Re: [asterisk-users] Mailing List Future
On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: > To that end, we’ve decided to discontinue the mailing lists effective > February 1st, 2024. That's a very sad news! :-( -- _ -- 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] audio from soft phone actual phone from cloud
On Wed, 2023-07-19 at 12:42 -0400, Jerry Geis wrote: > Why might I not be getting audio ? Make sure the RTP port range is correctly configured and open on your server's firewall. The port range is defined in /etc/asterisk/rtp.conf The same range of UDP ports must be correctly forwarded on your firewall from the outside to Asterisk. For example, in rtp.conf: [general] ; ; RTP start and RTP end configure start and end addresses ; rtpstart=10002 rtpend=10199 -- _ -- 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] Intro and question
Even I was confused, and the directions in that book seem like a complication of a simple affair, at least for my modest needs. Finally, I installed Asterisk with apt and created extensions.conf and pjsip.conf files. -- _ -- 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] Dial() after the h extension has been invoked?
On Fri, 2021-11-12 at 16:56 +, Antony Stone wrote: > I use Dial() commands with custom SIP headers to pass information > (eg: about > the current state of a call) between the front-end and back-end > machines, and > this works very well. > > I need to perform a Dial() > command after an inbound channel has hung up. I do not expect the > Dial() to > bridge to anything (the context being dialled simply does some > database > manipulation and then hangs up without even bothering to answer). > > > Any suggestions welcome :) Maybe you can use the "g" option in the first Dial(...) and proceed in the dial plan with the second Dial(...) g - Proceed with dialplan execution at the next priority in the current extension if the destination channel hangs up. Example: exten => 1234,1,Dial(SIP/deskphone,120,g) same => n,Dial(SIP/cordlessphone) same => n,Hangup() Extension 1234 dials a deskphone. If "deskphone" answer... bla bla bla... and after "deskphone" hangs up, the "cordlessphone" is dialed. -- _ -- 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] Notifying missed calls
On Sat, 2021-11-06 at 14:46 +0100, Luca Bertoncello wrote: > Really, I can't understand what you mean... I'm feeling really > dumb... No need to feel dumb. I'm not an expert and when I look to my extensions.conf... well... countless pulling my hairs out, head banging on the keyboard,,, :-) The "h" extension is executed whenever a call is hang up in that contexts. In your configuration it executes first the "s" extension (where you GoTo h,1) and once that is executed, the "h" extension is executed again. Take a look to the example I posted. It's very basic, but it does the job. -- _ -- 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] Notifying missed calls
On Fri, 2021-11-05 at 10:50 +0100, Luca Bertoncello wrote: > 1) The E-Mails will be sent "double" It sends the first mail by executing "noanswer,2" and a second mail because because of "main-incoming,h,2" > 2) The E-Mails will be sent for outgoing unanswered calls, too. Use the "h" extension only in the context for incoming calls > -- _ -- 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] Notifying missed calls
Here my configuration:
[incoming]
; Incoming from Swisscom
exten => +4191xxx,1,NoOp(Call from ${CALLERID(num)})
same => n,Dial(SIP/deskphone,120)
same => n,Hangup()
exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done)
exten => h,n,System(echo "Missed Call from ${CALLERID(num)}" | mail -s
"Missed Call from ${CALLERID(num)}" my-em...@address.here)
exten => h,n(done),NoOp()
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Re: [asterisk-users] Detect if people is talking
On Wed, 2020-12-30 at 12:09 -0300, Valter Nogueira wrote: > Is there any way to detect if an agent is speaking? https://wiki.asterisk.org/wiki/display/AST/Application_WaitForSilence https://wiki.asterisk.org/wiki/display/AST/Application_WaitForNoise -- _ -- 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] which linux for asterisk?
On Wed, 2020-12-09 at 11:03 +0400, Dmitry Melekhov wrote: > what is best choice ? Oracle? Ubuntu? I'm running Asterisk since several years on Ubuntu without any issues. Debian should be fine too. -- _ -- 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] Anyone that know of DECT "client" for asterisk?
On Sat, 2020-10-03 at 22:25 +0200, Sebastian Nielsen wrote: > many providers in sweden have started disabling SIP account details > and now require usage of their own ”router’s”. That's very irritating and make me angry. Few of my client had the same problem. The solution: write a letter asking the SIP credentials explaining you want configure your own equipment and tell them you switch to another provider in case of refusal. Good luck! I don't know if there is an appropriate hardware to build a DECT bridge and I doubt that fiddling with anything like that will not be a reliable solution. -- _ -- 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] Mail2Fax
On Wed, 2020-06-17 at 18:10 +0200, basti wrote: > txfax seem to be a port of spandsp. it is also old. > Is there a newer way to send fax via asterisk. I don't know if it's newer, but I use "sendfax" -- _ -- 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] email notification on missed call
On Sat, 2019-11-02 at 11:42 +0100, Antony Stone wrote:
> Doesn't that send an email for every call once it ends, not just
> unanswered ones?
Whoops! You are right! :-)
exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done)
exten => h,n,System(echo "Missed Call Open on Asterisk from
${CALLERID(num)}" | mail -s "Missed VIP Call on Asterisk from
${CALLERID(num)}" -a "From: Astersik PBX " myemailaddr
e...@example.com)
exten => h,n(done),NoOp()
exten => h,n,HangUp()
> > exten => h,n,HangUp()
> That looks most strange to me - calling Hangup() in the hangup
> extension...
:-D
Probably it is not necessary. But isn't a good practice to end any
extension with a "HangUp"?
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Re: [asterisk-users] email notification on missed call
On Wed, 2019-10-30 at 05:10 +0100, Fourhundred Thecat wrote:
> what is the best way to implement email notification on missed call ?
> Is there perhaps a better way to this than described above ?
This is my way:
exten => h,1,System(echo "Missed Call Open on Asterisk from
${CALLERID(num)}" | mail -s "Missed VIP Call on Asterisk from
${CALLERID(num)}" -a "From: Astersik PBX " myemailaddr
e...@example.com)
exten => h,n,HangUp()
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Re: [asterisk-users] Asterisk 13.28.0 Now Available
Thank you, dear Asterisk Development Team, for this great software! > The Asterisk Development Team would like to announce the release of > Asterisk 13.28.0. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS and SIM card
On Fri, 2019-04-26 at 14:39 +, bilal ghayyad wrote: > Any small example how to send gsm calls through chan_dognle and how > to send sms through chan_dongle? To send SMS, there is a CLI command. You can use the commands in your extensions.conf accordingly your needs. http://wiki.e1550.mobi/doku.php?id=usage dongle sms Send SMS to with the using -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS and SIM card
On Fri, 2019-04-26 at 14:39 +, bilal ghayyad wrote:
> Any small example how to send gsm calls through chan_dognle and how
> to send sms through chan_dongle?
In dongle.conf:
[gsmgateway]
context=gsm
imei=123456789012345
imsi=098765432112345
In extensions.conf:
[gsm]
; Incoming calls from GSM/3G
exten => +41776665544,1,Dial(Local/mydeskphone@voipphone)
same => n,Hangup()
; Phone call though GSM/3G
exten => _0.,n,Dial(Dongle/gsmgateway/${EXTEN},120)
same => n,Hangup()
; Incoming SMS to mail address and to sms.txt file
exten => sms,1,Noop(Incoming SMS from ${CALLERID(num)}
${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} -
${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}' >>
/var/log/asterisk/sms.txt)
exten => sms,n,System(echo "${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} -
${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}" |
mail -s "SMS from ${CALLERID(num)}" myemailaddr...@mymailbox.com)
exten => sms,n,Hangup()
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Re: [asterisk-users] Sending SMS and SIM card
You can use a cheap 3G-USB-dongle and chan_dongle. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asking
On Tue, 2018-09-18 at 20:28 +0200, modou lo wrote: > Hello, please can i have a code which help me to tax user every voip > services in asterisk means when user starts to call someone Check Asterisk2billing http://www.asterisk2billing.org/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Decoding SIP register hack
On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote:
> 3. How do I set up the server to block these ?
>
> 4. Can I stop the retransmitting of the 401 Unauthorized packets ?
I'm happy with Fail2Ban protecting my Asterisk 13. Here is my
configuration:
in /etc/asterisk/logger.conf:
messages => security,notice,warning,error
in /etc/asterisk/sip.conf:
allowguest=yes
context=unauthenticated
in /etc/asterisk/extensions.conf:
[unauthenticated]
;; Incomming calls from unauthenticated caller -> Fail2Ban
exten => _X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _X.,3,HangUp()
exten => _+X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _+X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _+X.,3,HangUp()
in /etc/fail2ban/jail.conf:
[asterisk]
filter = asterisk
action = iptables-allports[name=ASTERISK]
logpath = /var/log/asterisk/messages
maxretry = 1
findtime = 86400
bantime = 518400
enabled = true
in /etc/fail2ban/filter.d
# Fail2Ban configuration file
#
#
# $Revision: 250 $
#
[INCLUDES]
# Read common prefixes. If any customizations available -- read them
from
# common.local
#before = common.conf
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the
logfile. The
# host must be matched by a group named "host". The tag
"" can
# be used for standard IP/hostname matching and is only an
alias for
# (?:::f{4,6}:)?(?P\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for
':.*' - Wrong password
NOTICE.* .*: Call from '.*' \((:[0-9]{1,5})?\) to
extension '.*' rejected because extension not found in context
'unauthenticated'
NOTICE.* chan_sip.c: Call from '.*' \((:[0-
9]{1,5})?\) to extension '.*' rejected because extension not found in
context 'unauthenticated'
NOTICE.* .*: Registration from '.*' failed for
':.*' - Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for
':.*' - No matching peer found
NOTICE.* .*: Registration from '.*' failed for
':.*' - Not a local domain
NOTICE.* .*: Registration from '.*' failed for
':.*' - Peer is not supposed to register
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device does not match ACL
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device not configured to use this transport type
NOTICE.* .*: No registration for peer '.*' \(from
\)
NOTICE.* .*: Host failed MD5 authentication for
'.*' \(.*\)
NOTICE.* .*: Host denied access to register peer
'.*'
NOTICE.* .*: Host did not provide proper
plaintext password for '.*'
NOTICE.* .*: Registration of '.*' rejected: '.*' from:
''
NOTICE.* .*: Peer '.*' is not dynamic (from )
NOTICE.* .*: Host denied access to register peer
'.*'
SECURITY.* .*:
SecurityEvent="InvalidAccountID".*,Severity="Error",Service="SIP".*,Rem
oteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="FailedACL".*,Severity="Error",Service="SIP".*,RemoteAddr
ess="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="InvalidPassword".*,Severity="Error",Service="SIP".*,Remo
teAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="ChallengeResponseFailed".*,Severity="Error",Service="SIP
".*,RemoteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-
noservice' \(language '.*'\)
SECURITY.* .*:
SecurityEvent="ChallengeSent".*,Severity="Informational",Service="SIP".
*,AccountID="sip:.*@93.94.247.123".*,RemoteAddress="IPV[46]/(UDP|TCP|TL
S)//[0-9]+
WARNING.* .*: fail2ban=''
# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Values: TEXT
#
ignoreregex =
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Re: [asterisk-users] invite to conference by a call file
Maybe something like a local web page where your secretary can enter the list of phone numbers to call and a script that generates a call file and moves it to the Asterisk spool folder. But that's not an Asterisk issue. It's more a programmer's issue. :-) On Thu, 2018-03-22 at 16:06 +0200, Atux Atux wrote: > that's the problem. it is never the same people > I need the office secretary to edit a file (call file) > and place it in a particular folder in their windows PCs. this folder > is the outgoing folder of LINUX shared through samba in LAN. i need > to make it as easy as possible, please. -- _ -- 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] invite to conference by a call file
If the participants are always the same people, there is no need to change the dialplan. Just tells the office secretary "Please, place a conference call.". with the "Page" application, she picks up the phone, dials a predefined number and all the participants are called at once. Easy peasy. :-) On Thu, 2018-03-22 at 14:21 +0200, Atux Atux wrote: > All the aforementioned techniques need change everytime on the > dialplan. I need the office secretary to edit a file (call file) and > place it in a particular folder in their windows PCs. this folder is > the outgoing folder of LINUX shared through samba in LAN. i need to > make it as easy as possible, please. > > On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist@linuxista. > com> wrote: > > Here I'm using the "Page" application to make a conference call "on > > the fly". > > > > > > > > [office] > > > > exten => ,1,Dial(SIP/desk2,150) > > same => n,Hangup() > > > > exten => ,1,Dial(SIP/desk3,150) > > same => n,Hangup() > > > > exten => ,1,Dial(SIP/desk4,150) > > same => n,Hangup() > > > > exten => ,1,Dial(SIP/desk5,150) > > same => n,Hangup() > > > > exten => ,1,Dial(SIP/desk6,150) > > same => n,Hangup() > > > > ; Conference call > > exten => ,1,Answer > > exten => ,n,Page(Local/@office/@office/ > > @office/@office/@office,d) > > same => n,Hangup() > > > > -- > > ___ > > __ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > > -- > > > > Check out the new Asterisk community forum at: https://community.as > > terisk.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 > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.aste > risk.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 -- _ -- 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] invite to conference by a call file
Here I'm using the "Page" application to make a conference call "on the fly". [office] exten => ,1,Dial(SIP/desk2,150) same => n,Hangup() exten => ,1,Dial(SIP/desk3,150) same => n,Hangup() exten => ,1,Dial(SIP/desk4,150) same => n,Hangup() exten => ,1,Dial(SIP/desk5,150) same => n,Hangup() exten => ,1,Dial(SIP/desk6,150) same => n,Hangup() ; Conference call exten => ,1,Answer exten => ,n,Page(Local/@office/@office/@off ice/@office/@office,d) same => n,Hangup()-- _ -- 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] Blacklist failed attempts
On Thu, 2018-03-01 at 15:02 +0200, Atux Atux wrote:
> I have tried to implement it through fail2ban, but it doe snot seem
> to work for my asterisk implementation.
I'm happy with Fail2Ban protecting my Asterisk 13. Here is my
configuration:
in /etc/asterisk/logger.conf:
messages => security,notice,warning,error
in /etc/asterisk/sip.conf:
allowguest=yes
context=unauthenticated
in /etc/asterisk/extensions.conf:
[unauthenticated]
;; Incomming calls from unauthenticated caller -> Fail2Ban
exten => _X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _X.,3,HangUp()
exten => _+X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _+X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _+X.,3,HangUp()
in /etc/fail2ban/jail.conf:
[asterisk]
filter = asterisk
action = iptables-allports[name=ASTERISK]
logpath = /var/log/asterisk/messages
maxretry = 1
findtime = 86400
bantime = 518400
enabled = true
in /etc/fail2ban/filter.d
# Fail2Ban configuration file
#
#
# $Revision: 250 $
#
[INCLUDES]
# Read common prefixes. If any customizations available -- read them
from
# common.local
#before = common.conf
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the
logfile. The
# host must be matched by a group named "host". The tag
"" can
# be used for standard IP/hostname matching and is only an
alias for
# (?:::f{4,6}:)?(?P\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for
':.*' - Wrong password
NOTICE.* .*: Call from '.*' \((:[0-9]{1,5})?\) to
extension '.*' rejected because extension not found in context
'unauthenticated'
NOTICE.* chan_sip.c: Call from '.*' \((:[0-
9]{1,5})?\) to extension '.*' rejected because extension not found in
context 'unauthenticated'
NOTICE.* .*: Registration from '.*' failed for
':.*' - Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for
':.*' - No matching peer found
NOTICE.* .*: Registration from '.*' failed for
':.*' - Not a local domain
NOTICE.* .*: Registration from '.*' failed for
':.*' - Peer is not supposed to register
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device does not match ACL
NOTICE.* .*: Registration from '.*' failed for
':.*' - Device not configured to use this transport type
NOTICE.* .*: No registration for peer '.*' \(from
\)
NOTICE.* .*: Host failed MD5 authentication for
'.*' \(.*\)
NOTICE.* .*: Host denied access to register peer
'.*'
NOTICE.* .*: Host did not provide proper
plaintext password for '.*'
NOTICE.* .*: Registration of '.*' rejected: '.*' from:
''
NOTICE.* .*: Peer '.*' is not dynamic (from )
NOTICE.* .*: Host denied access to register peer
'.*'
SECURITY.* .*:
SecurityEvent="InvalidAccountID".*,Severity="Error",Service="SIP".*,Rem
oteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="FailedACL".*,Severity="Error",Service="SIP".*,RemoteAddr
ess="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="InvalidPassword".*,Severity="Error",Service="SIP".*,Remo
teAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
SECURITY.* .*:
SecurityEvent="ChallengeResponseFailed".*,Severity="Error",Service="SIP
".*,RemoteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-
noservice' \(language '.*'\)
SECURITY.* .*:
SecurityEvent="ChallengeSent".*,Severity="Informational",Service="SIP".
*,AccountID="sip:.*@93.94.247.123".*,RemoteAddress="IPV[46]/(UDP|TCP|TL
S)//[0-9]+
WARNING.* .*: fail2ban=''
# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Values: TEXT
#
ignoreregex =
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Re: [asterisk-users] email when certain numbers are called
On Mon, 2018-01-15 at 14:26 +0200, Atux Atux wrote:
> [DefaultPlan]
exten => _XX,1,System(echo "Dialed number ${EXTEN} on Asterisk
from ${CALLERID(num)}" | mail -s "Dialed number ${EXTEN} on Asterisk
from ${CALLERID(num)}" -a "From: Asterisk PBX " yo
urem...@address.com)
exten => _XX,2,Dial(SIP/VoipGate/${EXTEN},120,Tt)
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Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??
> fail2ban is most useful for blocking registration attempts. I
> handle
> non-registration call attempts by allowing guests, point them to a
> jail
> context, which runs Log(WARNING,fail2ban='${CHANNEL(peerip)}') I
> set a
> fail2ban rule to match that line logged from Asterisk.
Thanks for the suggestion. Works great! :-)
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Re: [asterisk-users] SIP trunks going to the wrong context
I don't know if it applies to your problem, but I also had some troubles with multiple account on same SIP provider. Here what works for me: In sip.conf: register => 11:qwe...@sip.provider.zz/11 ; Trunk1 register => 22:asd...@sip.provider.zz/22 ; Trunk2 register => 22:yxc...@sip.provider.zz/22 ; Trunk3 [trunk1] type=friend host=sip.provider.zz defaultuser=11 secret=qwertz canreinvite=no insecure=invite nat=force_rport,comedia qualify=yes context=trunkincoming description=Trunk 1 [trunk2] type=friend host=sip.provider.zz defaultuser=22 secret=asdfgh canreinvite=no insecure=invite nat=force_rport,comedia qualify=yes context=trunkincoming description=Trunk 2 [trunk3] type=friend host=sip.provider.zz defaultuser=33 secret=yxcvbn canreinvite=no insecure=invite nat=force_rport,comedia qualify=yes context=trunkincoming description=Trunk 3 In extensions.conf: [trunkincoming] exten => 11,1,GoTo(firstline,11,1) exten => 22,1,GoTo(secondline,22,1) exten => 33,1,GoTo(thirdline,33,1) [firstline] exten => 11,1,Dial(SIP/officephone,120,m) [secondline] exten => 22,1,Dial(SIP/livingroomphone,120,m) [thirdline] exten => 33,1,Dial(SIP/bedroomphone,120,m) -- _ -- 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] Looking for the carrier that owns a particular DID
On Thu, 2017-11-02 at 11:33 -0400, Tech Support wrote: > How do I find out which carrier owns the DID in question? Try here: https://www.twilio.com/lookup -- _ -- 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] PJSIP, NAT and STUN/ICE
On Tue, 2017-10-10 at 11:32 +0200, Frank Vanoni wrote: > On Mon, 2017-10-09 at 23:56 +0200, O. Hartmann wrote: > > > > > local_net= 192.168.254.1/24 > > It should be: > > localnet = 192.168.254.0/255.255.255.0 Whoops... local_net=192.168.254.0/24 -- _ -- 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] PJSIP, NAT and STUN/ICE
On Mon, 2017-10-09 at 23:56 +0200, O. Hartmann wrote: > local_net= 192.168.254.1/24 It should be: localnet = 192.168.254.0/255.255.255.0 -- _ -- 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] Dial an extension to modify dialplan
On Wed, 2017-05-10 at 12:56 +0200, Frank Vanoni wrote: > exten => 2001,1,Dial(SIP/Dial(SIP/deviceA/deviceB/deviceC) > > exten => 2002,1,Dial(SIP/Dial(SIP/deviceA/deviceB) Whoops... sorry for the typo (in the hurry of copy & paste)! exten => 2001,1,Dial(SIP/deviceA/deviceB/deviceC) exten => 2002,1,Dial(SIP/deviceA/deviceB) -- _ -- 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] Dial an extension to modify dialplan
Dear Digium List
First of all, I thank all of you for all the replies and the interesting
suggestions. I thank you very much. I can only learn from people like
you. :-)
I will remember all the different solutions for a future use in other
scenarios.
On Mon, 2017-05-08 at 16:35 +0200, Frank Vanoni wrote:
> Is there a better solution?
At the end, I cleaned up my dial plan by removing the previous mess and
I'm using now ASTDB, as suggested, in the following way:
exten => 4000,1,Set(DB(alldevices/status)=OFF)
exten => 4000,2,Playback(service)
exten => 4001,1,Set(DB(alldevices/status)=ON)
exten => 4001,2,Playback(service)
exten => 2000,1,GotoIf($[${DB(alldevices/status)}=ON]?2001,1:2002,1)
exten => 2001,1,Dial(SIP/Dial(SIP/deviceA/deviceB/deviceC)
exten => 2002,1,Dial(SIP/Dial(SIP/deviceA/deviceB)
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[asterisk-users] Dial an extension to modify dialplan
Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension to dial in order to modify the dialplan. Here is what I did... In extensions.conf: -- snip - [mynicecontext] #include "ringdevice.conf exten => 2000,1,GoTo(ringdevice,ring,1) exten => 4000,1,System(/bin/cat /etc/asterisk/twodevices.txt > /etc/asterisk/ringdevice.conf) exten => 4000,2,Wait(3) exten => 4000,3,System(/usr/sbin/asterisk -rx "dialplan reload") exten => 4000,4,Playback(service) exten => 4001,1,System(/bin/cat /etc/asterisk/alldevices.txt > /etc/asterisk/ringdevice.conf) exten => 4001,2,Wait(3) exten => 4001,3,System(/usr/sbin/asterisk -rx "dialplan reload") exten => 4001,4,Playback(service) -- end snip - twodevices.txt contains exten => ring,1,Dial(SIP/deviceA) alldevices.txt contains exten => ring,1,Dial(SIP/deviceA/deviceC) By dialing 4000 or 4001, the dialplan is modified and reloaded accordingly. Is there a better solution? Frank -- _ -- 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] log incoming calls without answering
On Thu, 2017-04-20 at 17:26 -0300, Fabio Moretti wrote: > Any idea? I used to play with an analog telephone line and Asterisk by using a Linksys SPA-3102 Voice Gateway. I think it is no longer manufactured, but maybe you con buy a used one on eBay or you can find an equivalent device from another manufacturer. -- _ -- 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] Disallow CALLS without registry
On Sat, 2017-02-11 at 12:25 +1300, Pete Mundy wrote: > > sip.conf configuration > > In the [general] section, define: > > [general] > > ... > > allowguest=no > > alwaysauthreject=yes > > ... > With the above configuration on my Asterisk, I obtain the following result: - if the phone is registered to Asterisk, I can place any call according to the dial plan. - if the phone is NOT registered and I try to place a call, the phone obtains a "403 forbidden" at any calling attempt. Now, English is not my native language, but as far as I can understand, "forbidden" means "not allowed" or "disallowed". -- _ -- 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] Disallow CALLS without registry
On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий wrote: > so the main question is -- how to Disallow CALLS without registering > on PBX sip.conf configuration In the [general] section, define: [general] ... allowguest=no alwaysauthreject=yes ... The "allowguest" line disables anonymous SIP calls to your PBX. Some SIP providers connect as a guest user, however, so this may be inappropriate for your situation. Also, if you want to accept anonymous SIP calls, this line would block them, so you wouldn't want that. But it is listed here because it is the safest configuration. The "alwaysauthreject" line is important. This causes a hacker to get the same response from your PBX when they try to guess passwords whether or not they guessed a valid username. This also has the side-effect of making poorly written scanning scripts (the vast majority of hacker scripts seem to be poorly written) take less resources on your Asterisk box, as even if they scan a valid username, they'll think it doesn't exist. (Source: https://www.voip-info.org/wiki/view/Asterisk+security ) -- _ -- 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] MOBILE SIMCARD ON ASTERISK
Hi Chris On Tue, 2016-12-06 at 04:36 +0200, christopher kamutumwa wrote: > Is it possible to have a simcard configured and become incoming line > and outgoing on asterisk and also have the IVR function? Yes, it is possible! :-) A cheap solution is using a 3G-UBS-dongle. I have two SIM cards working in my Asterisk. I'm using two 3G-dongles (one for each SIM), a Huawei E173 (firmware 11.126.85.00.209) and a Huawei E180 (firmware 11.126.10.01.68). Google for "asterisk chan dongle" and you will find plenty of infos. -- _ -- 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] missed call notification
On Mon, 2016-11-28 at 14:31 +0100, tux john wrote:
> Hi. i am running asterisk 11 in debian and i would like have a missed
> call notification down to extension level.
> so if i get a missed call to extension 6589 then send an email to the
> user's email address with a subject and a text message.
> is there a guide on how to create something like that?
>From my extensions.conf
exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done)
exten => h,n,System(echo "Missed Call from ${CALLERID(num)}" | mail -s
"Missed Call from ${CALLERID(num)}" myemailaddr...@myemailprovider.com)
exten => h,n(done),NoOp()
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Re: [asterisk-users] Blacklist callers from file
On Sat, 2016-08-27 at 17:59 +0200, tux john wrote: > Hi. I would like to blacklist a few callers Example: callers with CallerID 0123456789, 9876543210 and 7410258963 are sent to tt-monkeys. Callers from area code 555 are also blocked. In "extensions.conf" file add #include "blacklist.conf" In "blacklist.conf" exten => s/0123456789,1,playback(tt-monkeys) exten => s/9876543210,1,playback(tt-monkeys) exten => s/7410258963,1,playback(tt-monkeys) exten => s/_555XXX,1,playback(tt-monkeys) ... .. . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference 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] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload
On Fri, 2016-08-26 at 10:12 -0300, Vitor Mazuco wrote: > bindaddr = all Try: bindaddr=0.0.0.0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference 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] rasberry pi
I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with Ubuntu Server 14.04. Works fine! :-) Frank On Wed, 2016-07-06 at 01:10 -0700, Thufir wrote: > I'm debating between a cloud PBX or, perhaps, rasberry pi. For a > SOHO, maybe three hardphones, rasberry pi would suffice? I would be > amazed, but, if so, great. -- _ -- 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] Including doesn't have any effect
On Mon, 2016-06-06 at 08:08 -0700, Steve Edwards wrote: > The purpose of a subroutine (code that is entered by a gosub and exited by > a return) is to allow the creation of easily reusable code. [snip] Steve Thank you very, very much for your answer. I really appreciated your interesting and detailed explanation. I'll go over the books again and rewrite the little "black box" taking in consideration your suggestions. Thanks again! Best regards Frank -- _ -- 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] Including doesn't have any effect
On Mon, 2016-06-06 at 17:47 +0100, Julian Beach wrote:
> exten => s,n,GotoIf(${DB_EXISTS(blacklist/${CDR(src)})}?block) ; Check
> whether caller blacklisted
As far as I know, Asterisk's database/blacklist function only supports
exact match of caller ID.
If you want to block a specific area code or a block of numbers (eg.
321-654-8XXX) the blacklist function is useless.
Correct me if I'm wrong.
Frank
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Re: [asterisk-users] Including doesn't have any effect
On Sat, 2016-06-04 at 15:19 -0700, Steve Edwards wrote: > Using a 'goto' to exit from a gosub is a bad idea. Why? > A better idea would be > to set a channel variable and check it's value after the return, in the > calling context. The idea is to update the blacklist.conf whenever I want to add or remove a specific number or an entire area code and leave the extensions.conf untouched and to avoid complex regular expressions. > Also, can a 'goto' in a subroutine reference an extension in the calling > context? Seems weird, but 'dialplan' is a weird language :) Well... I'm not an expert and my approach is by "trial and error". It works perfectly. :-) Frank -- _ -- 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] Including doesn't have any effect
Another possible approach to blacklist two (or more) specific callers (098765432 and 012345678 as example) In extension.conf #include "blaklist.conf" exten => _+x.,1,Gosub(blacklist,s,1) exten => _+x.,n, exten => black,1,playback(tt-monkeys) In blacklist.conf exten => s/098765432,1,Goto(black,1) exten => s/012345678,1,Goto(black,1) exten => s,1,Return() -- _ -- 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] How to set outgoing sip callid ?
In sip.conf [devicename] callerid="Jon Doe" <+123456789> or in extensions.conf exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) exten => 1234,n,Dial(SIP/ -- _ -- 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] VoipRaider is true for FREE calls?
On Mon, 2016-05-09 at 19:43 -0300, Vitor Mazuco wrote: > VoipRaider the site, says calls to landlines in Brazil... I hope I'm not infringing any mailing list rule by recommending you to take a look to the following providers. I use them with my Asterisk, the rates are good and they allow multiple calls. callwithus.com freelycall.com -- _ -- 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] VoipRaider is true for FREE calls?
VoipRaider is a service from DELLMONT SARL. This company offers voip services under dozens of different domains (voipcheap, voipdiscount, onevoip,...) Search "Dellmont Sarl" in Google and read the user's reviews. Personally, I would never send a penny to them... Franky -- _ -- 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] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements
Just a few ideas... 1. Disable all mobile carrier's voicemail and configure a voicemail on your Asterisk. Let Asterisk handle the unanswered calls. 2. If your SIP provider allows multiple calls at the same time, configure Asterisk to call all your SIMs at once (instead of calling the first, wait... calling the second... wait and so on). 3. If your mobile carrier blocks SIP on your data plan, simply configure Asterisk <-> SIP client on your mobile phone to use another port. Or, even better, you can use IAX instead of SIP. On your mobile device install a client that supports IAX (for example, Zoiper). Frank -- _ -- 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] How to install Huawei E153 in a Asterisk 11 or 13?
On Wed, 2016-03-02 at 19:12 -0300, Vitor Mazuco wrote: > I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but > my Huawei E153 is not working in my Asterisk. > But not successes. A little more information from you would be helpful to identify the problem. I have a Huawei USB 3G-stick and it works fine on Asterisk 11. Take a look here: http://www.raspberry-asterisk.org/documentation/gsm-voip-gateway-with-chan_dongle/comment-page-1/ Not all Huawei USB modems work out of the box, on some of them voice calling capability has to be enabled first, some need to be upgraded with the latest firmware. Details on this can be found on the original chan_dongle wiki. https://github.com/bg111/asterisk-chan-dongle/wiki/Preparation Before inserting the SIM into your modem please deactivate the PIN on your card. This can be done with any phone. Insert the SIM into your phone, deactivate PIN and you’re done. -- _ -- 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] Handle a call if one phone of a ring group is busy
On Sun, 2016-02-28 at 01:43 +0100, Frank wrote:
> Question: How to give a "busy signal" back to the caller if one
> extension of a ring group is in use? Or redirect the call to voice mail?
Found a solution! :-)
exten => 7654321,1,GotoIf($["${DEVICE_STATE(SIP/111)}"="INUSE"]?Busy,1)
exten => 7654321,n,GotoIf($["${DEVICE_STATE(SIP/222)}"="INUSE"]?Busy,1)
exten => 7654321,n,GotoIf($["${DEVICE_STATE(SIP/333)}"="INUSE"]?Busy,1)
exten => 7654321,n,Dial(SIP/111/222/333)
exten => Busy,1,BUSY(10)
exten => Busy,n,Hangup
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[asterisk-users] Handle a call if one phone of a ring group is busy
Hi List I have a phone in my living room (ext. 111), a phone in the kitchen (ext. 222) and a phone in my bedroom (ext. 333). Both phones are part of a ring group. exten => 7654321,1,Dial(SIP/111/222/333) Everything work fine and, as expected, all phones are ringing by an incoming call and I can answer the call on the nearest phone. Problem: if there is a second incoming call while I'm talking on one of the three phones, the other phones ring and I cannot answer the second call since I cannot be at two places at the same time. :-) Question: How to give a "busy signal" back to the caller if one extension of a ring group is in use? Or redirect the call to voice mail? Any hint? Thanks in advance Frank -- _ -- 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] Voice recognition IVR Is it possible?
On Tue, 2016-02-23 at 22:05 +, Lefteris Zafiris wrote: > google indeed makes it very hard to figure out how to enable the speech API > and > get a key. I guess it is intentional () I really appreciated your hint. Many thanks! At the end, after checking several web sites, I gave up since everything seems a little too complicated for something I just wanted to try as a test. Thanks again! Frank :-) -- _ -- 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] Voice recognition IVR Is it possible?
On Tue, 2016-02-23 at 17:06 +, Steve Howes wrote: > Google?... Yeah... searched "google voice recognition api asterisk", browsed though various results. Nothing helpful for a beginner, very confusing bla bla... Thanks anyway for your help. F. -- _ -- 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] Voice recognition IVR Is it possible?
On Tue, 2016-02-23 at 00:43 +0100, Laszlo wrote: > > Requirements > > ... > Speech API key from Google Yes... OK... but... where and how can I obtain this API Key? Where and how do I install it into my Asterisk box? -- _ -- 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] Voice recognition IVR Is it possible?
Hi Daniel On Mon, 2016-02-22 at 19:40 +0100, Daniel Heckl wrote: > try this http://zaf.github.io/asterisk-speech-recog/. > I have tested it myself, it works very well. I wanted to try it, but I obtain the following error message: "speech-recog.agi,en-US: API key is missing. Aborting. " :-( What am I missing? Any hint? Frank -- _ -- 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] Windstream SIP Trunk settings
On Mon, 2016-02-22 at 08:20 -0500, James Cass wrote: > register string: :@:5060 Try: register => 5551231234:sec...@sipdomain.com/5551231234 -- _ -- 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] [dongle0] timedout while waiting 'OK' in response to 'AT'
On Fri, 2016-02-12 at 14:33 -0200, Vitor Mazuco wrote: > Yes I used. > > The problem can be the version of Asterisk? > > I use Asterisk 13 instead of 11. Try [dongle0] imei=347654458453667 imsi=976895757545778 -- _ -- 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] NAT traversal for mobile app softphones - best strategy?
Hi On Fri, 2016-02-05 at 00:44 +, Kevin Long wrote: > My asterisk systems sit behind a Meraki mx80 firewall at a data > center. I use static public IPs on the firewall and port forward > 5060,5061, and 10,000-20,000 so the clients can connect. > Given this scenario, I’m hoping for advice on the best strategy I have the same situation and with the following sip.conf settings, everything works fine. [general] externip= 12.34.56.78 localnet = 192.168.10.0/255.255.255.0 nat=force_rport,comedia bindport=5060 bindaddr=0.0.0.0 srvlookup=no dtmfmode=rfc2833 canreinvite=no disallow=all allow=alaw allow=ulaw tcpenable=yes Here the configuration for a mobile device with a softphone (Android and Zoiper) ;Mobile phone [mobile1] type=peer callerid="Frank " <+987654321> nat=force_rport,comedia qualify=6000 host=dynamic secret=mysupersecretpassword canreinvite=no context=privatephone call-limit=2 transport=tcp -- _ -- 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] Peer Reachable / Unreachable on TLS
Hello Asterisk users :-) Server: Asterisk 11.7.0~dfsg-1ubuntu1 on Raspberry Client: Zoiper on Android device If "Transport=tcp" everything works fine, without any trouble. If "Transport=tls" registration is fine. But after a few minutes, the peer continuously is "reachable...unreachable". -- [2016-02-04 11:00:38] NOTICE[2179]: chan_sip.c:29427 sip_poke_noanswer: Peer 'android1' is now UNREACHABLE! Last qualify: 9 -- Registered SIP 'android1' at 192.168.10.22:41270 [2016-02-04 11:01:26] NOTICE[20576]: chan_sip.c:23522 handle_response_peerpoke: Peer 'android1' is now Reachable. (7ms / 6000ms) [2016-02-04 11:04:35] NOTICE[2179]: chan_sip.c:29427 sip_poke_noanswer: Peer 'android1' is now UNREACHABLE! Last qualify: 28 -- Registered SIP 'android1' at 192.168.10.22:48515 [2016-02-04 11:05:27] NOTICE[20670]: chan_sip.c:23522 handle_response_peerpoke: Peer 'android1' is now Reachable. (17ms / 6000ms) [2016-02-04 11:09:38] NOTICE[2179]: chan_sip.c:29427 sip_poke_noanswer: Peer 'android1' is now UNREACHABLE! Last qualify: 43 -- Registered SIP 'android1' at 192.168.10.22:52955 [2016-02-04 11:10:03] NOTICE[20737]: chan_sip.c:23522 handle_response_peerpoke: Peer 'android1' is now Reachable. (49ms / 6000ms) -- The same happen with two other devices. TCP is fine, TLS is not. As a test, I configured two accounts on the same device with identical parameters, except one is TLS and the other is TCP. The result is the same: TCP is fine, TLS is intermittent. Any idea? Thanks for any hint Frank [android1] type=peer callerid="Abcd Efgh" <+1234567890> nat=force_rport,comedia qualify=6000 host=dynamic secret=qt528frh3bAW3 tcanreinvite=no context=venomphone call-limit=2 transport=tls encryption=yes -- _ -- 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] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11
On Mon, 2016-01-11 at 14:52 +0100, Juergen Sauer wrote: > It seems to be, that this fw can not deal with not-numeric-sip accounts. > I entered the extension number as name, account and it works. Glad to hear that. Very interesting. Good to know! > Solved by my self, using Try-and-error Metodic. :) You can be proud for your troubleshooting. :-) -- _ -- 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] ST2030 replacement
On Thu, 2016-01-07 at 17:35 +0100, Sil wrote: > Can you give me a return on the models you use ? Yealink T26P -- _ -- 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] placing calls with linphone.org SIP account
On Wed, 2016-01-06 at 11:27 +0100, Yves wrote: > how can I call other users > registered at other SIP-Providers? > I tried all well-known SIP URI Syntaxes but none worked... does anyone > reliably know, if it is possible at all and if so, what is the > dialstring looking like? It depends if the "other SIP-Provider" accepts calls from other networks. Some providers accept calls from other networks. Unfortunately, most providers do not. -- _ -- 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] Manipulating of a dialed sequence
Hi Asterisk List Given, as an example, the following sequence 012345*543210 I would like to store into a variable all digits before "*" (012345) and in a different variable all digits after the "*" (543210) for further processing in the dial plan. The length of the dialed sequence may be variable and "*" is the separator between the two values to store. Any idea? Thanks Francesco -- _ -- 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] Manipulating of a dialed sequence
Steve and Rafael, that is exactly what I was looking for! Many thanks for your help! :-) -- _ -- 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] How to hang-up a FXO call without answering it?
I'm using Asterisk 13.4.0 and DAHDI 2.10.2. I've got a FXO line that I
use for in and outgoing PSTN calls. Unfortunately I'm getting a lot of
spam calls on the number.
I had the extension configured to forward incoming calls to 2 SIP
extensions or go to voicemail. But now I'm getting loads of junk
voicemail messages, so I removed the voicemail command:
[from-pstn]
exten => s,1,Wait(1)
exten => s,2,Set(WHO=${CALLERID(num)})
exten => s,3,Verbose(CALLERID is ${CALLERID(num)})
exten => s,4,Verbose(Time is ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten => s,5,Dial(SIP/1000/1100,30)
;exten => s,6,Voicemail(1000,u)
exten => s,6,Hangup()
Now incoming calls will cause the SIP extensions to ring for 30 seconds,
but then the FXO line isn't disconnected. The [from-pstn] context seems
to keep looping on the Dial() command:
[Sep 19 11:16:59] -- Starting simple switch on 'DAHDI/4-1'
[Sep 19 11:17:00] -- Executing [s@from-pstn:1] Wait("DAHDI/4-1",
"1") in new stack
[Sep 19 11:17:01] -- Executing [s@from-pstn:2] Set("DAHDI/4-1",
"WHO=919961") in new stack
[Sep 19 11:17:01] -- Executing [s@from-pstn:3] Verbose("DAHDI/4-1",
"CALLERID is 919961") in new stack
[Sep 19 11:17:01] CALLERID is 919961
[Sep 19 11:17:01] -- Executing [s@from-pstn:4] Verbose("DAHDI/4-1",
"Time is 20150919-111701") in new stack
[Sep 19 11:17:01] Time is 20150919-111701
[Sep 19 11:17:01] -- Executing [s@from-pstn:5] Dial("DAHDI/4-1",
"SIP/1000/1100,30") in new stack
[Sep 19 11:17:01] == Using SIP RTP TOS bits 184
[Sep 19 11:17:01] == Using SIP RTP CoS mark 5
[Sep 19 11:17:01] == Using SIP RTP TOS bits 184
[Sep 19 11:17:01] == Using SIP RTP CoS mark 5
[Sep 19 11:17:01] -- Called SIP/1000
[Sep 19 11:17:01] -- Called SIP/1100
[Sep 19 11:17:01] -- SIP/1000-0095 is ringing
[Sep 19 11:17:01] -- SIP/1100-0096 is ringing
[Sep 19 11:17:31] -- Nobody picked up in 3 ms
[Sep 19 11:17:31] -- Executing [s@from-pstn:6] Hangup("DAHDI/4-1",
"") in new stack
[Sep 19 11:17:31] == Spawn extension (from-pstn, s, 6) exited non-zero
on 'DAHDI/4-1'
[Sep 19 11:17:31] -- Hanging up on 'DAHDI/4-1'
[Sep 19 11:17:31] -- Hungup 'DAHDI/4-1'
[Sep 19 11:17:35] -- Starting simple switch on 'DAHDI/4-1'
[2015-09-19 11:17:39.1] ERROR[27434][C-0079]: callerid.c:567
callerid_feed: No start bit found in fsk data.
[2015-09-19 11:17:39.1] WARNING[27434][C-0079]: chan_dahdi.c:1374
my_get_callerid: Failed to decode CallerID
[2015-09-19 11:17:39.1] WARNING[27434][C-0079]: sig_analog.c:2569
__analog_ss_thread: CallerID returned with error on channel 'DAHDI/4-1'
[Sep 19 11:17:39] -- Executing [s@from-pstn:1] Wait("DAHDI/4-1",
"1") in new stack
[Sep 19 11:17:40] -- Executing [s@from-pstn:2] Set("DAHDI/4-1",
"WHO=") in new stack
[Sep 19 11:17:40] -- Executing [s@from-pstn:3] Verbose("DAHDI/4-1",
"CALLERID is ") in new stack
[Sep 19 11:17:40] CALLERID is
[Sep 19 11:17:40] -- Executing [s@from-pstn:4] Verbose("DAHDI/4-1",
"Time is 20150919-111740") in new stack
[Sep 19 11:17:40] Time is 20150919-111740
[Sep 19 11:17:40] -- Executing [s@from-pstn:5] Dial("DAHDI/4-1",
"SIP/1000/1100,30") in new stack
[Sep 19 11:17:40] == Using SIP RTP TOS bits 184
[Sep 19 11:17:40] == Using SIP RTP CoS mark 5
[Sep 19 11:17:40] == Using SIP RTP TOS bits 184
[Sep 19 11:17:40] == Using SIP RTP CoS mark 5
[Sep 19 11:17:40] -- Called SIP/1000
[Sep 19 11:17:40] -- Called SIP/1100
[Sep 19 11:17:40] -- SIP/1000-0097 is ringing
[Sep 19 11:17:40] -- SIP/1100-0098 is ringing
[Sep 19 11:17:49] == Spawn extension (from-pstn, s, 5) exited non-zero
on 'DAHDI/4-1'
[Sep 19 11:17:49] -- Hanging up on 'DAHDI/4-1'
[Sep 19 11:17:49] -- Hungup 'DAHDI/4-1'
The caller just hears the line ring and ring and the SIP extensions are
dialed over and over until the caller hangs-up.
Is there anyway to force a hang-up or disconnection of the incoming call
if the SIP extensions don't answer?
I'd like to do this without actually answering the call if at all possible.
Frank
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[asterisk-users] Anyone running Asterisk on KVM virtual machine under SmartOS?
I'm thinking of condensing some of my boxes down to KVM virtual machines running under SmartOS. My Asterisk box is running Centos 6.4 and I'd like to include it. Is anyone running Asterisk on a virtual machine under SmartOS? Does DAHDI work? Thanks in advance. Frank -- _ -- 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] GSM Sip Gateway
So what is the official page to get those GoIP ? All I can find is on ebay.. On 2/24/13 5:23 AM, longst wrote: I am using GoIp 1 channel gateway. it is fine Sent from Shitian Long On Feb 24, 2013, at 1:54 AM, Frank fr...@efirehouse.com wrote: Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c If anyone has any (good) experience with another brand, I'll take the names and models. Thanks -- _ -- 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
Re: [asterisk-users] GSM Sip Gateway
Ha... Here is the other company I was looking for: http://www.yx.cl Anyone is using their GSM gateways ? On 2/26/13 11:56 AM, Frank wrote: So what is the official page to get those GoIP ? All I can find is on ebay.. On 2/24/13 5:23 AM, longst wrote: I am using GoIp 1 channel gateway. it is fine Sent from Shitian Long On Feb 24, 2013, at 1:54 AM, Frank fr...@efirehouse.com wrote: Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c If anyone has any (good) experience with another brand, I'll take the names and models. Thanks -- _ -- 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 -- _ -- 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] GSM Sip Gateway
USA, this will be use with a 4G network. On Feb 24, 2013, at 5:24 AM, longst longst...@gmail.com wrote: where are you from by the way Sent from Shitian Long On Feb 24, 2013, at 1:54 AM, Frank fr...@efirehouse.com wrote: Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c If anyone has any (good) experience with another brand, I'll take the names and models. Thanks -- _ -- 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
Re: [asterisk-users] GSM Sip Gateway
So anyone would know a gateway working on 3G/4G network ? I remember a website called XY something (I cant find it anymore. I don t remember if it was xywireless.com , or xytelecom.com , or something else) where they seemed to have good gateways, but I can't find it anymore. On 2/24/13 9:15 AM, John Novack wrote: From the Freq. list given on eBay, I don't thinkthey are. The listed freqs. are worldwide GSM since the mid 90's, but not 4G John Novack Hans Witvliet wrote: Are these 4G comaptible -Original Message- From: Frankfr...@efirehouse.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] GSM Sip Gateway Date: Sun, 24 Feb 2013 07:40:19 -0500 USA, this will be use with a 4G network. On Feb 24, 2013, at 5:24 AM, longstlongst...@gmail.com wrote: where are you from by the way Sent from Shitian Long On Feb 24, 2013, at 1:54 AM, Frankfr...@efirehouse.com wrote: Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c If anyone has any (good) experience with another brand, I'll take the names and models. Thanks -- _ -- Bandwidth and Colocation Provided byhttp://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 byhttp://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 byhttp://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 byhttp://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 -- Dog is my Co-pilot -- _ -- 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] GSM Sip Gateway
Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c If anyone has any (good) experience with another brand, I'll take the names and models. Thanks -- _ -- 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] Asterisk Realtime Extension... strange behaviour
Remove the line _X. , and try 3 digits other than 110 112 , let us know
if it works.
On 2/12/13 5:55 AM, Yves A. wrote:
Hi,
I encountered a strange behaviour using realtime extensions... (on
Asterisk 11.2)
when I use the following static dialplan, everything works as expected..:
[from-sip]
exten = 110,1,Dial(DAHDI/g0/${EXTEN})
exten = 112,1,Dial(DAHDI/g0/${EXTEN})
exten = _XXX,1,Dial(SIP/${EXTEN})
exten = _X.,1,Dial(DAHDI/g0/${EXTEN})
will say... if a sip phone calls 110 or 112 the call is routed into
PSTN (german emergency call)
if a sip phone calls any three digit number, the call should be routet
to the corresponding SIP user
and if a sip phone calls any other number the call should be routed into
PSTN... thats ok and works as expected.
when I change to realtime:
[from-sip]
switch = Realtime
and put the diaplan into the database
idcontextextenpriorityappappdata
1from-sip1101DialDAHDI/g0/${EXTEN}
2from-sip1121DialDAHDI/g0/${EXTEN}
3from-sip_XXX1DialSIP/${EXTEN}
4from-sip_X.1DialDAHDI/g0/${EXTEN}
only the emergency calls work and any other call goes to DAHDI... I cant
reach any other SIP phone.
Even when swapping the content of the rows 3 and 4 in the database to
idcontextextenpriorityappappdata
1from-sip1101DialDAHDI/g0/${EXTEN}
2from-sip1121DialDAHDI/g0/${EXTEN}
3from-sip_X.1DialDAHDI/g0/${EXTEN}
4from-sip_XXX1DialSIP/${EXTEN}
makes no difference...
I thought, using realtime extensions would read the dialplan from top to
bottom, ordered by id... but it
seems to be ignored somehow and the extension _X. catches the calls
before the extensionpattern _XXX is reached.
I _could_ avoid this be prefixing external numbers with a leading 0
for example... but I dont want to... as I said.. using
static extension via extensions.conf the dialplan works as expected...
Am I missing something?
regards,
yves
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[asterisk-users] Google talk not (re)connecting after network down
Hi all, I notice yesterday night while doing tests of uptime that if I unplug my network from the internet, then plug it back, my jabber still shows connection to google, but no outgoing calls are going out, and nothing is coming in (calls are going in google vmail since there is no connection to the pbx) My way of restarting jabber was to kill asterisk and restart it. I'm sure I could have unload then reload the module. Is there any safe feature that can make sure that when Jabber shows CONNECTED , it *IS* actually connected ? Thanks folks ! -- _ -- 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] Asterisk calls between 2 private networks
My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no The Digium on network B can register. I can see it when I do sip show peer xxx. When the phones are calling each other, the signaling is working. They ring. But when they pick up, there is no audio, in any way. Has anyone ever worked on the same configuration, and had success ? If yes, I'd love to hear your story and check your configuration. Thanks ! -- _ -- 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] Asterisk calls between 2 private networks
AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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] Asterisk calls between 2 private networks
I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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
Re: [asterisk-users] Asterisk calls between 2 private networks
I'm using Digium Phones. I still do not understand why it's not possible to do it the way the networks are right now. If the options I mentioned in my sip.conf are enough, then both phones should use Asterisk as a proxy, and Asterisk should handle all the media. I will run tcpdump traces tonight and will check it out. My router has a bug and won't let me mirror port. From tech support I need to reflash it. I'll do it and try it again. F. On 2/7/13 12:59 PM, Christopher Harrington wrote: Digium phones, which (as far as I can tell with my experience) do not support VPN yet. On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen jkil...@allamericanasphalt.com mailto:jkil...@allamericanasphalt.com wrote: Or if it's just a couple phones, you might be able to setup a vpn connection directly on the phone itself - have it vpn into 'HQ' and get an address on that network. I'm not sure which phones you're using though or what phones support that setup. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, February 07, 2013 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public IP of network A My SIP.CONF has: nat=yes localnet=192.168.1.0/255.255.255.0 http://192.168.1.0/255.255.255.0 externaddr=public_ip_of_network_a directmedia=no My (lazy) solution to this problem was to throw hardware at it . Bearing in mind that Asterisk will run on just about any old scrapper (or even a Raspberry Pi, if you feel so inclined), there's little point even trying to send SIP over the Internet. Just have an Asterisk box at each end, and then you only need a much simpler-to-configure IAX trunk between the two. The routers at each end then just need one port -- UDP 4569 -- forwarded to the Asterisk box (if it isn't configured as the default DMZ machine). -- _ -- 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
Re: [asterisk-users] Asterisk calls between 2 private networks
i think canreinvite is not part of Asterisk 1.8 anymore. Asterisk 1.8 added directmediapermit and directmediadeny to limit which peers can send direct media to each other. On 2/7/13 1:15 PM, Kevin Larsen wrote: Did you set canreinvite=no in sip.conf on the phone in network B? A phone that can connect but loses audio is almost a sure sign that it is reinviting and your rtp packets are not making it to the phone. By turning canreinvite off, it will keep asterisk in the middle of your sessions and should give you the audio. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: Frank fr...@efirehouse.com To: ch...@acsdi.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 02/07/2013 12:06 PM Subject: Re: [asterisk-users] Asterisk calls between 2 private networks Sent by: asterisk-users-boun...@lists.digium.com I'm using Digium Phones. I still do not understand why it's not possible to do it the way the networks are right now. If the options I mentioned in my sip.conf are enough, then both phones should use Asterisk as a proxy, and Asterisk should handle all the media. I will run tcpdump traces tonight and will check it out. My router has a bug and won't let me mirror port. From tech support I need to reflash it. I'll do it and try it again. F. On 2/7/13 12:59 PM, Christopher Harrington wrote: Digium phones, which (as far as I can tell with my experience) do not support VPN yet. On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen jkil...@allamericanasphalt.com mailto:jkil...@allamericanasphalt.com wrote: Or if it's just a couple phones, you might be able to setup a vpn connection directly on the phone itself - have it vpn into 'HQ' and get an address on that network. I'm not sure which phones you're using though or what phones support that setup. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, February 07, 2013 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering
Re: [asterisk-users] Asterisk calls between 2 private networks
And actually I did not have directmediadeny=0.0.0.0 But I had directmedia=no. So I will add the directmediadeny line, and will check it out again tonight. On 2/7/13 1:22 PM, Frank wrote: i think canreinvite is not part of Asterisk 1.8 anymore. Asterisk 1.8 added directmediapermit and directmediadeny to limit which peers can send direct media to each other. On 2/7/13 1:15 PM, Kevin Larsen wrote: Did you set canreinvite=no in sip.conf on the phone in network B? A phone that can connect but loses audio is almost a sure sign that it is reinviting and your rtp packets are not making it to the phone. By turning canreinvite off, it will keep asterisk in the middle of your sessions and should give you the audio. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: Frank fr...@efirehouse.com To: ch...@acsdi.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 02/07/2013 12:06 PM Subject: Re: [asterisk-users] Asterisk calls between 2 private networks Sent by: asterisk-users-boun...@lists.digium.com I'm using Digium Phones. I still do not understand why it's not possible to do it the way the networks are right now. If the options I mentioned in my sip.conf are enough, then both phones should use Asterisk as a proxy, and Asterisk should handle all the media. I will run tcpdump traces tonight and will check it out. My router has a bug and won't let me mirror port. From tech support I need to reflash it. I'll do it and try it again. F. On 2/7/13 12:59 PM, Christopher Harrington wrote: Digium phones, which (as far as I can tell with my experience) do not support VPN yet. On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen jkil...@allamericanasphalt.com mailto:jkil...@allamericanasphalt.com wrote: Or if it's just a couple phones, you might be able to setup a vpn connection directly on the phone itself - have it vpn into 'HQ' and get an address on that network. I'm not sure which phones you're using though or what phones support that setup. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Thursday, February 07, 2013 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I don't see how that would really solve anything - instead of the server sending the 192.168.x.x packets onto the local network, it will send them up toward the internet and get black-holed. What probably makes more sense would be to switch the subnet on one of the networks, AND put up a vpn between them, adding the routes for the private networks to cross thru the tunnels. Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same address block. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am envisaging. But I would really love to try to work out with my issue first. It will allow me to deploy more phones in separates buildlings in the future. If I do the IAX solution, it means that for every building, I need a box.. Which I would like to prevent. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone
[asterisk-users] [FIXED] Asterisk calls between 2 private networks
Got it to work tonight. So once again this is my network: Network A: 192.168.1.x Network B: 192.168.1.x In between, the internet. Asterisk is in Network A. 1 Digium phone is in network A. Router from network A does NAT and forward (for now): - 5060 TCP/UDP to internal IP of asterisk - 10k-20k TCP/UDP to internal IP of asterisk -I know TCP is not needed, but I will remove little by little options tomorrow, since nothing was working before- Network B has 1 Digium phone, that registers to the public IP of network A. My SIP.CONF looks like that for now: [general] context=unauthenticated allowguest=no transport=udp dtmfmode=auto nat=yes localnet=192.168.1.0/255.255.255.0 externaddr=network_a_public_ip_address directmedia=no [100] type=friend context=LocalSets host=dynamic disallow=all allow=ulaw host=dynamic secret=xxx mailbox=100@default [200] type=friend context=LocalSets host=dynamic disallow=all allow=ulaw secret=xxx mailbox=200@default nat=yes qualify=yes directmedia=no I added a file rtp.conf: [general] rtpstart=1 rtpend=10200 that's all folks ! -- _ -- 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] CallerID external call after Attended Transfer
What is the PAI option below that you are talking about, for sendrpid ? The manual only says that yes or no can be used.. On 2/4/13 9:39 AM, Kevin Larsen wrote: One thing you can try is to set the following in your sip.conf. sendrpid=pai trustrpid=yes You can put that on individual phone configurations in sip.conf or, as I do, in a template that is applied to a set of phones. I believe that was what I had to set so that the remote caller ID would show up properly on my Polycom phones. I made no changes to the Polycom configuration to make it work. It might work with the Yealink T32G phones as well. In the case originally presented, I get the following: Call comes into Operator showing cell phone caller id. Operator performs an attended transfer. I get the Operator caller ID. Upon completion of the transfer, I get the cell phone caller ID. If a blind transfer is performed, I get the cell phone caller ID (there might be a flash of the operators caller ID for just the split second it takes her to hit the transfer button a second time to turn it from attended to blind transfer on my phones). Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: Steven Howes steve-li...@geekinter.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 02/04/2013 08:31 AM Subject: Re: [asterisk-users] CallerID external call after Attended Transfer Sent by: asterisk-users-boun...@lists.digium.com On 4 Feb 2013, at 13:45, Jonas Kellens wrote: The IP-phones in this case are Yealink T32G. What setting is needed in this IP-phone ? Quick google doesn't turn up any results. Handsets probably dont support it. Steve-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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
Re: [asterisk-users] Google voice with no voice
Chris,
I covered the whole 74.125.225.* subnet.
Even if I open the ports mentioned below for all (not limited to IP
addresses) I still have the same issue.
Have anyone ever succeeded in such configuration? :
Digium phones on 2 different private networks (2 different buildings)
Asterisk server in the internet with a public IP
Use Google Voice
Even if you have asterisk on a private network, but have the same kind
of solution working for you, I'd love to hear your story..
On 1/22/13 9:55 AM, Christopher Harrington wrote:
On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com
mailto:fr...@efirehouse.com wrote:
Actually, the funny thing is that it works randomly.
This may be due to the fact that voice.google.com
http://voice.google.com actually resolves to a range of IP addresses.
When you set up your firewall, it may not be including all of the
possible resolutions for voice.google.com...
voice.l.google.com http://voice.l.google.com.300INA74.125.225.36
voice.l.google.com http://voice.l.google.com.300INA74.125.225.46
voice.l.google.com http://voice.l.google.com.300INA74.125.225.33
voice.l.google.com http://voice.l.google.com.300INA74.125.225.32
voice.l.google.com http://voice.l.google.com.300INA74.125.225.41
voice.l.google.com http://voice.l.google.com.300INA74.125.225.38
voice.l.google.com http://voice.l.google.com.300INA74.125.225.35
voice.l.google.com http://voice.l.google.com.300INA74.125.225.39
voice.l.google.com http://voice.l.google.com.300INA74.125.225.40
voice.l.google.com http://voice.l.google.com.300INA74.125.225.34
voice.l.google.com http://voice.l.google.com.300INA74.125.225.37
(ie 74.125.225.32-41 and 74.125.225.46)
Since these are short TTL values (the 300 means 5 minutes) there may be
a brief period where your devices and your firewall agree, before one or
both change their mind about the IP address behind that hostname.
I just tried out of the blue calling from D70 through Google Voice
to a cell phone, and it worked. I hung up, redial, and no audio at all.
On 1/21/13 10:38 PM, Frank wrote:
Greetings all,
I was reading the documentation tonight, and decided to try
Google voice
with my asterisk.
I was able to setup iksemel, connect to google using jabber, and
connect
to google voice using gtalk.
Here is my physical configuration:
Digium D70 -- private network 192.168.1.x -- Airport express --
Internet -- Asterisk with public IP
My asterisk has the following ports open:
5060 tcp/udp from my Airport Express public IP and from
voice.google.com http://voice.google.com
10,000:20,000 from my Airport Express public IP and from
voice.google.com http://voice.google.com
My issue is that when I place a call with google voice, I have
no audio
path at all in both way.
When a call is received on google voice (and sent to the D70),
if I pick
up, nothing happen, and the caller still hear the ringing tone.
My D70 is setup as follow in the sip.conf:
[D70]
type=friend
nat=yes
qualify=yes
directmedia=no
host=dynamic
secret=takapoum
disallow=all
allow=ulaw
context=LocalSets
mailbox=D70@default
my gtalk.conf is setup as follow:
[general]
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk_incoming
connection=asterisk
and finally, the interesting parts in my extensions.conf are
setup as
follow:
;Dialing out on google voice:
exten =
_1zxxzxx,1,Dial(Gtalk/__asterisk/+${EXTEN}@voice.__google.com
mailto:exten...@voice.google.com)
same = n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten = r...@gmail.com mailto:r...@gmail.com,1,Verbose(0,
Incoming gtalk from ${CALLERID(all)})
same = n,Answer()
same = n,Wait(2)
same = n,Dial(SIP/D70)
same = Hangup()
I would appreciate if anyone could give me a hint about the
audio path.
This is a project that we I will try to setup in a small fire
department, and before I try it, I would like to make sure that my
Digium phones will be able to get full audio path behind private
networks.
Thanks a ton for the help !
--
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Re: [asterisk-users] Google voice with no voice
Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http://voice.l.google.com.300INA74.125.225.32 voice.l.google.com http://voice.l.google.com.300INA74.125.225.41 voice.l.google.com http://voice.l.google.com.300INA74.125.225.38 voice.l.google.com http://voice.l.google.com.300INA74.125.225.35 voice.l.google.com http://voice.l.google.com.300INA74.125.225.39 voice.l.google.com http://voice.l.google.com.300INA74.125.225.40 voice.l.google.com http://voice.l.google.com.300INA74.125.225.34 voice.l.google.com http://voice.l.google.com.300INA74.125.225.37 (ie 74.125.225.32-41 and 74.125.225.46) Since these are short TTL values (the 300 means 5 minutes) there may be a brief period where your devices and your firewall agree, before one or both change their mind about the IP address behind that hostname. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. On 1/21/13 10:38 PM, Frank wrote: Greetings all, I was reading the documentation tonight, and decided to try Google voice with my asterisk. I was able to setup iksemel, connect to google using jabber, and connect to google voice using gtalk. Here is my physical configuration: Digium D70 -- private network 192.168.1.x -- Airport express -- Internet -- Asterisk with public IP My asterisk has the following ports open: 5060 tcp/udp from my Airport Express public IP and from voice.google.com http://voice.google.com 10,000:20,000 from my Airport Express public IP and from voice.google.com http://voice.google.com My issue is that when I place a call with google voice, I have no audio path at all in both way. When a call
Re: [asterisk-users] Google voice with no voice
Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http://voice.l.google.com.300INA74.125.225.32 voice.l.google.com http://voice.l.google.com.300INA74.125.225.41 voice.l.google.com http://voice.l.google.com.300INA74.125.225.38 voice.l.google.com http://voice.l.google.com.300INA74.125.225.35 voice.l.google.com http://voice.l.google.com.300INA74.125.225.39 voice.l.google.com http://voice.l.google.com.300INA74.125.225.40 voice.l.google.com http://voice.l.google.com.300INA74.125.225.34 voice.l.google.com http://voice.l.google.com.300INA74.125.225.37 (ie 74.125.225.32-41 and 74.125.225.46) Since these are short TTL values (the 300 means 5 minutes) there may be a brief period where your devices and your firewall agree, before one or both change their mind about the IP address behind that hostname. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. On 1/21/13 10:38 PM, Frank wrote: Greetings all, I was reading the documentation tonight, and decided to try Google voice with my asterisk. I was able to setup iksemel, connect to google using jabber, and connect
Re: [asterisk-users] Google voice with no voice
Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com
Re: [asterisk-users] Google voice with no voice
*CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial
Re: [asterisk-users] Google voice with no voice
*CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls
Re: [asterisk-users] Google voice with no voice
That's idle. If I call from D70 (working scenario) the result of the command is the same. gtalk show channels shows this when I call from D70 (again, working scenario): Channel Jabber ID Resource Read Write Gtalk/+1x@voice.googl +1xx...@voice.google.com srvres-MTAuMjI3 ulaw ulaw When I call google voice, gtalk show channels shows the following: While ringing: *CLI gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw slin 1 active gtalk channel Once I pick up *CLI -- SIP/D70-0004 answered Gtalk/+xxx-2c8e gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw ulaw 1 active gtalk channel The only difference is the WRITE column that changes from SLIN to ULAW On 1/22/13 2:22 PM, Danny Nicholas wrote: This is incoming, outgoing or idle (no call)? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:21 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice
Re: [asterisk-users] Google voice with no voice
OK, so here is the new.. By mistake, when I picked up the D70 , I pushed the 2 button. I suddenly heard google voice saying Okay, I'll send the caller to voicemail. So I called again.. picked up.. I could not hear anything on the D70.. But if I push 1 (which is the google voice option to pickup the screened call), then the audio path works in both way. So the real issue is that when google voice talks when I pick up to let me know who's calling, I can't hear anything, until I press a digit. If I press 1, I get the call connected. If I press 2, I can hear google voice. The question is why can't I hear google voice right away without pushing a digit ? I tried to go into google voice configuration and remove the call screening, but it looks like for calls on gtalk , the screening is always active. So I guess I will know that I need to press 1 or 2 from the D70 for everything to work. It slightly sucks, but I'll take it. On 1/22/13 2:29 PM, Danny Nicholas wrote: This sounds like a codec issue. Set your verbose to 10 and retry the incoming call. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:26 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice That's idle. If I call from D70 (working scenario) the result of the command is the same. gtalk show channels shows this when I call from D70 (again, working scenario): Channel Jabber ID Resource Read Write Gtalk/+1x@voice.googl +1xx...@voice.google.com srvres-MTAuMjI3 ulaw ulaw When I call google voice, gtalk show channels shows the following: While ringing: *CLI gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw slin 1 active gtalk channel Once I pick up *CLI -- SIP/D70-0004 answered Gtalk/+xxx-2c8e gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw ulaw 1 active gtalk channel The only difference is the WRITE column that changes from SLIN to ULAW On 1/22/13 2:22 PM, Danny Nicholas wrote: This is incoming, outgoing or idle (no call)? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:21 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used
Re: [asterisk-users] Google voice with no voice
Hi , So I tried Answer() Wait(1) SendDTMF(1) But I got an error in the console: [Jan 22 14:54:13] WARNING[28067]: pbx.c:4458 pbx_extension_helper: No application 'SendDTMF' for extension (gtalk_incoming, r...@gmail.com, 4) If I do core show application sendDTMF , nothing comes up. If there anything special to compile for this ? Thanks On 1/22/13 2:54 PM, Joshua Colp wrote: Frank wrote: OK, so here is the new.. By mistake, when I picked up the D70 , I pushed the 2 button. I suddenly heard google voice saying Okay, I'll send the caller to voicemail. So I called again.. picked up.. I could not hear anything on the D70.. But if I push 1 (which is the google voice option to pickup the screened call), then the audio path works in both way. So the real issue is that when google voice talks when I pick up to let me know who's calling, I can't hear anything, until I press a digit. If I press 1, I get the call connected. If I press 2, I can hear google voice. The question is why can't I hear google voice right away without pushing a digit ? This is a Google Voice thing. Even the Google talk client itself sends a digit of 1 when you answer the call. That being said you can do this from inside of Asterisk dialplan with a combination of Answer, Wait, and SendDTMF(1) -- _ -- 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] Google voice with no voice
My bad, I found it not loaded in my modules.conf. This is now working. What a pain. Is there a wiki page I can update in order to share the configuration and how to have this work, with everybody ? On 1/22/13 2:58 PM, Joshua Colp wrote: Frank wrote: Hi , So I tried Answer() Wait(1) SendDTMF(1) But I got an error in the console: [Jan 22 14:54:13] WARNING[28067]: pbx.c:4458 pbx_extension_helper: No application 'SendDTMF' for extension (gtalk_incoming, r...@gmail.com, 4) The app_senddtmf.so module has to be built and loaded. You can load it explicitly using module load app_senddtmf.so. If that fails then it was not built and you will have to look into why not. -- _ -- 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] Asterisk, Digium phones, and voicemail.
Hi all, I registered my Digium D70 using a name (D70) instead of a number. Is there a way to program Asterisk (or the phone?) so when I press the MSGS button, it automatically goes to the correct voicemail, with or without asking me for a password ? As of now, it asks me for my mailbox number, which is D70. If I press D70 on the phone (or 370), asterisk does not recognize my phone, of course. Thanks -- _ -- 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] Asterisk, Digium phones, and voicemail.
That worked, thank you. Is there a way to program the keys of the Digiums D70 from asterisk ? Or does everything needs to be done on the phone itself ? On 1/22/13 3:31 PM, Danny Nicholas wrote: Theoretically you can do this Exten = 370,1,Voicemailmain(D70@default) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk, Digium phones, and voicemail. Hi all, I registered my Digium D70 using a name (D70) instead of a number. Is there a way to program Asterisk (or the phone?) so when I press the MSGS button, it automatically goes to the correct voicemail, with or without asking me for a password ? As of now, it asks me for my mailbox number, which is D70. If I press D70 on the phone (or 370), asterisk does not recognize my phone, of course. Thanks -- _ -- 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
Re: [asterisk-users] Asterisk, Digium phones, and voicemail.
I'm going to try to use DPMA. I think I did everything right, but when comes the time to load the res_digium module I have an error showing up: *CLI module load res_digium_phone.so == Host-ID: a:b:c:d:e == Found license 'DPMA-xxx' == Found total of 1 DPMA licenses Unable to load module res_digium_phone.so Command 'module load res_digium_phone.so' failed. Any way to get more information about what's wrong ? Thanks On 1/22/13 4:27 PM, Christopher Harrington wrote: On Tue, Jan 22, 2013 at 2:27 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Hi all, I registered my Digium D70 using a name (D70) instead of a number. Is there a way to program Asterisk (or the phone?) so when I press the MSGS button, it automatically goes to the correct voicemail, with or without asking me for a password ? You don't need to use DPMA if you're provisioning the phones via XML. It's up to you. View https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=21463877 and search for voicemail on the page. That parameter sets what the phone automatically dials; set that value to the extension for that phone's voicemail (as per Danny's email). -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Google voice with no voice
Greetings all,
I was reading the documentation tonight, and decided to try Google voice
with my asterisk.
I was able to setup iksemel, connect to google using jabber, and connect
to google voice using gtalk.
Here is my physical configuration:
Digium D70 -- private network 192.168.1.x -- Airport express --
Internet -- Asterisk with public IP
My asterisk has the following ports open:
5060 tcp/udp from my Airport Express public IP and from voice.google.com
10,000:20,000 from my Airport Express public IP and from voice.google.com
My issue is that when I place a call with google voice, I have no audio
path at all in both way.
When a call is received on google voice (and sent to the D70), if I pick
up, nothing happen, and the caller still hear the ringing tone.
My D70 is setup as follow in the sip.conf:
[D70]
type=friend
nat=yes
qualify=yes
directmedia=no
host=dynamic
secret=takapoum
disallow=all
allow=ulaw
context=LocalSets
mailbox=D70@default
my gtalk.conf is setup as follow:
[general]
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk_incoming
connection=asterisk
and finally, the interesting parts in my extensions.conf are setup as
follow:
;Dialing out on google voice:
exten = _1zxxzxx,1,Dial(Gtalk/asterisk/+${EXTEN}@voice.google.com)
same = n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten = r...@gmail.com,1,Verbose(0, Incoming gtalk from ${CALLERID(all)})
same = n,Answer()
same = n,Wait(2)
same = n,Dial(SIP/D70)
same = Hangup()
I would appreciate if anyone could give me a hint about the audio path.
This is a project that we I will try to setup in a small fire
department, and before I try it, I would like to make sure that my
Digium phones will be able to get full audio path behind private networks.
Thanks a ton for the help !
--
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Re: [asterisk-users] Google voice with no voice
Actually, the funny thing is that it works randomly.
I just tried out of the blue calling from D70 through Google Voice to a
cell phone, and it worked. I hung up, redial, and no audio at all.
On 1/21/13 10:38 PM, Frank wrote:
Greetings all,
I was reading the documentation tonight, and decided to try Google voice
with my asterisk.
I was able to setup iksemel, connect to google using jabber, and connect
to google voice using gtalk.
Here is my physical configuration:
Digium D70 -- private network 192.168.1.x -- Airport express --
Internet -- Asterisk with public IP
My asterisk has the following ports open:
5060 tcp/udp from my Airport Express public IP and from voice.google.com
10,000:20,000 from my Airport Express public IP and from voice.google.com
My issue is that when I place a call with google voice, I have no audio
path at all in both way.
When a call is received on google voice (and sent to the D70), if I pick
up, nothing happen, and the caller still hear the ringing tone.
My D70 is setup as follow in the sip.conf:
[D70]
type=friend
nat=yes
qualify=yes
directmedia=no
host=dynamic
secret=takapoum
disallow=all
allow=ulaw
context=LocalSets
mailbox=D70@default
my gtalk.conf is setup as follow:
[general]
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk_incoming
connection=asterisk
and finally, the interesting parts in my extensions.conf are setup as
follow:
;Dialing out on google voice:
exten = _1zxxzxx,1,Dial(Gtalk/asterisk/+${EXTEN}@voice.google.com)
same = n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten = r...@gmail.com,1,Verbose(0, Incoming gtalk from ${CALLERID(all)})
same = n,Answer()
same = n,Wait(2)
same = n,Dial(SIP/D70)
same = Hangup()
I would appreciate if anyone could give me a hint about the audio path.
This is a project that we I will try to setup in a small fire
department, and before I try it, I would like to make sure that my
Digium phones will be able to get full audio path behind private networks.
Thanks a ton for the help !
--
_
-- 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
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--
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Re: [asterisk-users] DIDForSale spam
With all my respect guys, I do have my asterisk mailing list setup as send-as-soon-as-their-is-a-message. I'm getting too many email from this thread that I seriously don't care about, and that should be taking out of here. If you guys want to discuss, I suggest you email between each other, leaving the asterisk-users out of that. Thank you for your comprehension. On 1/10/13 5:43 PM, C. Savinovich wrote: There is a big difference between publicly posting offering services to the list and harvesting all the email addresses and them contacting everyone privately No way in the world I am going to take side with those guys, I don't know them from a whole in the wall. But your reply begs for the following question: Are you saying that if they would have posted in the regular forum offering their services, then it would have been okay with you? Christian Savinovich */VoIP Telephony Consultant/* 646-982-3572 Original Message Subject: Re: [asterisk-users] DIDForSale spam From: chris tknch...@gmail.com mailto:tknch...@gmail.com Date: Thu, January 10, 2013 5:34 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com There is a big difference between publicly posting offering services to the list and harvesting all the email addresses and them contacting everyone privately On Thu, Jan 10, 2013 at 5:32 PM, C. Savinovich c.savinov...@itntelecom.com mailto:c.savinov...@itntelecom.com wrote: Isn't this precisely the raison d'être for [asterisk-biz]? Oh my goodness!, the asteriz-biz? nooo, they will kill you if you try to post anything offering your services!... that list ceased to provide any value and died a long time ago precisely because its members ran each other away from it. A while back, I wrote a nice click-to-call service and I dared put a post indicating that I was offering it for a fee, and in no time they called me spammer. There is really no incentive to reward someone else's achievements, unless you tell them that you are given them your code for free, then they want it (totally contradicting the meaning of the word business). Christian Savinovich VoIP Telephony Consultant 646-982-3572 Original Message Subject: Re: [asterisk-users] DIDForSale spam From: Chris Bagnall aster...@lists.minotaur.cc mailto:aster...@lists.minotaur.cc Date: Thu, January 10, 2013 5:17 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com On 10 Jan 2013, at 22:09, C. Savinovich c.savinov...@itntelecom.com mailto:c.savinov...@itntelecom.com wrote: Unfortunately, there is a fine line between being a forum where people can exchange ideas, and being a forum where people can find asterisk consultants, and both don't seem to co-exist well together. Isn't this precisely the raison d'être for [asterisk-biz]? Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided byhttp://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 byhttp://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 -- _ -- Bandwidth
[asterisk-users] Auto ban IP addresses
Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? Thank you -- _ -- 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] Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network?
Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network? I have been having some troubles with a Linksys Sipura 2100 series, which suffers from NO AUDIO after a few calls.. Because it is on the same subnet as Asterisk it is configured with nat=no. When you think of it because the Sipura 2100 is a broadband router, the voice part may be considered as being behind NAT, as are other devices plugged into its yellow socket defintely are. In theory is it likely to be better that way? /voipfc -- _ -- 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] Asterisk V/s FreeSwitch
Freeswitch was engineered from scratch by some Asterisk developers who wanted to start afresh on a cleaner programming base. Asterisk is like Topsy, She just growed and had to maintain backward compatibility. The latest versions of Asterisk are reported to be much improved in that respect. On 7 February 2012 15:40, Gilles codecompl...@free.fr wrote: On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- _ -- 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
