, and not going through the asterisk server, but
although the call is staying through the asterisk box, it's still not
possible to transfer an incoming call from the iaxy to one of the cisco
phones.
Basically, this is what works and doesn't
Iax provider - asterisk server - iaxy = iaxy cannot transfer the call
Iax provider - asterisk server - cisco 7960 = 7960 can transfer the call
Cisco 7960 - asterisk server - iaxy = whoever makes the call, both users
can transfer.
The blind transfer is being done by using the # key, we're using asterisk
1.0.9 (downgraded after trying a higher version (think it was .23ish) that
dropped external calls after 3 minutes).
The (I think) relevant bits from extensions.conf, sip.conf, and iax.conf
(suitably munged for public distribution ;) ) are below. I've tried adding
Tt to the end of every dial string I can, and even tried it on the end of
the GotoIfTime line of the [iaxprovider-in] section of extensions.conf,
which I doubt will make any difference if it's there or not.
The DTMF detection is working fine for both the iaxy and the cisco phone,
both users can use the voicemail application fine, and dtmf tones get passed
through to call centres etc.
Has anybody come across anything like this in the past, where certain
extensions can only sometimes forward calls? I have noticed that in the iaxy
provisioning it's possible to disable call transfer, does this mean that the
iaxy has it's own key combination for call transfer?
Cheers in advance,
Mat
extensions.conf
[default]
exten = 23,1,dial(SIP/sipuser,12,Tt)
exten = 23,2,Voicemail(su23)
exten = sipuser,1,goto(23,1)
exten = 34,1,dial(IAX2/[EMAIL PROTECTED],20,Tt)
exten = 34,2,Voicemail(su34)
[iaxprovider-in]
exten = incomingiaxprovidernumber,1,Answer
exten = incomingiaxprovidernumber,2,Wait,1
exten = incomingiaxprovidernumber,3,NoOp(--- ${CALLERID} calling on
INCOMING IAX PROVIDER (${EXTEN}) ---)
exten = incomingiaxprovidernumber,4,Wait,1
exten =
incomingiaxprovidernumber,5,GotoIfTime(9:00-17:00|mon-fri|*|*?office-hours,s
,1,Tt)
exten = incomingiaxprovidernumber,6,Background(officeclosed)
exten = incomingiaxprovidernumber,7,Voicemail(s01)
exten = incomingiaxprovidernumber,8,Hangup
[office-hours]
exten = s,1,NoOp()
exten = s,2,NoOp()
exten = s,3,NoOp()
exten = s,4,Dial(SIP/sipuserIAX2/[EMAIL PROTECTED],18,Tt)
exten = s,5,Answer
exten = s,6,Wait,1
exten = s,7,Voicemail(su01)
exten = s,8,Hangup
iax.conf:
[iaxy1]
type=friend
accountcode=iaxy
host=dynamic
notransfer=yes
username=iaxy1
secret=secret
context=default
disallow=all
allow=ulaw
callerid=IAXy 1 34
trunk=no
sip.conf
[sipuser]
type=friend
host=dynamic
dtmfmode=inband
username=ciscophone
secret=ciscophone
qualify=200
reinvite=no
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
nat=yes
mailbox=23,01
callgroup=1
pickupgroup=1
callerid=Mat 23
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RE: [Asterisk-Users] sometimes dtmf passed, sometimes not (cisco 7960 SIP)
Just to answer my own query, I needed to set the devices to dtmfmode=inband in my sip.conf, and on the 7960 set Sip configuration - Out of Band DTMF - none The benefits of a good nights sleep :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mat Stace, Colewood Sent: 13 September 2005 22:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] sometimes dtmf passed,sometimes not (cisco 7960 SIP) [major snippage] I hope the above makes some sense, it's basically is it an asterisk or 7960 setting to make it pass dtmf whilst on a call Cheers (and apologies for semi-coherance) Mat -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.24/101 - Release Date: 13/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sometimes dtmf passed, sometimes not (cisco 7960 SIP)
Hi list, I'm hoping that I'm being stupid, and someone can tell me
what's going on, but for the life of me I can't figure it out. (it's
been a long day, and I'm now in the last 3 weeks of organising my
wedding, so I hope this makes sense ;) )
When at my desk, accessing (for example) my voicemail, the dtmf tones
are passed perfectly, I can enter password, change folders, etc etc.
I'm trying to get the phone set up so I can use it from a remote
location, and due to both ends being behind nat, I figured that the best
way+ would be to have a sipgate account that the 7960 registers to, a
sipgate account that the * box registers to, and when the * box receives
a call, it can pass it to [EMAIL PROTECTED] (7960 at home), rather than a
local extension.
+OK, easiest way. I know the best way would be to have a 2nd * server
and hook them up via IAX ;-)
This works great, so I figured I'd then set up the sipgate account on
the * box with DISA, so that the remote phone user can dial the *
sipgate account, enter the extension for disa while the background sound
is played, then dial an internal extension/external line.
from my extensions.conf
exten = SIPGATEID,1,Answer
exten = SIPGATEID,2,NoOp(--- ${CALLERID} calling on Sipgate
(${EXTEN}) ---)
exten = SIPGATEID,3,Wait,1
exten = SIPGATEID,4,Background(bgsound)
exten = SIPGATEID,5Voicemail(s23)
exten = SIPGATEID,6,Hangup
exten = 000,1,DISA(no-password|default)
The problem is that when the bgsound is playing, I dial 000 on the 7960,
and the bgsound keeps playing. This also happens when the 7960 is in my
office, hooked up to * as a local extension. I have tried all three
out-of-band DTFM settings on the 7960, with no change. In my sip.conf,
the sipgate account is set up with dtmfmode=info which I thought might
have been causing a problem until I tried ringing the sipgate DID from
my mobile, which just worked. Typical.
I'm starting to think that there must be another hidden setting
somewhere on the 7960 to allow dtmf to be passed whilst on a call, as
some of the things from features.conf don't work either (* based
transfer and parking), but I don't think this is the case due to being
able to use the voicemail properly.
I hope the above makes some sense, it's basically is it an asterisk or
7960 setting to make it pass dtmf whilst on a call
Cheers (and apologies for semi-coherance)
Mat
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Re: [Asterisk-Users] VoipBuster with astersisk?
I'm running voipbuster via IAX, though you'll have to change the
dialstring, as I only use it for UK landline numbers :)
In my iax.conf
[voipbuster]
type=peer
host= 213.61.187.150
secret=YOURPASSWORD
notransfer=yes
context=default
In My extensions.conf:
exten = _770[12].,1,SetCallerID(CID Name CIDNUMBER)
exten = _770[12].,2,Dial,IAX2/[EMAIL PROTECTED]/0044${EXTEN:3}
I don't actually know if the first line works (never actually tested it
that far :-| ) and you'll probably want the 2nd line to be something
like this if you want to use it for all calls worldwide
exten = _9.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:1}
This should give you the 9 for a line mdoe of operation, and require
you to dial full international numbers.
Cheers
Mat
(standard disclaimer - while the above works for me, it's for a
particular purpose. YMMV, don't sue me if it breaks, etc etc etc) ;-D
[EMAIL PROTECTED] wrote:
Hi, all
Here is a something I found on the web:
http://www.voipbuster.com
And it works OK too. Now, I want to use it via asterisk, so I ccan use my
normal phones instead of PC application.
Did anyone try to connect astersisk and VoipBuster?
Thanks,
Rudolf
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Re: [Asterisk-Users] VoipBuster with astersisk?
No problems, and the first thing I did was put the money on my account.
I did use it a few times with the client first though - setting it up
with * was actually a rush job at work when our current outgoing
provider was down, I used my personal voipbuster account, hence locking
it down to numbers that wouldn't generate a charge. Maybe they haven't
yet noticed that Im using * with their service ;-)
Cheers
Mat
[EMAIL PROTECTED] wrote:
Thanks,
I'll try it. From what I read on the Internet, people start to have problems when they pput money on their account.
They say it works ok when account is empty, but when 1euro is deposited, client still works, but asterisk does not. Did you have any problems?
Rudolf
Mat Stace, Colewood [EMAIL PROTECTED] wrote:
I'm running voipbuster via IAX, though you'll have to change the
dialstring, as I only use it for UK landline numbers :)
In my iax.conf
[voipbuster]
type=peer
host= 213.61.187.150
secret=YOURPASSWORD
notransfer=yes
context=default
In My extensions.conf:
exten = _770[12].,1,SetCallerID("CID Name" CIDNUMBER)
exten = _770[12].,2,Dial,IAX2/[EMAIL PROTECTED]/0044${EXTEN:3}
I don't actually know if the first line works (never actually tested it
that far :-| ) and you'll probably want the 2nd line to be something
like this if you want to use it for all calls worldwide
exten = _9.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:1}
This should give you the "9 for a line" mdoe of operation, and require
you to dial full international numbers.
Cheers
Mat
(standard disclaimer - while the above works for me, it's for a
particular purpose. YMMV, don't sue me if it breaks, etc etc etc) ;-D
[EMAIL PROTECTED] wrote:
Hi, all
Here is a something I found on the web:
http://www.voipbuster.com
And it works OK too. Now, I want to use it via asterisk, so I ccan use my normal phones
instead of PC application.
Did anyone try to connect astersisk and VoipBuster?
Thanks,
Rudolf
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RE: [Asterisk-Users] Google introduces text/audio chat client andservice
Lifted from the developer page of the google talk site (http://www.google.com/talk/developer.html) 5. What protocols are used for voice calls? Google Talk supports a custom XMPP-based signaling protocol and peer-to-peer communication mechanism. We will fully document this protocol. In the near future, we plan to support SIP signaling. Looks like it's not quite there yet. I'm willing to bet there'll be a massive sign up (mostly from this list) when they do start supporting SIP - could Google be the driving force to a massive take up in * - when all the regular joes who sign up to google want to extend their google voip with answerphones etc? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Glomph Black Sent: 24 August 2005 06:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Google introduces text/audio chat client andservice http://www.google.com/talk/about.html They claim it's all based on 'open standards' (as opposed to Skype, the Roach Motel of telephony). The text IM portion uses Jabber, and they encourage the use of any Jabber client. Works great in Gaim and iChat, in my brief successful trials. The big question is: will there be some plausible hook into this telephony system using Asterisk?How open is Open, this time? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.14/79 - Release Date: 22/08/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.15/80 - Release Date: 23/08/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
As of 22:45 GMT it's working for me Jerry Glomph Black wrote: This service has been working well lately, but as of this morning is promptly blowing off IAX connections with the dreaded 'No Authority Found' error. Any concrete info greatly appreciated! Dr G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/08/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Forbidden - wrong password on authentication forNOTIFY
sip show registry ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald_Wiplinger Sent: 09 August 2005 09:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Forbidden - wrong password on authentication forNOTIFY How can I find out which phone and what is missing? WARNING[10532]: chan_sip.c:8669 handle_response: Forbidden - wrong password on authentication for NOTIFY bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.4/66 - Release Date: 09/08/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.4/66 - Release Date: 09/08/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple MWI on a single phone?
I currently have 4 lines on my Cisco 7960G, between these 4 lines there are 3 mail boxes (one work, one personal, and two testing lines sharing a mailbox). It's not so much multiple MWI as multiple lines with their own MWI, but it does the job Cheers M -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Hirsch Sent: 08 August 2005 14:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Multiple MWI on a single phone? Hey all...I'm trying to find a phone that will support multiple MWI so that I can have a shared central phone with say 4 users who can see visually that hey have messages waiting. Is there any phone that will do this possibly by re-assigning a soft-button? Can the Polycoms do this since those seem to be the phone of choice these days? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.2/65 - Release Date: 07/08/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.2/65 - Release Date: 07/08/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can you caculate with me?
-Original Message- Adam Dobrin Bob Goddard wrote: On Thursday 28 Jul 2005 13:07, Ronald Wiplinger wrote: before I accuse somebody to overbill I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers:0.7269 or 0.2929 ??? Assuming it is per minute; 930 * 0.0189 / 60 = 0.29295 B I get .31$. Where did you all go to school? Is there a connection charge? I went to school in a place they tought maths (or math if you're from that site of the pond ;-P). I make it 0.29295. There could be other charges alongside conenction charges too - there could be a minimum duration charge for calls too. M -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.6/59 - Release Date: 27/07/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playtones not passing sound to incoming SIP connection
Hi everyone,
I'm in the very early stages of rolling out an asterisk box at work, and one
of the things I'm setting up is a trap for telemarketers ;)
What I want to do is have a sipgate number in the UK here which rings for 10
seconds without calling a hard or softphone, then goes to a voicemailbox.
The problem I'm having is that Playtones doesn't seem to be sending any
sound to the incoming SIP connection. I have added the following to my
[incoming_sipgate] context (which has two other sipgate numbers in there
which both work for incoming and outgoing calls), and on the console I can
see all the lines being executed.
exten = SIPGATEID,1,Wait,1
exten = SIPGATEID,2,NoOp(--- ${CALLERID} calling on Telemarket Divert
Sipgate (${EXTEN}) ---)
exten = SIPGATEID,3,Answer
exten = SIPGATEID,4,Playtones,ring
exten = SIPGATEID,5,Wait,10
exten = SIPGATEID,6,StopPlaytones
exten = SIPGATEID,7,Voicemail(666)
exten = SIPGATEID,8,Hangup
When dialing in via the PSTN number, or from a remote SIP softphone however,
the ten seconds which whould be the Playtones is silence. When the voicemail
kicks in, I can hear the announcements, and leave a message, so I don't
think it's a ports problem. For the playtones line, I have also tried exten
= SIPGATEID,5,Playtones(ring) but doesn't seem to make any difference.
Internally, I set up an extension (in my [default] context - I should have
an [internal] one, I know, ;-P ) with the same commands:
exten = 6613,1,Wait,1
exten = 6613,3,Answer
exten = 6613,4,Playtones,ring
exten = 6613,5,Wait,10
exten = 6613,6,StopPlaytones
exten = 6613,7,Voicemail(666)
exten = 6613,8,Hangup
And when dialing 6613, I get ten seconds of ringtone, then to the
answerphone as expected. There is a difference in the two, in that the
sipgate one has the NoOp line in, but I initially tried the 6613 extension
with that line in (I removed it for ease of differentiation in the console).
Anyone got any ideas on this one? The fact that I can hear the voicemail
announement and leave a message has really thrown me. Maybe I'll just have
to create a .gsm recording of the ringer and use a Playback instead.
Cheers in advance,
Mat
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